MTX DGW v2-0 SCG

Discover the Power of 5
Software Configuration Guide
Dgw v2.0 Application

Document Revision 38
October 08, 2014
Media5 Corporation
Dgw v2.0 Application Software Configuration Guide

2014, Media5 Corporation
All rights reserved. No part of this publication may be reproduced or used in any form or by any means
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Trademarks
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All other trademarks and registered trademarks are the property of their respective owners.
Third-Party Software Copyright Information
The Dgw v2.0 Application firmware aggregates some third-party software modules (open source and
commercial) that are distributed to you in accordance with their respective licenses. Refer to the Third
Party Software Copyright Information addendum available on the Mediatrix Download Portal, which
lists the third-party software modules along with any copyright and license information.
Contents
Preface
About this Manual .......................................................................................................... xxiii
Document Objectives............................................................................................................................... xxiv
Intended Audience.................................................................................................................................... xxv
Related Documentation ............................................................................................................................ xxv
Document Structure.................................................................................................................................. xxv
Document Conventions ........................................................................................................................... xxix
Warning Definition ................................................................................................................................................... xxix
Other Conventions .................................................................................................................................................. xxix
SCN vs. PSTN......................................................................................................................................................... xxix
Standards Supported .............................................................................................................................................. xxix
Chapter 1
System Overview ................................................................................................................ 1
Introduction ...................................................................................................................................................1
Mediatrix 3000 Series.................................................................................................................................................. 1
Mediatrix LP Series ..................................................................................................................................................... 4
Mediatrix C7 Series ..................................................................................................................................................... 4
Key Features ................................................................................................................................................5

Bypass Feature (Mediatrix 4100 Series) ......................................................................................................6
DSP Limitation ..............................................................................................................................................6
Mediatrix 3716............................................................................................................................................................. 8
Mediatrix 3731............................................................................................................................................................. 9
Mediatrix 3732............................................................................................................................................................. 9
Mediatrix 3734............................................................................................................................................................. 9
Mediatrix 3741............................................................................................................................................................. 9
Mediatrix 3742........................................................................................................................................................... 10
Mediatrix 4100 Series................................................................................................................................................ 10
Mediatrix LP Series ................................................................................................................................................... 10
Mediatrix C7Series .................................................................................................................................................... 11
Mediatrix 4400 Series................................................................................................................................................ 11
Management Choices.................................................................................................................................12
RESET/DEFAULT Button ...........................................................................................................................14
At Run-Time .............................................................................................................................................................. 14
At Start-Time ............................................................................................................................................................. 14
Partial Reset.............................................................................................................................................................. 15
Factory Reset ............................................................................................................................................................ 16
User Access................................................................................................................................................17
Secure Password Policies ......................................................................................................................................... 17
Partial Reset.............................................................................................................................................................. 18
Where to Go From Here .............................................................................................................................18
iii
Contents
Chapter 2
Command Line Interface (CLI) ......................................................................................... 19
Introduction .................................................................................................................................................19
Configuring the CLI.....................................................................................................................................19
Partial Reset.............................................................................................................................................................. 20
Accessing the CLI.......................................................................................................................................20

Accessing the CLI through the Serial Console Port (Mediatrix 3000 Series only) .................................................... 20
Accessing the CLI via a Telnet Session .................................................................................................................... 21
Accessing the CLI via a SSH Session....................................................................................................................... 22
Working in the CLI ......................................................................................................................................23

Contexts .................................................................................................................................................................... 23
Exiting from the CLI................................................................................................................................................... 24
Command Completion............................................................................................................................................... 24
Macros....................................................................................................................................................................... 24
History ....................................................................................................................................................................... 24
Services Restart ........................................................................................................................................................ 24
Syslog Messages ...................................................................................................................................................... 25
Configuring the Mediatrix unit with the CLI................................................................................................................ 25
Current Unit Status.................................................................................................................................................... 25
Welcome Message.................................................................................................................................................... 25
Help ........................................................................................................................................................................... 25
List of Commands / Keywords ....................................................................................................................27

access ....................................................................................................................................................................... 27
alias / unalias............................................................................................................................................................. 28
cd............................................................................................................................................................................... 29
columnars.................................................................................................................................................................. 29
commands................................................................................................................................................................. 29
defval......................................................................................................................................................................... 30
dump ......................................................................................................................................................................... 30
get ............................................................................................................................................................................. 31
help............................................................................................................................................................................ 32
indexes ...................................................................................................................................................................... 33
keys ........................................................................................................................................................................... 33
logs off....................................................................................................................................................................... 34
logs on....................................................................................................................................................................... 34
ls................................................................................................................................................................................ 34
name ......................................................................................................................................................................... 35
objects ....................................................................................................................................................................... 35
PCapture ................................................................................................................................................................... 36
ping (IPv4) ................................................................................................................................................................. 36
ping (IPv6) ................................................................................................................................................................. 37
scalars ....................................................................................................................................................................... 37
set.............................................................................................................................................................................. 37
show .......................................................................................................................................................................... 38
sysinfo ....................................................................................................................................................................... 39
tables......................................................................................................................................................................... 39
type............................................................................................................................................................................ 39
Command Execution ................................................................................................................................................. 40
Chapter 3
Web Interface Configuration ............................................................................................ 41
Introduction .................................................................................................................................................41
HTTP User-Agent Header Format............................................................................................................................. 43
Using the Web Interface .............................................................................................................................44

Menu Items................................................................................................................................................................ 45
iv
Submitting Changes ...................................................................................................................................50

Where to Go From Here? ...........................................................................................................................50
Using Secure Communication................................................................................................................................... 51
System Parameters
Chapter 4
Services ............................................................................................................................. 55
Services Table ............................................................................................................................................55
Graceful Restart of Services ..................................................................................................................................... 58
Restarting a Service via MIB ..................................................................................................................................... 58
Chapter 5
Hardware Parameters ....................................................................................................... 61
Hardware Card Configuration .....................................................................................................................61
BRI Hardware Configuration ..................................................................................................................................... 61
PRI Hardware Configuration ..................................................................................................................................... 62
BRI Power Feeding Configuration............................................................................................................................. 64
Bridging Configuration............................................................................................................................................... 65
Chapter 6
Endpoints State Configuration ........................................................................................ 69
Unit Configuration .......................................................................................................................................69
Endpoints Configuration .............................................................................................................................69
Administration .............................................................................................................................................70
Unit Shutting Down Behaviour .................................................................................................................................. 71
Chapter 7
Syslog Configuration ........................................................................................................ 73
Syslog Daemon Configuration ....................................................................................................................73
Configuring PCM Capture ......................................................................................................................................... 76
Configuring the Syslog Daemon Application ............................................................................................................. 77
Chapter 8
Events Configuration ........................................................................................................ 79
Notification Events ......................................................................................................................................79
Deleting a Rule.......................................................................................................................................................... 81
Monitoring Parameters .............................................................................................................................................. 81
Contents
Chapter 9
Local Log ........................................................................................................................... 83
Local Log Status and Entries......................................................................................................................83
Network Parameters
Chapter 10
IPv4 vs. IPv6 ...................................................................................................................... 87
Introduction .................................................................................................................................................87
IPv4 vs. IPv6 Availability.............................................................................................................................87
IPv6 Scope Identifier...................................................................................................................................88
When Contacting the unit using its IPv6 link-local Address ...................................................................................... 88
When Configuring the Mediatrix unit to use an IPv6 link-local Address .................................................................... 89
Chapter 11
Host Parameters................................................................................................................ 91
General Configuration.................................................................................................................................91
Host Configuration ......................................................................................................................................91
Default Gateway Configuration...................................................................................................................93
DNS Configuration......................................................................................................................................94
SNTP Configuration....................................................................................................................................95
Time Configuration .....................................................................................................................................96
STD / DST ................................................................................................................................................................. 97
OFFSET .................................................................................................................................................................... 97
START / END ............................................................................................................................................................ 97
Example .................................................................................................................................................................... 98
Additional Parameters ................................................................................................................................98

Configuring DNS Records Randomization ................................................................................................................ 98
Configuring Pre-resolved Static FQDNs.................................................................................................................... 99
Updating the "sysname" or "syslocation"................................................................................................................. 100
Chapter 12
Interface Parameters....................................................................................................... 101
Reserving an IP Address ..........................................................................................................................101
Link Connectivity Detection ......................................................................................................................101
Partial Reset .............................................................................................................................................101
Interfaces Configuration............................................................................................................................102
IPv6 Autoconfiguration Interfaces ........................................................................................................................... 105
Network Interface Priority ........................................................................................................................................ 106
Rescue Interface Configuration ................................................................................................................106

PPPoE Configuration................................................................................................................................107
PPP Negotiation ...................................................................................................................................................... 108
DHCP Client Identifier Presentation ........................................................................................................................ 108
LLDP Configuration ..................................................................................................................................109

Ethernet Link Configuration ......................................................................................................................110
EAP 802.1x Configuration ........................................................................................................................111
vi
DHCP Server Configuration .....................................................................................................................112

DHCP Negotiation ................................................................................................................................................... 112
Ethernet Connection Speed......................................................................................................................113

Speed and Duplex Detection Issues ....................................................................................................................... 114
Chapter 13
VLAN Parameters ............................................................................................................ 115
VLAN Configuration ..................................................................................................................................115
Chapter 14
Local QoS (Quality of Service) Configuration .............................................................. 117
Introduction ...............................................................................................................................................117
Differentiated Services (DS) Field ...........................................................................................................117
IEEE 802.1q..............................................................................................................................................119
Specific Service Class Configuration........................................................................................................119
Network Traffic Control Configuration.......................................................................................................120
Chapter 15
Local Firewall Configuration .......................................................................................... 123
Managing the Local Firewall .....................................................................................................................123
Partial Reset............................................................................................................................................................ 123
Setting the Default Policy ........................................................................................................................................ 123
Creating/Editing a Firewall Rule .............................................................................................................................. 124
Moving a Firewall Rule ............................................................................................................................................ 126
Deleting a Firewall Rule .......................................................................................................................................... 127
Disabling the Local Firewall ......................................................................................................................127
Chapter 16
IP Routing Configuration................................................................................................ 129
Managing IP Routing ................................................................................................................................129
IPv4 Forwarding ...................................................................................................................................................... 129
Creating/Editing an IP Routing Rule........................................................................................................................ 130
Moving an IP Routing Rule...................................................................................................................................... 131
Deleting an IP Routing Rule .................................................................................................................................... 131
Static IPv4 Routes................................................................................................................................................... 132
DHCPv4 Classless Static Route Option .................................................................................................................. 133
DHCPv4 User Class Route Option.......................................................................................................................... 133
Network Configuration Examples .............................................................................................................134

Forward Packets from the Lan1 Network to the Uplink Network with NAT ............................................................. 134
Configure Port Forwarding for a Web Server Located on the LAN ......................................................................... 134
Chapter 17
Network Firewall Configuration ..................................................................................... 137
Managing the Network Firewall ................................................................................................................137
Setting the Default Policy ........................................................................................................................................ 137
Creating/Editing a Network Firewall Rule ................................................................................................................ 138
vii
Contents
Moving a Network Firewall Rule .............................................................................................................................. 141

Deleting a Network Firewall Rule ............................................................................................................................ 141
Disabling the Network Firewall .................................................................................................................141
Chapter 18
NAT Configuration .......................................................................................................... 143
Introduction ...............................................................................................................................................143
Partial Reset............................................................................................................................................................ 143
Creating/Editing a Source NAT Rule ........................................................................................................143

Creating/Editing a Destination NAT Rule..................................................................................................147
Moving a NAT Rule...................................................................................................................................150
Deleting a NAT Rule .................................................................................................................................150
Disabling the NAT.....................................................................................................................................150
Chapter 19
DHCP Server Settings..................................................................................................... 151
Introduction ...............................................................................................................................................151
Subnet Server ......................................................................................................................................................... 151
Leases..................................................................................................................................................................... 151
Configuration Parameters ....................................................................................................................................... 151
Default vs. Specific Configurations.......................................................................................................................... 152
DHCP Basic Configuration .......................................................................................................................153

Lease Time (Option 51) ............................................................................................................................153
Domain Name (Option 15)........................................................................................................................154
Default Gateway (Option 3) ......................................................................................................................154
DNS (Option 6) .........................................................................................................................................155
NTP (Option 42)........................................................................................................................................156
NBNS (Option 44).....................................................................................................................................157
DHCP Static Leases Configuration...........................................................................................................157
POTS Parameters
Chapter 20
POTS Configuration ........................................................................................................ 161
POTS Status.............................................................................................................................................161
Line Status .............................................................................................................................................................. 161
FXO Line Status ...................................................................................................................................................... 162
General POTS Configuration....................................................................................................................162

Caller ID Information ............................................................................................................................................... 164
FXS Configuration ....................................................................................................................................165

FXS Country Customization .....................................................................................................................167
Calling Party Name of the Caller ID ........................................................................................................................ 168
FXS Bypass ..............................................................................................................................................169

FXO Configuration ....................................................................................................................................170
FXO Dialing Configuration....................................................................................................................................... 171
FXO Answering Configuration................................................................................................................................. 172
FXO Incoming Call Configuration............................................................................................................................ 173
viii
FXO Line Verification .............................................................................................................................................. 173

FXO Force End of Call ............................................................................................................................................ 173
ISDN Parameters
Chapter 21
ISDN Configuration ......................................................................................................... 179
Introduction ...............................................................................................................................................179
ISDN Reference Points ........................................................................................................................................... 179
Inband Tones Generation........................................................................................................................................ 180
Setting PRI Hardware Parameters ...........................................................................................................181

ISDN Auto-Configuration ..........................................................................................................................182
Preset .......................................................................................................................................................183
Partial Reset............................................................................................................................................................ 183
PRI ISDN Statistics...................................................................................................................................183

PRI Configuration .....................................................................................................................................186
InformationFollowing Operation .............................................................................................................................. 196
BRI Configuration .....................................................................................................................................197

Bypass Feature (Mediatrix 3404/3408/3734/3741/3742 Models)............................................................................ 207
Bypass Feature (Mediatrix 4402plus / 4404plus Models) ....................................................................................... 207
Interop Parameters Configuration.............................................................................................................208

Play Local Ringback when no Media Stream .......................................................................................................... 210
ISDN Timers Configuration .......................................................................................................................212

Services Configuration..............................................................................................................................213
IP Sync Configuration ...............................................................................................................................218
Enabling/Disabling IP Synchronization.................................................................................................................... 219
Master Units Configuration...................................................................................................................................... 219
Slave Units Configuration........................................................................................................................................ 220
Resetting the Synchronization Data ........................................................................................................................ 220
R2 CAS Parameters
Chapter 22
R2 CAS Configuration..................................................................................................... 223
Introduction ...............................................................................................................................................223
Line Signals for the Digital Version of MFC/R2 ....................................................................................................... 223
Interregister Signals ................................................................................................................................................ 223
Selecting the R2 Signaling Protocol .........................................................................................................224

R2 Auto-Configuration ..............................................................................................................................225
Preset .......................................................................................................................................................225
Partial Reset............................................................................................................................................................ 226
R2 Channel Associated Signaling ............................................................................................................226

R2 Signaling Variants ...............................................................................................................................230
Override Default Country Settings........................................................................................................................... 230
R2 Signaling Variants.............................................................................................................................................. 231
R2 Timers Variants ...................................................................................................................................234

ix
Contents
R2 Timers Variants.................................................................................................................................................. 235
R2 Digit Timers Variants...........................................................................................................................238

R2 Digit Timers Variants ......................................................................................................................................... 239
R2 Link Timers Variants ...........................................................................................................................239

R2 Link Timers Variants .......................................................................................................................................... 240
R2 Tones Variants ....................................................................................................................................241

R2 Tones Forward Groups ...................................................................................................................................... 243
R2 Tones Backward Groups ................................................................................................................................... 244

PRI R2 CAS Statistics ..............................................................................................................................250
E&M CAS Parameters

Chapter 23
E&M CAS Configuration ................................................................................................. 255
Introduction ...............................................................................................................................................255
Line Signals for the Digital Version of E&M............................................................................................................. 255
Selecting the E&M Signalling Protocol .....................................................................................................256

E&M Auto-Configuration ...........................................................................................................................257
Preset .......................................................................................................................................................257
Partial Reset............................................................................................................................................................ 258
E&M Channel Associated Signaling .........................................................................................................258

E&M Signalling Variants ...........................................................................................................................262
Override Default Signaling Settings ........................................................................................................................ 262
E&M Signalling Variants.......................................................................................................................................... 263
E&M Timers Variants................................................................................................................................265

E&M Timers Variants .............................................................................................................................................. 266
E&M Digit Timers Variants........................................................................................................................267

E&M Digit Timers Variants ...................................................................................................................................... 268
E&M Link Timers Variants ........................................................................................................................269

Override Default Signalling Settings........................................................................................................................ 269
E&M Link Timers Variants....................................................................................................................................... 270
E&M Tones Variants.................................................................................................................................270

Override Default Signalling Settings........................................................................................................................ 270
E&M Tones Variants ............................................................................................................................................... 271

PRI E&M Statistics....................................................................................................................................274
SIP Parameters
Chapter 24
SIP Gateways................................................................................................................... 279
SIP Gateways Configuration.....................................................................................................................279
Chapter 25
SIP Servers ...................................................................................................................... 283
Introduction ...............................................................................................................................................283
TLS Persistent Connections Status ..........................................................................................................284
SIP Servers Configuration ........................................................................................................................284
Multiple SIP Gateways..............................................................................................................................285
SIP Gateway Specific Registrar Servers................................................................................................................. 285
SIP Gateway Specific Messaging Servers .............................................................................................................. 286
SIP Gateway Specific Proxy Servers ...................................................................................................................... 287
Keep Alive ............................................................................................................................................................... 287
SIP Gateway Specific Keep Alive Destinations ....................................................................................................... 289
Outbound Proxy Loose Router Configuration...........................................................................................289
Chapter 26
SIP Registration............................................................................................................... 291
Endpoints Registration..............................................................................................................................291
Contact Domain....................................................................................................................................................... 293
Unit Registration .......................................................................................................................................293

Registration Configuration ........................................................................................................................293
Number of Registrations ...........................................................................................................................295
Additional Registration Refresh Parameters.............................................................................................296
Default Registration Retry Time .............................................................................................................................. 296
Registration Refresh................................................................................................................................................ 296
Registration Expiration ............................................................................................................................................ 297
Expiration Value in Registration .............................................................................................................................. 297
Gateway Specific Registration Retry Time.............................................................................................................. 298
Unregistered Endpoint Behaviour ........................................................................................................................... 298
Unregistered Unit Behaviour ................................................................................................................................... 299
Behaviour on Initial-Registration Reception ............................................................................................................ 300
Registration Delay Value......................................................................................................................................... 301
SIP User Agent Header ............................................................................................................................301
Chapter 27
SIP Authentication .......................................................................................................... 303
Authentication Configuration.....................................................................................................................303
Creating/Editing an Authentication Entry................................................................................................................. 304
Moving an Authentication Entry............................................................................................................................... 306
Deleting an Authentication Entry ............................................................................................................................. 306
xi
Contents
Chapter 28
SIP Transport Parameters .............................................................................................. 307
SIP Transport Type...................................................................................................................................307
Additional Transport Parameters ..............................................................................................................309
Transport TLS Cipher Suite Settings....................................................................................................................... 310
UDP Source Port Behaviour.................................................................................................................................... 310
TLS Client Authentication........................................................................................................................................ 311
Force DNS NAPTR In TLS...................................................................................................................................... 312
SIP Failover Conditions ........................................................................................................................................... 312
Chapter 29
Interop Parameters.......................................................................................................... 315
Behavior on T.38 INVITE Not Accepted ...................................................................................................315
SIP Interop................................................................................................................................................316
SDP Interop ..............................................................................................................................................318
TLS Interop ...............................................................................................................................................322
Misc Interop ..............................................................................................................................................323
Additional Interop Parameters ..................................................................................................................324
Call Waiting Private Number Criteria for SIP INFO ................................................................................................. 324
Max-Forwards Header............................................................................................................................................. 324
Direction Attributes in a Media Stream .................................................................................................................... 325
Local Ring Behaviour on Provisional Response ..................................................................................................... 328
Session ID and Session Version Number in the Origin Field of the SDP................................................................ 329
Register Home Domain Override ............................................................................................................................ 330
DNS SRV Record Lock ........................................................................................................................................... 330
Listening for Early RTP ........................................................................................................................................... 331
Resolve Route Header ............................................................................................................................................ 331
ACK Branch Matching ............................................................................................................................................. 332
Ignore Require Header............................................................................................................................................ 332
Reject Code for Unsupported SDP Offer ................................................................................................................ 333
SIP User-Agent Header Format .............................................................................................................................. 334
SIP INFO Without Content Answer ......................................................................................................................... 334
Keep Alive Option Format ....................................................................................................................................... 335
Unsupported Content-Type ..................................................................................................................................... 335
Chapter 30
Miscellaneous SIP Parameters ...................................................................................... 337
SIP Penalty Box........................................................................................................................................337
Penalty Box vs Transport Types ............................................................................................................................. 337
Penalty Box Configuration....................................................................................................................................... 338
Error Mapping ...........................................................................................................................................339

SIP to Cause Error Mapping ................................................................................................................................... 340
Cause to SIP Error Mapping ................................................................................................................................... 342
Additional Headers ...................................................................................................................................345

PRACK .....................................................................................................................................................346
Forked Provisional Responses Behaviour .............................................................................................................. 347
Session Refresh .......................................................................................................................................347

Background Information .......................................................................................................................................... 348
SIP Gateway Configuration ......................................................................................................................349

SIP Blind Transfer Method........................................................................................................................350
Diversion Configuration ............................................................................................................................350
xii
DNS Configuration....................................................................................................................................351
Event Handling Configuration ...................................................................................................................351
Messaging Subscription............................................................................................................................353
Advice of Charge Configuration................................................................................................................353
Additional DNS Parameters......................................................................................................................354
DNS Failure Concealment....................................................................................................................................... 354
Media Parameters
Chapter 31
Voice & Fax Codecs Configuration ............................................................................... 357
Codec Descriptions...................................................................................................................................357
G.711 A-Law and -Law.......................................................................................................................................... 357
G.723.1.................................................................................................................................................................... 358
G.726....................................................................................................................................................................... 358
G.729....................................................................................................................................................................... 359
Clear Mode.............................................................................................................................................................. 360
Clear Channel ......................................................................................................................................................... 360
X-CCD Clear Channel ............................................................................................................................................. 361
T.38 ......................................................................................................................................................................... 361
Codec Parameters....................................................................................................................................362
Codec vs. Bearer Capabilities Mapping....................................................................................................363
Generic Voice Activity Detection (VAD) ....................................................................................................364
G.711 Codec Parameters .........................................................................................................................365
G.726 Codecs Parameters .......................................................................................................................367
Clear Mode Codec Parameters ................................................................................................................371
Clear Channel Codec Parameters............................................................................................................372
X-CCD Clear Channel Codec Parameters ...............................................................................................374
Fax Parameters ........................................................................................................................................375
Clear Channel Fax .................................................................................................................................................. 375
T.38 Fax .................................................................................................................................................................. 376
T.38 Parameters Configuration ............................................................................................................................... 377
Data Codec Selection Procedure ............................................................................................................................ 379
Chapter 32
Security ............................................................................................................................ 381
Introduction ...............................................................................................................................................381
Security Parameters .................................................................................................................................381
Enforcing Symmetric RTP ....................................................................................................................................... 383
Chapter 33
RTP Statistics Configuration.......................................................................................... 385
Statistics Displayed...................................................................................................................................385
Statistics Configuration .............................................................................................................................387
Channel Statistics .....................................................................................................................................388
xiii
Contents
Chapter 34
Miscellaneous Media Parameters .................................................................................. 391
Jitter Buffer Configuration .........................................................................................................................391
About Changing Jitter Buffer Values ....................................................................................................................... 393
Starting a Call in Voiceband Data Mode ................................................................................................................. 394
DTMF Transport Configuration .................................................................................................................394

DTMF Transport over the SIP Protocol ................................................................................................................... 396
DTMF Detection ...................................................................................................................................................... 396
Using the Payload Type Found in the Answer ........................................................................................................ 398
Quantity of initial packets sent to transmit a DTMF Out-of-Band using RTP .......................................................... 399
Machine Detection Configuration..............................................................................................................399

Base Ports Configuration..........................................................................................................................401
Telephony Parameters
Chapter 35
DTMF Maps Configuration.............................................................................................. 405
Introduction ...............................................................................................................................................405
Syntax.......................................................................................................................................................405
Special Characters .................................................................................................................................................. 406
How to Use a DTMF Map ........................................................................................................................................ 406
General DTMF Maps Parameters.............................................................................................................408

Configuring Timeouts per Endpoint......................................................................................................................... 409
Allowed DTMF Maps ................................................................................................................................409

Refused DTMF Maps................................................................................................................................411
Chapter 36
Call Forward Configuration ............................................................................................ 413
Call Forward On Busy...............................................................................................................................413
Configuring Call Forward on Busy via Handset....................................................................................................... 415
Call Forward On No Answer .....................................................................................................................416

Configuring Call Forward on Answer via Handset................................................................................................... 417
Call Forward Unconditional.......................................................................................................................418

Configuring Call Forward on Unconditional via Handset ......................................................................................... 419
Chapter 37
Telephony Services Configuration ................................................................................ 421
General Configuration...............................................................................................................................421
Automatic Call ......................................................................................................................................................... 423
Call Completion ....................................................................................................................................................... 424
Call Transfer ............................................................................................................................................................ 428
Call Waiting ............................................................................................................................................................. 430
Conference.............................................................................................................................................................. 433
Delayed Hot Line..................................................................................................................................................... 437
Direct IP Address Call ............................................................................................................................................. 438
Call Hold.................................................................................................................................................................. 439
xiv
Second Call ............................................................................................................................................................. 440
Message Waiting Indicator .......................................................................................................................441

Visual Message Waiting Indicator Type .................................................................................................................. 443
Emergency Call Override..........................................................................................................................443

Call Statistics ............................................................................................................................................444
Default Outbound Priority Call Routing.....................................................................................................445
Chapter 38
Tone Customization Parameters Configuration........................................................... 447
Current Tone Definition.............................................................................................................................447
Tone Override...........................................................................................................................................448
Chapter 39
Music on Hold Parameters Configuration..................................................................... 451
MP3 File Download Server .......................................................................................................................451
Configuring the TFTP Server .................................................................................................................................. 451
Configuring the HTTP Server .................................................................................................................................. 451
Music on Hold Configuration.....................................................................................................................451
Chapter 40
Country Parameters Configuration ............................................................................... 455
Country Configuration...............................................................................................................................455
Additional Country Settings ......................................................................................................................456
Input/Output User Gain ........................................................................................................................................... 456
Dialing Settings ....................................................................................................................................................... 457
Fax Calling Tone Detection ..................................................................................................................................... 459
Chapter 41
Call Detail Record............................................................................................................ 461
CDR (Call Detail Record)..........................................................................................................................461
Call Router Parameters

Chapter 42
Call Router Configuration............................................................................................... 467
Introduction ...............................................................................................................................................467
Limitations ............................................................................................................................................................... 468
Regular Expressions ............................................................................................................................................... 469
Routing Type ........................................................................................................................................................... 470
Call Properties Parameters ..................................................................................................................................... 476
SIP/ISDN Call Default Values ................................................................................................................................. 483
Call Routing Status.................................................................................................................................................. 484
xv
Contents
Routes ......................................................................................................................................................485
Creating/Editing a Route ......................................................................................................................................... 486
Moving a Route ....................................................................................................................................................... 490
Deleting a Route...................................................................................................................................................... 491
Mappings ..................................................................................................................................................491
Creating/Editing a Mapping Type ............................................................................................................................ 491
Creating/Editing a Mapping Expression .................................................................................................................. 493
Moving a Mapping Type or Expression Row........................................................................................................... 500
Deleting a Mapping Type or Expression Row ......................................................................................................... 500
Signalling Properties.................................................................................................................................501
Creating/Editing a Signalling Property..................................................................................................................... 501
Moving a Signalling Property Row .......................................................................................................................... 505
Deleting a Signalling Property Row......................................................................................................................... 505
SIP Headers Translations .........................................................................................................................505

Creating/Editing a SIP Headers Translation............................................................................................................ 505
Moving a SIP Headers Translation Row ................................................................................................................. 507
Deleting a SIP Headers Translation Row................................................................................................................ 507
Call Properties Translations......................................................................................................................508

Creating/Editing a Call Properties Translation ........................................................................................................ 508
Moving a Call Properties Translation Row .............................................................................................................. 510
Deleting a Call Properties Translation Row............................................................................................................. 510
Hunt Service .............................................................................................................................................511

Creating/Editing a Hunt ........................................................................................................................................... 511
Call Rejection (Drop) Causes .................................................................................................................................. 514
Moving a Hunt ......................................................................................................................................................... 518
Deleting a Hunt........................................................................................................................................................ 518
SIP Redirects............................................................................................................................................519
Creating/Editing a SIP Redirect............................................................................................................................... 519
Moving a SIP Redirect............................................................................................................................................. 520
Deleting a SIP Redirect ........................................................................................................................................... 520
Hairpinning................................................................................................................................................521
Configuration Examples............................................................................................................................521
Chapter 43
Auto-Routing Configuration........................................................................................... 523
Auto-Routing.............................................................................................................................................523
Endpoints Auto-Routing .......................................................................................................................................... 524
Manual Routing ....................................................................................................................................................... 526
Management Parameters
Chapter 44
Configuration Script........................................................................................................ 531
Configuration Script Server.......................................................................................................................531
Configuring the SNTP Server.................................................................................................................................. 532
Configuring the HTTPS Server................................................................................................................................ 532
Scripts Transfer Cipher Suite Settings .................................................................................................................... 533
Generating a Configuration Script from the Running Configuration .........................................................536

Exporting a Configuration to a URL......................................................................................................................... 536
xvi
Exporting a Configuration Script to your PC............................................................................................................ 537
Executing Configuration Scripts Settings..................................................................................................539

Executing a Script from the Configuration Script Server ......................................................................................... 539
Executing a Script from your PC ............................................................................................................................. 543
Configuration Download Procedure..........................................................................................................544

Automatic Configuration Update...............................................................................................................544
Automatic Update on Restart .................................................................................................................................. 544
Automatic Update at Periodic Intervals ................................................................................................................... 545
DHCPv4 Auto-Provisioning ..................................................................................................................................... 546
Number of Retries ................................................................................................................................................... 547
Creating a Configuration Script.................................................................................................................548
Chapter 45
Configuration Backup/Restore ...................................................................................... 549
Configuration Backup Download Server...................................................................................................549
Image Transfer Cipher Suite Settings ..................................................................................................................... 551
Backup/Restore Configuration..................................................................................................................552
Performing a Backup/Restore to/from on an Image Server or File System ............................................................ 553
Transferring Images Through the Web Browser ..................................................................................................... 557
Chapter 46
Firmware Download ........................................................................................................ 559
What is a Firmware Pack?........................................................................................................................559
Before Performing a Firmware Upgrade or Downgrade ...........................................................................559
Mediatrix Firmware Pack HTTPS Transfer Cipher Suite Settings ........................................................................... 561
Firmware Packs Configuration..................................................................................................................563

Transfer Configuration ..............................................................................................................................565
Certificate Validation ............................................................................................................................................... 566
Default Settings On Install....................................................................................................................................... 567
Firmware Pack Update Procedure............................................................................................................567

Extracting the Firmware Pack Zip File..................................................................................................................... 567
Launching a Firmware Pack Update ....................................................................................................................... 567
Firmware Pack Downgrade ..................................................................................................................................... 568
Firmware Pack Update Status ..................................................................................................................568

Spanning Tree Protocol (STP)..................................................................................................................568
Chapter 47
Certificates Management................................................................................................ 569
Introduction ...............................................................................................................................................569
HTTPS Transfer Cipher Suite Settings ................................................................................................................... 570
Managing Certificates ...............................................................................................................................571

Certificate Authorities................................................................................................................................571
Certificate Upload through the Web Browser ...........................................................................................572
xvii
Contents
Transferring a Certificate via Configuration Script................................................................................................... 572
Host Certificate Associations ....................................................................................................................573
Chapter 48
SNMP Configuration ....................................................................................................... 575
Introduction ...............................................................................................................................................575
SNMP Configuration Section ....................................................................................................................576
Partial Reset............................................................................................................................................................ 580
SNMP Statistics ........................................................................................................................................580
Chapter 49
CWMP Configuration ...................................................................................................... 581
Introduction ...............................................................................................................................................581
Licence Key Activation of TR-069 ........................................................................................................................... 582
CWMP Configuration Section ...................................................................................................................583

General Configuration ............................................................................................................................................. 583
ACS Configuration................................................................................................................................................... 584
ACS Configuration Parameters ............................................................................................................................... 585
Periodic Inform Configuration.................................................................................................................................. 586
TR-069 Configuration .............................................................................................................................................. 588
Transport HTTPS Cipher Suite Settings ................................................................................................................. 592
Transport Certificate Validation ............................................................................................................................... 594
Allow Unauthenticated UDP Connection Requests................................................................................................. 594
Parameter Type Validation...................................................................................................................................... 595
MAC Address Format .............................................................................................................................................. 595
ACS Access to the Local Log Table........................................................................................................................ 596
Supported TR-069 Methods and Parameters...........................................................................................597

Methods................................................................................................................................................................... 597
Parameters.............................................................................................................................................................. 598
Chapter 50
Access Control Configuration ....................................................................................... 607
Users ........................................................................................................................................................607
Partial Reset............................................................................................................................................................ 608
Services Access Control Type..................................................................................................................608

Partial Reset............................................................................................................................................................ 609
Radius Servers .........................................................................................................................................609

Access Rights Description ........................................................................................................................611
Chapter 51
File Manager .................................................................................................................... 613
File Manager.............................................................................................................................................613
Partial Reset............................................................................................................................................................ 613
HTTPS Transfer Cipher Suite Settings ................................................................................................................... 614
xviii
Chapter 52
Miscellaneous.................................................................................................................. 617
Management Interface Configuration .......................................................................................................617
Partial Reset............................................................................................................................................................ 617
Activate Licence........................................................................................................................................617
Appendices
Appendix A
Country-Specific Parameters ......................................................................................... 621
Definitions .................................................................................................................................................621
Conventions ............................................................................................................................................................ 621
Distinctive Ring........................................................................................................................................................ 623
Australia....................................................................................................................................................624
Australia 1 ............................................................................................................................................................... 624
Australia 2 ............................................................................................................................................................... 625
Austria.......................................................................................................................................................626
Brazil.........................................................................................................................................................627
China ........................................................................................................................................................628
Czech Republic.........................................................................................................................................629
Denmark ...................................................................................................................................................630
France.......................................................................................................................................................631
Germany ...................................................................................................................................................632
Germany 1............................................................................................................................................................... 632
Germany 2............................................................................................................................................................... 633
Israel2 .......................................................................................................................................................634
Italy ...........................................................................................................................................................635
Japan ........................................................................................................................................................636
Mexico ......................................................................................................................................................637
Netherlands ..............................................................................................................................................638
New Zealand.............................................................................................................................................639
North America...........................................................................................................................................640
North America 1 ...................................................................................................................................................... 640
Russia.......................................................................................................................................................641
Spain.........................................................................................................................................................642
Sweden.....................................................................................................................................................643
Switzerland ...............................................................................................................................................644
United Arab Emirates................................................................................................................................645
United Arab Emirates 2 ........................................................................................................................................... 645
UK.............................................................................................................................................................648
Appendix B
Scripting Language......................................................................................................... 649
General Scripting Language Syntax .........................................................................................................649
Supported Characters ............................................................................................................................................. 650
Assigning Scalar Values ...........................................................................................................................651

xix
Contents
Assigning Table Cell Values .....................................................................................................................651

Executing Commands...............................................................................................................................652
Normal Commands ................................................................................................................................................. 652
Row Commands ...................................................................................................................................................... 653
DeleteAllRows Command ....................................................................................................................................... 653
Variable Values (Enums) ..........................................................................................................................654

Call Router Specific Information ...............................................................................................................654
Examples ..................................................................................................................................................654
Basic Call ................................................................................................................................................................ 654
Management Functions ........................................................................................................................................... 656
Debugging ............................................................................................................................................................... 656
Advanced Configuration .......................................................................................................................................... 657
Appendix C
Maximum Transmission Unit (MTU) .............................................................................. 661
What is MTU? ...........................................................................................................................................661
Mediatrixs MTU........................................................................................................................................661
Possible Hardware Problem .....................................................................................................................661
Appendix D
Web Interface SNMP Variables Mapping.................................................................... 663
System Page ............................................................................................................................................663
Information Sub-Page ............................................................................................................................................. 663
Services Sub-Page.................................................................................................................................................. 663
Hardware Sub-Page................................................................................................................................................ 664
Endpoints Sub-Page ............................................................................................................................................... 665
Syslog Sub-Page..................................................................................................................................................... 665
Events Sub-Page .................................................................................................................................................... 666
Local Log Sub-Page................................................................................................................................................ 667
Network Page ...........................................................................................................................................667

Status Sub-Page ..................................................................................................................................................... 667
Host Sub-Page ........................................................................................................................................................ 669
Interfaces Sub-Page................................................................................................................................................ 671
VLAN Sub-Page ...................................................................................................................................................... 672
Local Firewall Sub-Page ......................................................................................................................................... 672
IP Routing Sub-Page............................................................................................................................................... 673
Network Firewall Sub-Page..................................................................................................................................... 674
NAT Sub-Page ........................................................................................................................................................ 674
DHCP Server Sub-Page.......................................................................................................................................... 675
QoS Sub-Page ........................................................................................................................................................ 677
ISDN Page................................................................................................................................................678
Status Sub-Page ..................................................................................................................................................... 678
Statistics Section (PRI)............................................................................................................................................ 678
Interface Configuration Section (PRI)...................................................................................................................... 679
Interface Configuration Section (BRI)...................................................................................................................... 680
Interop Configuration Section.................................................................................................................................. 681
Timer Configuration Section.................................................................................................................................... 682
Services Configuration Section ............................................................................................................................... 682
IP Sync Section ....................................................................................................................................................... 683
R2 Page (Mediatrix 3600 Series only) ......................................................................................................683

Status Sub-Page ..................................................................................................................................................... 683
R2 Chanel Associated Signaling Config Sub-Page................................................................................................. 684
Signaling Sub-Page................................................................................................................................................. 684
Timers Sub-Page .................................................................................................................................................... 685
xx
Digit Timers Sub-Page ............................................................................................................................................ 686

Link Timers Sub-Page............................................................................................................................................. 687
Tones Sub-Page ..................................................................................................................................................... 687
IP Sync Section ....................................................................................................................................................... 688
E&M Page (Mediatrix 3500/3600 Series only)..........................................................................................689

Status Sub-Page ..................................................................................................................................................... 689
E&M Channel Associated Signaling Config Sub-Page ........................................................................................... 689
Signaling Sub-Page................................................................................................................................................. 690
Timers Sub-Page .................................................................................................................................................... 691
Digit Timers Sub-Page ............................................................................................................................................ 691
Link Timers Sub-Page............................................................................................................................................. 692
Tones Sub-Page ..................................................................................................................................................... 692
IP Sync Section ....................................................................................................................................................... 692
POTS Page...............................................................................................................................................693
Status Sub-Page ..................................................................................................................................................... 693
Config Sub-Page ..................................................................................................................................................... 694
FXS Config Sub-Page ............................................................................................................................................. 694
FXO Config Sub-Page............................................................................................................................................. 695
SIP Page...................................................................................................................................................696
Gateways Sub-Page ............................................................................................................................................... 696
Servers Sub-Page ................................................................................................................................................... 697
Registrations Sub-Page .......................................................................................................................................... 698
Authentication Sub-Page......................................................................................................................................... 699
Transport Sub-Page ................................................................................................................................................ 700
Interop Sub-Page .................................................................................................................................................... 700
Misc Sub-Page ........................................................................................................................................................ 702
Media Page...............................................................................................................................................704
CODECS Sub-Page ................................................................................................................................................ 704
Security Sub-Page .................................................................................................................................................. 710
RTP Stats ................................................................................................................................................................ 710
Misc Sub-Page ........................................................................................................................................................ 711
Telephony Page........................................................................................................................................712
DTMF Maps Sub-Page............................................................................................................................................ 712
Call Forward Sub-Page ........................................................................................................................................... 714
Services Sub-Page.................................................................................................................................................. 715
Tone Customization Sub-Page ............................................................................................................................... 718
Music on Hold Sub-Page......................................................................................................................................... 719
Misc Sub-Page ........................................................................................................................................................ 719
Call Router Page ......................................................................................................................................720

Status Sub-Page ..................................................................................................................................................... 720
Route Configuration Sub-Page ............................................................................................................................... 721
Auto-Routing Sub-Page .......................................................................................................................................... 725
Management Page ...................................................................................................................................725

Configuration Scripts Sub-Page .............................................................................................................................. 725
Backup / Restore Sub-Page.................................................................................................................................... 727
Firmware Upgrade Sub-Page.................................................................................................................................. 727
Certificates Sub-Page ............................................................................................................................................. 728
SNMP Sub-Page ..................................................................................................................................................... 729
CWMP Sub-Page .................................................................................................................................................... 730
Access Control Sub-Page ....................................................................................................................................... 731
File Sub-Page.......................................................................................................................................................... 731
Misc Sub-Page ........................................................................................................................................................ 731
Appendix E
Glossary ........................................................................................................................... 733
xxi
Contents
Appendix F
List of Acronyms ............................................................................................................. 743
Appendix G
List of Standards Supported .......................................................................................... 747
xxii
R E F A C E
About this Manual

Thank you for purchasing one or more Mediatrix units supporting the Dgw v2.0 application.
The Dgw v2.0 application runs on several Mediatrix devices. Provider-specific profiles ensure that the
Mediatrix unit is a genuine plug and play solution. It offers a low total cost of ownership as it reduces installation
and maintenance costs.
The following Mediatrix units are supported:
Table 1: Mediatrix Units Models
Model
VoIP Call
Capacity
Interfaces
Mediatrix 3000 Series
Mediatrix 3208
6 FXO ports + 2 FXS ports
Up to 8
Mediatrix 3216
Up to 16
Mediatrix 3308
1 FXO port + 7 FXS ports
Mediatrix 3316
16
Mediatrix 3404
5 BRI ports
up to 8
Mediatrix 3408
10 BRI ports
up to 16
Mediatrix 3531
1xT1-PRI interface
up to 23
Mediatrix 3532
2xT1-PRI interface
up to 46
Mediatrix 3621
1xE1-PRI interface
up to 15
Mediatrix 3631
1xE1-PRI interface
up to 30
Mediatrix 3632
2xE1-PRI interface
up to 60
Mediatrix 3716
Up to 16

Mediatrix 3731
1xT1/E1 PRI interface
up to 38
1 x 1 FXO port + 7 FXS ports

Mediatrix 3732
Up to 38

Mediatrix 3734
up to 38
5 BRI ports
Mediatrix 3741
1 x 1 FXO port + 7 FXS ports up to 16

5 BRI ports
Mediatrix 3742
Up to 16
5 BRI ports
Mediatrix 4102S
2 FXS ports
2 Ethernet ports
up to 2
Mediatrix 4104
4 FXS ports
2 Ethernet ports
up to 4
xxiii
Preface - About this Manual
Document Objectives
Table 1: Mediatrix Units Models (Continued)

Model
Interfaces
VoIP Call
Capacity
Mediatrix 4108
8 FXS ports
2a Ethernet ports
up to 8
Mediatrix 4116
16 FXS ports
2a Ethernet ports
up to 16
Mediatrix 4124
24 FXS ports
2a Ethernet ports
up to 24
Mediatrix LP Series
Mediatrix LP16b
16 FXS ports
2a Ethernet ports
up to 16
Mediatrix LP24c
24 FXS ports
2a Ethernet ports
up to 24

Mediatrix 4401
1 BRI port
up to 2
Mediatrix 4401plus
1 BRI port
up to 2
Mediatrix 4402
2 BRI ports
up to 4
Mediatrix 4402plus
2 BRI ports
up to 4
Mediatrix 4404
4 BRI ports
up to 8
Mediatrix 4404plus
4 BRI ports
up to 8
Mediatrix C7 Series
Mediatrix C710
4 FXS ports
up to 4
Mediatrix C711
8 FXS ports
up to 8
Mediatrix C730
4 FXO ports
up to 4
Mediatrix C733
8 FXO ports
up to 8
Mediatrix C731
4 FXS ports + 4 FXO ports
up to 8
a. The
2nd
Ethernet port my not be present on older models.
b. This model can be used to connect to external phone line cables.

c. This model can be used to connect to external phone line cables.
For a more details description of each Mediatrix unit model, please refer to Chapter 1 - System Overview on
page 1.
Document Objectives
The Dgw v2.0 application Software Configuration Guide provides technical information on how to configure
and operate the application for your Mediatrix unit.
Use the Dgw v2.0 application Software Configuration Guide in conjunction with the appropriate publications
listed in Related Documentation on page xxv.
xxiv
Intended Audience
Intended Audience
This Software Configuration Guide is intended for the following users:
System administrators who are responsible for installing and configuring networking equipment
and who are familiar with the Mediatrix unit.
System administrators with a basic networking background and experience, but who might not
be familiar with the Mediatrix unit.
Operators.
Installers.
Maintenance technicians.
Related Documentation
In addition to this manual, the Mediatrix unit document set includes the following:
Model-Specific Hardware Installation Guide

Describes how to install the hardware of your specific Mediatrix unit.
Model-Specific Installation Guide

This booklet allows you to quickly setup and work with the Mediatrix unit.
Configuration Reference Guide

Lists all the parameters, tables, and commands available in the Dgw v2.0 application.
Notification Reference Guide

Lists and describes all syslog messages and notification messages that the Dgw v2.0 application
may send.
Third Party Software Copyright Information

This document lists the third-party software modules used in the Mediatrix unit along with any
copyright and license information.
Be sure to read any readme files, technical bulletins, or additional release notes for important information.
Document Structure
The Dgw v2.0 application Software Configuration Guide contains the following information.
Table 2: Software Configuration Guide Chapter/Appendices
Title
Summary
Chapter 1 - System Overview on page 1
Provides an overview of the Dgw v2.0 application as

well as the units that support it.
Chapter 2 - Command Line Interface (CLI) on

page 19
Describes how to access the CLI environment in order

to perform configuration tasks.
Chapter 3 - Web Interface Configuration on

page 41
Describes how to access the embedded web server of

the Mediatrix unit.
System Parameters
Chapter 4 - Services on page 55
Describes how to view and start/stop system and

network parameters.
xxv
Document Structure
Table 2: Software Configuration Guide Chapter/Appendices (Continued)

Title
Summary
Chapter 5 - Hardware Parameters on page 61
Describes the hardware installed on your Mediatrix

unit.
Chapter 6 - Endpoints State Configuration on

page 69
Describes how to set the administrative state of the

Mediatrix unit endpoints.
Chapter 7 - Syslog Configuration on page 73
Describes how the Mediatrix unit handles syslog

messages and notification messages.
Chapter 8 - Events Configuration on page 79
Describes how to associate a NOTIFICATION

message and how to send it (via syslog or via a SIP
NOTIFY packet).
Network Parameters
Chapter 10 - IPv4 vs. IPv6 on page 87
This chapter describes the differences between IPv4

and IPv6 addressing.
Chapter 11 - Host Parameters on page 91
Describes how to set the host information used by the

Mediatrix unit, as well as the default gateway, DNS
servers and SNTP servers configuration source.
Chapter 12 - Interface Parameters on page 101
Describes how to set the interfaces of the Mediatrix

unit.
Chapter 13 - VLAN Parameters on page 115
Describes how to create and manage dynamic VLANs.
Chapter 14 - Local QoS (Quality of Service)

Configuration on page 117
Describes how to configure packets tagging sent from

the Mediatrix unit.
Chapter 15 - Local Firewall Configuration on

page 123
Describes how to configure the local firewall feature.
Chapter 16 - IP Routing Configuration on

page 129
Describes how to configure the unit's IP routing

parameters.
Chapter 17 - Network Firewall Configuration on

page 137
Describes how to configure the network firewall

parameters.
Chapter 18 - NAT Configuration on page 143
Describes the configuration parameters to define the

Mediatrix units NAT.
Chapter 19 - DHCP Server Settings on

page 151
Describes how to configure the embedded DHCP

server of the Mediatrix unit.
POTS Parameters
Chapter 20 - POTS Configuration on page 161
Describes how to configure the POTS (Plain Old

Telephony System) line service. This applies to the
Mediatrix 4100 and C7 Series only.
ISDN Parameters
Chapter 21 - ISDN Configuration on page 179
Describes how to configure the Integrated Services

Digital Network (ISDN) Basic Rate Interfaces (BRI) or
Primary Rate Interfaces (PRI) parameters. This
applies to the Mediatrix 3000 and Mediatrix 4400
Series only.
R2 CAS Parameters
xxvi
Document Structure

Title
Summary
Chapter 22 - R2 CAS Configuration on

page 223
Describes how to configure the R2 Channel

Associated Signalling (CAS) parameters of the
Mediatrix unit. This applies to the Mediatrix 3600
Series only.
E&M CAS Parameters
Chapter 23 - E&M CAS Configuration on

page 255
Describes how to configure the E&M Channel

Associated Signalling (CAS) parameters of the
Mediatrix unit. This applies to the Mediatrix 3621/3631/
3632 only.
SIP Parameters
Chapter 24 - SIP Gateways on page 279
Describes how to add and remove SIP gateways.
Chapter 25 - SIP Servers on page 283
Describes how to configure the SIP server and SIP

user agent parameters.
Chapter 26 - SIP Registration on page 291
Describes how to configure the registration

parameters of the Mediatrix unit.
Chapter 27 - SIP Authentication on page 303
Describes how to configure authentication parameters

of the Mediatrix unit
Chapter 28 - SIP Transport Parameters on

page 307
Describes the SIP transport parameters you can set.
Chapter 29 - Interop Parameters on page 315
Describes the SIP interop parameters you can set.
Chapter 30 - Miscellaneous SIP Parameters on

page 337
Describes how to configure the SIP penalty box and

SIP transport parameters of the Mediatrix unit.
Media Parameters
Chapter 31 - Voice & Fax Codecs Configuration
on page 357
Describes the various voice and fax codecs

parameters you can set.
Chapter 32 - Security on page 381
Describes how to properly configure the security

Chapter 33 - RTP Statistics Configuration on

page 385
Describes how to read and configure the RTP

statistics.
Chapter 34 - Miscellaneous Media Parameters

on page 391
Describes how to configure parameters that apply to

all codecs.
Chapter 35 - DTMF Maps Configuration on

page 405
Describes how to configure and use the DTMF maps.
Chapter 36 - Call Forward Configuration on

page 413
Describes how to set and use three types of Call

Forward. This applies to the Mediatrix 4100 Series
and C710 / C711 / C731 only.
Chapter 37 - Telephony Services Configuration

on page 421
Describes how to set the Mediatrix unit subscriber

services.
Chapter 38 - Tone Customization Parameters

Describes how to override the pattern for a specific

tone defined for the selected country.
xxvii
Document Structure

Title
Summary
Chapter 39 - Music on Hold Parameters

Describes how to configure the Music on Hold (MOH)

parameters.
Chapter 40 - Country Parameters Configuration

on page 455
Describes how to set the Mediatrix unit with the proper

country settings.

Chapter 42 - Call Router Configuration on
page 467
Describes the call router service.
Chapter 43 - Auto-Routing Configuration on

page 523
Describes the auto-routing feature.
Chapter 44 - Configuration Script on page 531
Describes how to use the configuration scripts

download feature to update the Mediatrix unit
configuration.
Chapter 45 - Configuration Backup/Restore on

page 549
Describes how to backup and restore the Mediatrix

unit configuration.
Chapter 46 - Firmware Download on page 559
Describes how to download a firmware pack available

on the designated update files server into the
Mediatrix unit.
Chapter 47 - Certificates Management on

page 569
Describes how to transfer and manage certificates

into the Mediatrix unit.
Chapter 48 - SNMP Configuration on page 575
Describes to configure the SNMP privacy parameters

of the Mediatrix unit.
Chapter 49 - CWMP Configuration on page 581
Describes how to set the CWMP parameters of the

Mediatrix unit.
Chapter 50 - Access Control Configuration on

page 607
Describes how to set the Access Control parameters

of the Mediatrix unit.
File Manager on page 613
This chapter describes how to use the units File

Manager.
Chapter 52 - Miscellaneous on page 617
Describes how to set various parameters used to

manage the Mediatrix unit.
Appendices
xxviii
Appendix A - Country-Specific Parameters on

page 621
Lists the various parameters specific to a country such

as loss plan, tones and rings, etc.
Appendix B - Scripting Language on page 649
Describes the Media5 proprietary scripting language. It

also lists a few configuration samples that can be
pasted or typed into the CLI or downloaded into the
Mediatrix unit via the Configuration Script feature.
Appendix C - Maximum Transmission Unit

(MTU) on page 661
Describes the MTU (Maximum Transmission Unit)

requirements of the Mediatrix unit.
Appendix D - Web Interface SNMP Variables

Mapping on page 663
Lists the SNMP variables corresponding to the web

interface of the Mediatrix unit
Document Conventions
The following information provides an explanation of the symbols that appear on the Mediatrix unit and in the
documentation for the product.
Warning Definition
Warning: Means danger. You are in a situation that could cause bodily injury. Before you work on any
equipment, you must be aware of the hazards involved with electrical circuitry and be familiar with standard
practices for preventing accidents.
Where to find Translated Warning Definition

For safety and warning information, refer to Appendix A - Standards Compliance and Safety Information in
the Mediatrix unit Hardware Installation Guide. This Appendix describes the international agency compliance
and safety information for the Mediatrix unit. It also includes a translation of the safety warning listed in the
previous section.
Other Conventions
The following are other conventions you will encounter in this manual.
Caution: Indicates a potentially hazardous situation which, if not avoided, may result in minor or moderate
injury and/or damage to the equipment or property.
Note: Indicates important information about the current topic.

Standards Supported
Indicates which RFC, Draft or other standard document is supported for a

specific feature.
SCN vs. PSTN

In Media5 and other vendors documentation, the terms SCN and PSTN are used. A SCN (Switched Circuit
Network) is a general term to designate a communication network in which any user may be connected to any
other user through the use of message, circuit, or packet switching and control devices. The Public Switched
Telephone Network (PSTN) or a Private Branch eXchange (PBX) are examples of SCNs.
Standards Supported
When available, this document lists the standards onto which features are based. These standards may be
RFCs (Request for Comments), Internet-Drafts, or other standard documents.
The Dgw v2.0 applications implementations are based on the standards, so its possible that some behaviour
differs from the official standards.
For more information on and a list of RFCs and Internet-Drafts, refer to the IETF web site at http://www.ietf.org.
xxix
xxx
H A P T E R
System Overview
This chapter provides an overview of the Mediatrix devices supported by the Dgw v2.0 application:
Introduction to the Mediatrix devices and the models available.

Description of the Bypass feature for models that support it.
Description of the various ways to manage the Mediatrix unit.
How to use the DEFAULT/RESET button (partial reset and factory reset procedures).
How to configure user access to the Mediatrix unit.
Introduction
The Dgw v2.0 application runs on several Mediatrix devices. Provider-specific profiles ensure that the
Mediatrix unit is a genuine plug and play solution. It offers a low total cost of ownership as it reduces installation
and maintenance costs.
Moreover, the Mediatrix unit integrates features such as TLS, SRTP, and HTTPS designed to bring enhanced
security for network management, SIP signalling and media transmission aspects.
The following describes the devices that the application supports.

The Mediatrix 3000 series ISDN VoIP Digital Gateways allow enterprises to lower communications costs over
any IP link. Designed specifically for enterprise applications, the Mediatrix 3000 digital gateways make use of
existing broadband access equipment to connect to any standards-based VoIP network.
The Mediatrix 3000 digital gateways meet the requirements of enterprises that want to connect their ISDN
equipment, such as PBXs, through a BRI (Basic Rate Interface) or PRI (Primary Rate Interface) interface to
an IP network or as a gateway to the PSTN.
Mediatrix 3000 digital gateways provide transparent ISDN port extensions over an IP network. The remote
ISDN terminals can be managed centrally and benefit from PBX services such as calling groups and voice
mail.
The following are the current Mediatrix 3000 Series models:
Table 3: Mediatrix 3000 Models
Model
Interfaces
VoIP Calls Capacity Service Slot Used
Mediatrix 3404
5 BRI ports
up to 8
Slot 2
Mediatrix 3408
10 BRI ports
up to 16
Slot 2 and Slot 3
Mediatrix 3531
1xT1-PRI interface
up to 23
Slot 2
Mediatrix 3532
2xT1-PRI interface
up to 46
Slots 2 and 3
Mediatrix 3621
1xE1-PRI interface
up to 15
Slot 2
Mediatrix 3631
1xE1-PRI interface
up to 30
Slot 2
Mediatrix 3632
2xE1-PRI interface
up to 60
Slots 2 and 3
Chapter 1 - System Overview
Introduction
The Mediatrix 3200 Series offers a mix of FXS and FXO interfaces. Its FXO interfaces make it the ideal solution
to deploy private or hosted toll bypass networks. It provides a simple, transparent and cost-effective way of
maintaining a connection to the PSTN.
The FXO interfaces provide a gateway to the PSTN for IP-based PBX and Key Systems. Thereby, Mediatrix
3200 gateways allow the deployment of VoIP Remote Line Extension and Branch Office Connectivity solutions
without sacrificing any local PSTN access points.
By connecting CO lines from a selected site to VoIP networks, the Mediatrix 3200 series enable service
providers and enterprises to use VoIP connections between pre-determined local networks.
Model
Interfaces
Mediatrix 3208
Up to 8
Slot 2
Mediatrix 3216
Up to 16
Slot 2 and Slot 3
See the Hardware Configuration Guide for more details.

The Mediatrix 3300 is an essential component of any enterprise network, for voice, data, and security needs.
It connects analog phones, faxes, and lines to an IP network to allow cost-effective communication scenarios
for immediate savings and enhanced productivity.
It also offers an FXO port that can provide SCN access for various VoIP endpoints such as IP phones, FXS
devices, softphones and IP-based PBX and Key Systems (see FXO Configuration on page 170 for more
details).
The following are the current 3300 Series models:
Table 7: Mediatrix unit Models
Model
Mediatrix 3308
Interfaces
7 FXS Lines

up to 8
Slot 2
up to 16
Slots 2 and 3
1 FXO Line
Mediatrix 3316
14 FXS Lines
2 FXO Lines
See the Hardware Configuration Guide for more details.

The Mediatrix 3700 series models offer different combinations of T1/E1, BRI and FXS/FXO interfaces in a
unique analog/digital offering to allow enterprises to lower communications costs over any IP link. Designed
specifically for enterprise applications, the Mediatrix 3700 series models make use of existing broadband
access equipment to connect to any standards-based VoIP network.
The Mediatrix 3700 series models meet the requirements of enterprises that want to connect their ISDN
equipment, such as PBXs, through a BRI (Basic Rate Interface) or PRI (Primary Rate Interface) interface to
an IP network or as a gateway to the PSTN.
They also offer to connect analog phones, faxes, and lines to an IP network to allow cost-effective
communication scenarios for immediate savings and enhanced productivity.
Introduction

Model
Mediatrix 3716
Interfaces
Up to 16
Slots 2 and 3
up to 38
Slots 2 and 3
Up to 38
Slots 2 and 3
up to 38
Slots 2 and 3
up to 16
Slots 2 and 3
Up to 16
Slots 2 and 3

Mediatrix 3731

7 FXS ports
Mediatrix 3732

Mediatrix 3734

5 BRI ports
Mediatrix 3741
7 FXS ports
5 BRI ports
Mediatrix 3742

5 BRI ports
The Mediatrix 3716, 3731, 3732, 3741 and 3742 models offer true FXO interfaces, which makes them the ideal
solution to deploy private or hosted toll bypass networks. They provide a simple, transparent and costeffective way of maintaining a connection to the PSTN.
The FXO interfaces provide a gateway to the PSTN for IP-based PBX and Key Systems. Thereby, Mediatrix
3700 gateways with FXO interfaces allow the deployment of VoIP Remote Line Extension and Branch Office
Connectivity solutions without sacrificing any local PSTN access points.
By connecting CO lines from a selected site to VoIP networks, the Mediatrix 3700 series enable service
providers and enterprises to use VoIP connections between pre-determined local networks.

The Mediatrix 4100 is a standalone Internet telephony access device that connects to virtually any business
telephone system supporting standard analog lines.
The following are the Mediatrix 4100 Series models currently available:
Table 11: Mediatrix 4100 Series Models
Model
Interfaces
Mediatrix 4102S
2 FXS ports
2 Ethernet ports
Mediatrix 4104
4 FXS ports
2 Ethernet ports
Mediatrix 4108
8 FXS ports
2a Ethernet ports
Mediatrix 4116
16 FXS ports
2a Ethernet ports
Mediatrix 4124
24 FXS ports
2a Ethernet ports
Bypass
a. The 2nd Ethernet port my not be present on older models.
Introduction
Mediatrix LP Series
The Mediatrix LP Series access devices are high-quality and cost efficient VoIP gateways connecting larger
branch offices and multi-tenant buildings to an IP network, while preserving investment in analog telephones
and faxes. They offer features such as 60 VRMS ringing voltage, 2 kilometers loop distance and integrated
secondary lightning protection, which allow them to be connected to external phone line cables.
The following are the Mediatrix LP Series models currently available:
Table 13: Mediatrix LP Series Models
Model
Interfaces
Mediatrix LP16
16 FXS ports
2a Ethernet ports
Mediatrix LP24
24 FXS ports
2a Ethernet ports
a. The 2nd Ethernet port my not be present on older models.

The Mediatrix 4400 series ISDN BRI VoIP Digital Gateways allow enterprises to lower communications costs
over any IP link. Designed specifically for enterprise applications, the Mediatrix 4400 digital gateways make
use of existing broadband access equipment to connect to any standards-based VoIP network.
The Mediatrix 4400 digital gateways meet the requirements of enterprises that want to connect their ISDN
equipment, such as PBXs, through a BRI (Basic Rate Interface) to an IP network or as a gateway to the PSTN.
Mediatrix 4400 digital gateways provide transparent ISDN port extensions over an IP network. The remote
ISDN terminals can be managed centrally and benefit from PBX services such as calling groups and voice
mail.
The following are the Mediatrix 4400 models currently available:
Table 14: Mediatrix 4400 Series Models
Model
Interfaces
VoIP Call
Capacity
Mediatrix 4401
1 BRI port
up to 2
Mediatrix 4401plus
1 BRI port
up to 2
Mediatrix 4402
2 BRI ports
up to 4
Mediatrix 4402plus
2 BRI ports
up to 4
Mediatrix 4404
4 BRI ports
up to 8
Mediatrix 4404plus
4 BRI ports
up to 8
BRI Power
Feeding
Power over
Ethernet
Bypass
Mediatrix C7 Series
The Mediatrix C7 Series are multi-function devices combining VoIP Analog Adapter, Gateway and QoS control
in a secure and powerful platform.
This platform, featuring FXS/FXO interfaces, provides an ideal solution for enterprise voice applications or for
connecting to a service provider's broadband access.
With FXO interfaces, it is the ideal solution to deploy private or hosted toll bypass networks. It provides a
simple, transparent and cost-effective way of maintaining a connection to the PSTN.
The Mediatrix C7 Series also allows Enterprises, Service Providers, and System Integrators to deploy secure
systems and generate additional revenue streams.
The Mediatrix C7 Series are available in the following models:
Key Features
Table 15: Mediatrix C7 Series Models

Model
Interfaces
Mediatrix C710
4 FXS ports
Mediatrix C711
8 FXS ports
Mediatrix C730
4 FXO ports
Mediatrix C733
8 FXO ports
Mediatrix C731
4 FXS ports +
4 FXO ports
Key Features
The following are the key features offered by the various models available.
Table 17: Mediatrix Units Key Features
Feature
3400/
3500/
3600
3200/
3300
C7
Series
4400
up to 60 up to 14 up to 38 up to 24 up to 8
up to 8
3700
4100/
LP
IP connectivity for analog phones and faxes

Number of simultaneous calls
FXS interface ports
FXO interface ports
ISDN BRI interface ports
ISDN T1/E1 PRI interface ports
HTTP, SNMP, FTP and TFTP for configuration and
management
True Plug-and-Play
Automatic configuration script download
Call Routing service
Secure SIP signalling
Secure Media transmission
SNMPv3 and web management
DHCP Client
PPPoE Client
T.38 support
Command Line Interface (CLI)
SSL/TLS Encryption
Optional Power Feeding Over Ethernet (plus models).
Optional Power Feeding over ISDN Line (plus
models).
Bypass Feature (Mediatrix 4100 Series)
Table 17: Mediatrix Units Key Features (Continued)

Feature
3400/
3500/
3600
3200/
3300
3700
4100/
LP
C7
Series
4400
60 VRMS ringing voltage, 2 kilometres loop distance

and integrated secondary lightning protection to
connect to external phone line cables (Mediatrix LP
16/Mediatrix LP24/Mediatrix C710/Mediatrix C711/
Mediatrix C731 models).
Bypass Feature (Mediatrix 4100 Series)

During normal operation, the SCN line connected to the Bypass connector is switched out of the circuit through
commuting relays. The Bypass connector can be activated by two different conditions:
When power is removed from the Mediatrix unit.

When the IP network is down.
Note: The Mediatrix 4102S does not have the Bypass feature.
This is indicated by the In Use LED being steady ON (except when the power is removed). If one of these
conditions is met, a phone/fax used on FXS connector 1 (Mediatrix 4104/4108/4116) or analog line 1
(Mediatrix 4124) is directly connected to the SCN Bypass line. FXS connector 1 (Mediatrix 4104/4108/4116)
or analog line 1 (Mediatrix 4124) stays in Bypass connection until:
The error conditions have been cleared.

The device connected to it is on-hook and a delay has elapsed.
DSP Limitation
Some of the Mediatrix unit models currently suffer from a limitation of their DSPs. When using a codec other
than G.711, enabling Secure RTP (SRTP) and/or using conferences has an impact on the Mediatrix units
overall performance as SRTP and conferences require CPU power. This means there is a limitation on the
lines that can be used simultaneously, depending on the codecs enabled and SRTP. This could mean that a
user picking up a telephone on these models may not have a dial tone due to lack of resources in order to not
affect the quality of ongoing calls. See Security on page 201 for more details on SRTP limitations.
The DSPs offer channels as resources to the Mediatrix unit. The Mediatrix unit is limited to two conferences
per DSP. See Conference on page 433 for more details on Conference limitations.
Please note that:
One FXS line requires one channel.

There is a maximum of 2 conferences per DSP
Each conference requires one additional channel
In the following tables, compressed RTP refers to codecs other than G.711. Numbers in Bold indicate a
possible under-capacity.
DSP Limitation

Table 18 describes the Mediatrix 3200 series processing capacity.
Table 18: Mediatrix 3200 Offered Channels vs. Processing Capacity
Offered Chanels
Processing Capacity
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
Mediatrix 3208
10
Mediatrix 3216
16
4a
20
16
16
12
Model
a. Two conferences available for lines 1-7 and two conferences available for lines 8-14.

Offered Chanels
Model
Mediatrix 3308
Mediatrix 3316
Processing Capacity
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
10
20
16
16
12
16

Offered Chanels
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
Mediatrix 3404
(1 x 5 BRI
Card)
Mediatrix 3408
(2 x 5 BRI
Card)
16
16
16
16
12
Model
Processing Capacity
DSP Limitation

Offered Chanels
Processing Capacity
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
Mediatrix 3531
(1 PRI Card)
23
23
23
23
23
Mediatrix 3532
(2 PRI Cards)
46
46
46
46
46
Model

Offered Chanels
Processing Capacity
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
15
15
15
15
15
Mediatrix 3631
(1 PRI Card)
30
30
30
30
30
Mediatrix 3632
(2 PRI Cards)
60
60
60
60
60
Model
Mediatrix 3621
(1 PRI Card)
Mediatrix 3716
Table 23 describes the Mediatrix 3716 processing capacity.
Offered Chanels
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
2 FXS-6 FXO
10
7 FXS-1 FXO
10
Card
Processing Capacity
DSP Limitation
Mediatrix 3731
Offered Chanels
Processing Capacity
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
7 FXS-1 FXO
10
PRI
30
30
30
30
30
Card
Mediatrix 3732
Offered Chanels
Processing Capacity
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
2 FXS-6 FXO
10
PRI
30
30
30
30
30
Card
Mediatrix 3734
Offered Chanels
Processing Capacity
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
5 BRI
PRI
30
30
30
30
30
Card
Mediatrix 3741
Offered Chanels
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
5 BRI
7 FXS-1 FXO
10
Card
Processing Capacity
DSP Limitation
Mediatrix 3742
Offered Chanels
Processing Capacity
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
5 BRI
2 FXS-6 FXO
10
Card

Offered Chanels
Processing Capacity
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
Mediatrix 4102
Mediatrix 4104
10
Mediatrix 4108
10
20
16
16
12
30
24
24
18
Model
Mediatrix 4116
16
Mediatrix 4124
34
6b
b. Two conferences available for lines 1-8, two conferences available for lines 9-16, and two conferences available for lines 17-24.
Mediatrix LP Series
Table 30 describes the Mediatrix LP series processing capacity.
Table 30: Mediatrix LP Offered Channels vs. Processing Capacity
Offered Chanels
Processing Capacity
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
Mediatrix
LP16
16
4a
20
16
16
12
Mediatrix
LP24
34
6b
30
24
24
18
Model
b. Two conferences available for lines 1-8, two conferences available for lines 9-16, and two conferences available for lines 17-24.
10
DSP Limitation
Mediatrix C7Series
Table 31 describes the Mediatrix C7 series processing capacity.
Table 31: Mediatrix C7 Offered Channels vs. Processing Capacity
Offered Chanels
Processing Capacity
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
Mediatrix
C710 (4 FXS)
10
10
10
10
Mediatrix
C711 (8 FXS)
10
10
10
10
Mediatrix
C730(4 FXO)
10
10
10
10
Mediatrix
C733 (8 FXO)
10
10
10
10
Mediatrix
C731 (4 FXS 4 FXO)
10
10
10
10
Model

Table 32 describes the Mediatrix series processing capacity.
Table 32: Mediatrix Offered Channels vs. Processing Capacity
Offered Chanels
Phys.
Channels
3-way Conf.
Channels
G.711 RTP
Channels
Compr. RTP
Channels
G.711 SRTP
Channels
Compr.
SRTP
Channels
Mediatrix 4401
10
Mediatrix 4402
10
Mediatrix 4404
10
Model
Processing Capacity
11
Management Choices
Management Choices
The Mediatrix unit offers various management options to configure the unit.
Figure 1: Management Interfaces
Local
Configuration
Management
Configuration Files
Mediatrix Unit
Manager
Network (UMN)
Provisioning Server
Command Line
Interface (CLI)
Mediatrix Unit
SNMP
Queries
Web GUI
User
External EMS
Table 33: Management Options

Management Choice
Web GUI
Description
The Mediatrix unit web interface offers
the following options:
Password-protected access
via basic HTTP
authentication, as described
in RFC 2617
User-friendly GUI
Refer to Chapter 3 - Web Interface

Configuration on page 41 for more
details.
SNMPv1/2/3
The Mediatrix unit SNMP feature offers

the following options:
Remote management
Simultaneous management
Features
The Mediatrix unit web interface allows
you to configure the following
information:
Network attributes
SIP parameters
VoIP settings
Management settings such

as configuration scripts,
restore / backup, etc.
The Mediatrix unit SNMP feature allows

you to configure all the MIB services.
Refer to Chapter 48 - SNMP

Configuration on page 575 for more
details.
12
Management Choices
Table 33: Management Options (Continued)

Management Choice
Command Line
Interface (CLI)
Description
The Mediatrix unit uses a proprietary
CLI to configure all the units
parameters.
Features
The Mediatrix unit CLI feature allows
you to configure all the MIB services.
Refer to Chapter 2 - Command Line

Interface (CLI) on page 19 for more
details.
Unit Manager
Network
The Unit Manager Network (UMN) is a

PC-Windows based element
management system designed to
facilitate the deployment, configuration
and provisioning of Mediatrix access
devices and gateways.
The UMN offers the following:
Auto-discovery
Group provisioning
SNMP access and remote

management.
The UMN enables the simple and

remote configuration and deployment of
numerous Mediatrix units.
13
RESET/DEFAULT Button
The RESET/DEFAULT button allows you to:
Cancel an action that was started.
Reconfigure a unit.
Revert to known factory settings if the Mediatrix unit refuses to work properly for any reason or
the connection to the network is lost.
At Run-Time
You can use the RESET/DEFAULT button at run-time you can press the button while the Mediatrix unit is
running without powering the unit off. Table 34 describes the actions you can perform in this case.
Table 34: RESET/DEFAULT Button Interaction
RESET/
DEFAULT Button
Pressed for:
2 to 6 seconds
Action
Restarts the
Mediatrix unit
Comments
No changes are made to the
Mediatrix unit settings.
LEDs Pattern
Power LED:
blinkinga, 1Hz,
50% duty
All other LEDs:
7 to 11 seconds
12 to 16 seconds
17 seconds and
more
OFF
Sets the Mediatrix

unit in Partial Reset
Mode
Sets some of the Mediatrix unit

configuration to pre-determined
values.
All LEDs
Restarts the
Mediatrix unit in
Factory Reset
Deletes the persistent

configuration values, creates a
new configuration file with the
default factory values, and then
restarts the unit.
All LEDs
No action is taken
The RESET/DEFAULT button

pressed event is ignored.
N/A
blinking, 1Hz,
50% duty
steady ON
a. The Power LEDs of the Mediatrix 3000 Series models blink green.
At Start-Time
You can use the RESET/DEFAULT button at start-time you power the unit off, and then depress the button
until the LEDs stop blinking and remain ON. This applies the Factory Reset procedure (see Factory Reset
on page 16). This feature reverts the Mediatrix unit back to its default factory settings.
14
Partial Reset
The Partial reset provides a way to contact the Mediatrix unit in a known and static state while keeping most
of the configuration unchanged.
Following a partial reset, the Mediatrix unit management interface is set to the Rescue interface. The default
IPv4 address for this interface is 192.168.0.1/24 and has its corresponding link-local IPv6 available and printed
on the sticker under the Mediatrix unit (see Chapter 10 - IPv4 vs. IPv6 on page 87 for more details). Any
existing network interface that conflicts with the Rescue interface address is disabled.
You can contact the Mediatrix unit at this address to access its configuration parameters. It is not advised to
access the unit on a regular basis through the Rescue network interface. You should reconfigure the units
network interfaces as soon as possible in order to access it through another interface.
In a partial reset, the following services and parameters are also affected:
AAA service: User(s) from profile are restored with their factory password.
SNMP service: Resets the enableSnmpV1, enableSnmpV2, enableSnmpV3 and snmpPort
values to their default values.
WEB service: Resets the serverPort to its default value.
LFW service: When a partial reset is triggered and the firewall is enabled, the configuration is
rolled back if it was being modified. A new rule is then automatically applied in the firewall to
allow access to the 'Rescue' interface. However, if the firewall is disabled, the configuration is
rolled back but no rule is added.
HOC service: The Management Interface reverts back to its default value.
CLI service: The CLI variables revert back to their default value.
NAT service: The configuration is rolled back if it was being modified. A new rule is then
automatically applied in the source and in the destination NAT tables to prevent incorrect rules
from blocking access to the unit. If those rules are not the first priority, they are raised. If there
are no rules in the tables, the new rules are not added since there are no rules to override.
BNI service: The Rescue interface is configured and enabled with:
its hidden IPv4 link configuration values
its hidden IPv4 address configuration
an IPv6 link-local address on all network links
Hidden values are set by the unit's profile.

Just before the Rescue is configured, all IPv4 network interfaces that could possibly conflict with the
Rescue interface are disabled.
If the BNI Service is stopped when the partial reset occurs, it is started and the above configuration
is applied.
To trigger the Partial Reset:

1.
Insert a small, unbent paper clip into the RESET/DEFAULT hole located at the rear of the Mediatrix
unit. While pressing the RESET/DEFAULT button, restart the unit.
Do not depress before all the LEDs start blinking (between 7-11 seconds).
2.
Release the paper clip.

This procedure can also be performed at run-time.
Disabling the Partial Reset

This section describes configuration that is available only in the MIB parameters of the Mediatrix unit. You can
configure these parameters as follows:
by using a MIB browser

by using the CLI
by creating a configuration script containing the configuration variables
15
You can disable the partial reset procedure, even if users depress the Reset/Default button. The following
parameters are supported:
Table 35: Partial Reset Parameters
Parameter
All
Description
All the actions are allowed: reset, partial reset and factory reset.
DisablePartialReset All actions are allowed except the partial reset.
The reset action restarts the unit.
The factory reset action reverts the unit back to its default factory settings.
The partial reset action provides a way to contact the unit in a known and static state while
keeping most of the configuration unchanged.
To change the partial reset behaviour:

1.
In the hardwareMIB, set the ResetButtonManagement variable to the proper behaviour.

You can also use the following line in the CLI or a configuration script:
hardwareMIB.ResetButtonManagement="Value"
where:
Value may be as follows:

Table 36: Partial Reset Values
Value
Meaning
100
All
200
DisablePartialReset
Factory Reset
The Factory reset reverts the Mediatrix unit back to its default factory settings. It deletes the persistent MIB
values of the unit, including:
The firmware pack download configuration files.

The SNMP configuration, including the SNMPv3 passwords and users.
The PPPoE configuration, including the PPP user names and passwords.
The Factory reset creates a new configuration file with the default factory values. It should be performed with
the Mediatrix unit connected to a network with access to a DHCP server. If the unit cannot find a DHCP server,
it sends requests indefinitely.
The following procedure requires that you have physical access to the Mediatrix unit. However, you can also
trigger a factory reset remotely:
via the web interface of the Mediatrix unit. See Firmware Packs Configuration on page 563
for more details.
via the Command Line Interface of the Mediatrix unit by using the fpu.defaultsetting
command. See Chapter 2 - Command Line Interface (CLI) on page 19 for more details on the
CLI.
To trigger the Factory Reset:

1.
Power the Mediatrix unit off.
2.
Insert a small, unbent paper clip into the RESET/DEFAULT hole located at the rear of the Mediatrix
unit. While pressing the RESET/DEFAULT button, restart the unit.
Do not depress before the LEDs stop blinking and are steadily ON. This could take up to 30
seconds.
3.
16
Release the paper clip.

User Access
The Mediatrix unit restarts.

This procedure resets all variables in the MIB modules to their default value.
When the Mediatrix unit has finished its provisioning sequence, it is ready to be used with a DHCPprovided IP address and MIB parameters.
This procedure can also be performed at run-time.
Note: The Factory reset alters any persistent configuration data of the Mediatrix unit.
User Access

by using the CLI
The following describes how to configure user access to the Mediatrix unit. The access information is available
for the SNMP and Web interface management methods.
Note: Currently, the user name cannot be modified. To access the unit via SNMPv1, you must use the user
name as being the community name and there must be no password for this user name.
To configure the Mediatrix unit user access:

1.
In the aaaMIB, set the password associated with the user name in the usersPassword variable.
aaa.users.Password[UserName="User_Name"]="Value"
Only the admin and public user names are available for the moment.
2.
Set the user name that is used for scheduled tasks in the batchUser variable.
aaa.batchUser="Value"
For instance, if you are using an automatic configuration update everyday at midnight, the relevant
service will use the batchUser user to execute the request.
Secure Password Policies

It is possible to validate a password against some password policies to be considered as valid. These policies
may only be activated via customized profiles created by Media5. The available policies are:
Table 37: Secure Password Policies
Policy
Description
Minimum Length of User Password The minimum length the user password must have to be considered
as valid.
Upper and Lower Case Required
on User Password
Indicates if the user password is required to contain an upper and a

lower case characters to be considered as valid.
Here is an example of a valid password : 'Password' and examples of
invalid passwords : '1234', 'password', '1password', 1PASSWORD.
17
Where to Go From Here
Table 37: Secure Password Policies (Continued)

Policy
Numeral character Required on
User Password
Description
Indicates if the user password is required to contain a numeral
character to be considered as valid.
Here is an example of a valid password : '1password2' and examples
of invalid passwords : 'password', 'Password'.
Special character Required on

User Password
Indicates if the user password is required to contain a special character

to be considered as valid.
Here is an example of a valid passwords : 'pass$word', 'pass_word#'
and examples of invalid passwords : 'password', 'Password', '1234',
'1Password'.
For more information on how to get a customized user profile, please refer to your Media5 representative.
Partial Reset
AAA service: User(s) from profile are restored with their factory password.
Where to Go From Here

The current manual offers reference information on the features that the Mediatrix unit supports.
If you plan on using the web interface configuration, go to Chapter 3 - Web Interface
Configuration on page 41.
If you plan on using the CLI configuration, go to Chapter 2 - Command Line Interface (CLI)
on page 19.
If you plan on using the SNMP configuration, go to Chapter 48 - SNMP Configuration on

page 575
Appendix B - Scripting Language on page 649 also offers a few configuration samples that can be pasted or
typed into the CLI or downloaded into the Mediatrix unit via the Configuration Script feature.
18
H A P T E R

This chapter describes how to access the CLI environment in order to perform configuration tasks.
Introduction
Configuring the CLI
Accessing the CLI
Accessing the CLI through the Serial Console Port (Mediatrix 3000 Series only)
Accessing the CLI via a Telnet Session
Accessing the CLI via a SSH Session
Working in the CLI
Contexts
Exiting from the CLI
Command Completion
Macros
History
Service Restart
Configuring the Mediatrix unit with the CLI
List of Commands / Keywords
Introduction
You can configure the Mediatrix unit parameters through a proprietary Command Line Interface (CLI)
environment. It allows you to configure the unit parameters by Serial port (Mediatrix 3000 Series only), Telnet
or SSH.
The CLI uses the Media5 proprietary scripting language as described in Appendix B - Scripting Language on
page 649.
Configuring the CLI

You must configure the CLI access. This can be done via the MIB variables. Once you have access to the CLI,
you can also use it to configure the access.
To configure the CLI access:

1.
In the cliMIB, set the inactivity expiration delay for exiting the CLI session in the
inactivityTimeOut variable.
If there is no activity during the delay defined, the CLI session is closed. This value is expressed in
minutes.
2.
Enable remote Telnet access if applicable by setting the EnableTelnet variable to Enable.
By default, Telnet is not enabled.
3.
Set the port on which the Telnet service should listen for incoming Telnet requests in the IpPort.
variable.
4.
Enable remote SSH access if applicable by setting the EnableSsh variable to Enable.
19
Chapter 2 - Command Line Interface (CLI)
5.
Accessing the CLI
Set the port on which the SSH service should listen for incoming SSH requests in the IpPort.
variable.
The configuration is loaded when it is started. It configures and starts Telnet and SSH according to the options
offered through the configuration variables. The configuration can be updated by the CLI service while running.
Partial Reset
When a partial reset is triggered, the CLI variables revert back to their default value.
Accessing the CLI

You can access the CLI through the console port of the Mediatrix unit (Mediatrix 3000 Series only) or through
a Telnet or SSH session.
Only one session at a time is allowed. These sections describe how to access the CLI:
Accessing the CLI through the Serial Console Port (Mediatrix 3000 Series only) on page 20
Accessing the CLI via a Telnet Session on page 21
Opening a Telnet Session with the Unit Manager Network on page 21
Accessing the CLI via a SSH Session on page 22
Which method you choose depends primarily on your preference and level of experience with one or all of the
options provided. None precludes using other configuration methods. Note that after performing a factory reset
or a firmware update, accessing the CLI may take up to one minute, even if the web and SNMP interfaces are
already accessible.
Note: When performing a partial reset, the root password is removed. See Partial Reset on page 15 for
more details.
Accessing the CLI through the Serial Console Port (Mediatrix 3000 Series only)
Serial console access requires a computer, serial terminal software, and null modem cable. Many networking
environments have serial terminals available or a laptop configured to act as a serial terminal setup for use
in configuration of network devices such as routers. In this case, it may be simplest to use the serial console
access.
To access the CLI through the console port:

1.
Connect a null modem cable, or crossover serial cable, to the serial console port on the front panel
of the Mediatrix unit and to a serial port on your serial terminal computer.
2.
Using a terminal software, connect by using the following parameters:
3.
9600 baud
no parity
8 data bits
1 stop bit
no flow control
If the Mediatrix unit is not on, power it up. The startup display should appear on the screen. If the
Mediatrix unit is already started, you will see this display:
Login:
4.
Type the following command:

public
Do not type a password, just press <Enter>. After you successfully connect to the Mediatrix unit
through the console port, you can start using the CLI to configure the Mediatrix unit.
20
Accessing the CLI
If the connection is unsuccessful, check that the cable is properly connected or that you have the
proper cable. Check also the connection parameters of your terminal software.
Accessing the CLI via a Telnet Session

Standards Supported
RFC 854: Telnet Protocol Specification
Connecting via Telnet requires a computer with a Telnet remote client running on a PC that acts as a Telnet
host. The Telnet host accesses the Mediatrix unit via its LAN or WAN network interface.
To access the CLI from a remote host using Telnet:

1.
Set up the Mediatrix unit as described in the Hardware Installation Guide.
2.
Power on your Mediatrix unit. Wait 60 seconds before proceeding to the next step.
3.
Open a Telnet session to the Mediatrix unit (Mediatrix 3000 Series only):
a.
You can use the Mediatrix unit WAN IP address if you know it.
b.
You can use the default Mediatrix unit LAN IP address 192.168.0.10 if the LAN interface is
enabled.
This procedure also requires some knowledge of how to configure the network settings of your
desktop PC. When the unit ships from Media5, it has been assigned a default LAN IP address
of 192.168.0.10. In order to log in to the unit across a network connection, you must use a
machine located on the 192.168.0.0 subnet, or provide a static route in the routing table of your
PC to reach the 192.168.0.0 subnet.
If you are using a Telnet port other than 23, (as configured in Configuring the CLI on page 19) you
must also specify it.
4.
Open a Telnet session to the Mediatrix unit by using one of the following IP addresses (Mediatrix
4100/4400/C7 Series):
obtained dynamically from the DHCP server
you have configured statically
after performing a partial reset (192.168.0.1)
the link-local IPv6 available and printed on the sticker under the Mediatrix unit (see
Chapter 10 - IPv4 vs. IPv6 on page 87 for more details)
If you are using a Telnet port other than 23, (as configured in Configuring the CLI on page 19) you
5.
When prompted for a login, type the following:

public
Do not type a password, just press <Enter>. After you successfully connect to the Mediatrix unit by
using Telnet, you can start using the CLI to configure the unit.
Opening a Telnet Session with the Unit Manager Network

You can use the Media5 Unit Manager Network (UMN) product to launch a Telnet client session to configure
the parameters of the Mediatrix unit. You can define which Telnet client to use in the UMN.
The Telnet session is opened from the PC where the client application is installed. It thus establishes a direct
connection to the unit. This could cause some problems if the client PC cannot directly access the unit because
of firewall restrictions, etc.
To open a Telnet session via UMN:

1.
In the UMN, autodetect the Mediatrix unit at one of the IP addresses listed in Accessing the CLI via
a Telnet Session on page 21.
Refer to the Unit Manager Network Administration Manual for more details on how to perform this
task.
21
Accessing the CLI
2.
Right-click the unit for which to open a Telnet session.
3.
Select the Open Telnet Session option in the context sensitive menu that opens.
The following window opens:
Figure 2: Telnet Session Login
This window may differ if you are not using the default Windows Telnet client.
Accessing the CLI via a SSH Session

Standards Supported
RFC 4251: The Secure Shell (SSH) Protocol Architecture
Connecting via a Secure Socket Shell (SSH) session requires a computer with a SSH or OpenSSH compatible
remote shell client running on a PC that acts as a SSH host. All communication between a client and server
is encrypted before being sent over the network, thus packet sniffers are unable to extract user names,
passwords, and other potentially sensitive data.
To access the CLI from a remote host using SSH:

1.
Set up the Mediatrix unit as described in the Hardware Installation Guide.
2.
Power on your Mediatrix unit. Wait 60 seconds before proceeding to the next step.
3.
Open a SSH session to the Mediatrix unit (Mediatrix 3000 Series only):
a.
You can use the Mediatrix unit WAN IP address if you know it.
b.
You can use the default Mediatrix unit LAN IP address 192.168.0.10 if the LAN interface is
enabled.
This procedure also requires some knowledge of how to configure the network settings of your
desktop PC. When the Mediatrix unit ships from Media5, it has been assigned a default LAN
IP address of 192.168.0.10. In order to log in to the unit across a network connection, you must
use a machine located on the 192.168.0.0 subnet, or provide a static route in the routing table
of your PC to reach the 192.168.0.0 subnet.
If you are using a SSH port other than 22, (as configured in Configuring the CLI on page 19) you
4.
Open a SSH session to the Mediatrix unit by using one of the following IP addresses (Mediatrix
4100/4400/C7 Series):
obtained dynamically from the DHCP server
you have configured statically
after performing a partial reset (192.168.0.1)
If you are using a SSH port other than 22, (as configured in Configuring the CLI on page 19) you
5.
When prompted for a login, type the following:

public
22
Working in the CLI
Do not type a password, just press <Enter>. If you are accessing the unit through the CLI for the
first time or after a factory reset, you may be presented with a warning message regarding the units
identification. You can accept the message and continue.
After you successfully connect to the Mediatrix unit by using Telnet, you can start using the CLI to
configure the Mediatrix unit.
Working in the CLI

The command interpreter interface of the CLI is a program called by the serial port (Mediatrix 3000 Series only)
Telnet or SSH client.
It allows you to browse the parameters of the unit. It also allows you to write the command lines and the CLI
interprets and executes it. The following figure illustrates the CLI in the global context after preforming a ls
command:
Figure 3: CLI Global Context
Contexts
The CLI has various contexts. A context is defined as a service name indicated by its textual key (for instance,
the Conf service). Upon entering the CLI, you are located in the Global context. This is indicated by the
following prompt:
Global>
You can change context by using the cd (change directory) command with the following syntax:
cd Service_Name
This allows you to enter into a service context. You can thus execute commands without writing the service
name. For instance:
Global>cd Conf
The prompt then changes to:

Conf>
You can use the following to get back to the global context:
Conf>cd
You can also access another service context from the Conf context:
Conf>cd Bni
23
Working in the CLI
Executing a command is different depending on if you are in the global context or a service context. See
commands on page 29 for examples.
Exiting from the CLI

To exit the CLI, type the exit command from the global context or any of the service contexts.
Command Completion
The CLI command completion function works on everything in the CLI including aliases, macros, commands,
names, etc. It is case insensitive, which means that typing interface is the same as typing Interface.
However, names that are unique are case sensitive, such as interface names.
To display all possible commands or statements, enter at least one character and press the Tab key to
complete the command line. If more than one possibility exists, they are listed and you can select the one you
want.
Lets say for instance that you type the following command in the Bni context:
Bni>Net[+ Tab key]
The CLI displays the following choices:

NetworkInterfaces
Bni>NetworkInterfaces
NetworkInterfacesStatus
Macros
Macros are internal hardcoded commands that are frequently used. The CLI currently supports the following
macros:
Table 38: Macros
Macro
Description
Reboot Reboots the unit.

Restart Restarts the services when in a service context or restarts the unit in the global context.
For instance:
Global>Mipt.Restart
You can see the list of available macros by typing the following command from anywhere in the CLI:
Conf.Macros
This will return a table similar to the following:

_____________________________
| Name
| Description
|
|__________|__________________|
| Reboot
| Reboot unit
|
| Restart | Restart service |
|__________|__________________|
History
You can recall the history commands and navigate through the history using the up and down arrows.
Services Restart
Whenever you perform changes in the configuration, this usually means that you must restart a service for the
changes to take effect. When this is the case, the following message appears in the CLI:
Need Restart
Use the Restart macro as described in Macros on page 24.
24
Working in the CLI
Syslog Messages
You can access the notifications, diagnostic traces and SIP signalling logs of the Mediatrix unit. Use the logs
on command to display Syslog traces as soon as they are sent. Use the logs off command to stop displaying
the logs.
Configuring the Mediatrix unit with the CLI

Once you are in the CLI, you can configure all the parameters of the Mediatrix unit with the various keywords
available. These keywords are described in List of Commands / Keywords on page 27. You must however
have a good understanding of the parameters structure
A good way of working with the CLI is to create the complete configuration in a text file, then copy and paste
chunks of the configuration in the CLI. This avoids to type all the commands in the CLI itself. However, be
aware that you must not copy configuration when a service needs to be restarted. You must first restart the
service before continuing.
Lets say for instance you are in the Global context and you want to see the inactivity timeout value of the CLI.
Type the following:
Global>get Cli.InactivityTimeOut
The CLI displays the current value. If you want to change this value to 10 minutes, type the following:
Global>set Cli.InactivityTimeOut=10
Refer to Appendix B - Scripting Language on page 649 for samples of configurations you can use in the CLI.
The samples include the configuration required to perform a basic call between an ISDN telephone and an
analog telephone. These samples may also be used in configuration scripts that you can download into the
Mediatrix unit.
Current Unit Status

The current unit status is displayed every time a user is authenticated by the CLI. You can also display the
same information during a session by executing the sysinfo command. The information displayed is:
System Description
Serial Number
Firmware Version
Host Name
Mac Address
System Uptime
System Time
Snmp Port
Installed Hardware Information (Name, description, location).
Welcome Message
You can define a message that is displayed when connecting to the CLI by typing the following:
Global>set Cli.WelcomeMessage=Value
Where Value is the actual message you want displayed. The following escape characters are supported:
\n for new line

\t for tab
\\ for the \ character.
Other characters are left unchanged.
Help
The CLI allows you to get help on the various keywords supported. You can have access to general or
contextual help.
You can access the general help by typing the help keyword:
25
Working in the CLI
Global>help
In that case, the CLI displays the list of all keywords available.
Figure 4: CLI Global Help
You can also access a more specific general help by typing the help keyword in a context.
Conf>help
In that case, the CLI displays the list of all keywords available as well as a description of the context.
Figure 5: CLI Global Help Variation
You can access the contextual help by typing the help keyword followed by the keyword.
Global>help set
26
Figure 6: CLI Contextual Help
See List of Commands / Keywords on page 27 for a list of keywords available.

Finally, you can access a more specific contextual help by typing the help keyword followed by the name of
the expression (scalar, table, command, column, service).
Global>help Cli.InactivityTimeOut
Figure 7: CLI Expression Help

The following sections describe the commands and keywords and their syntax depending on the context in
which you are located. Each syntax also has an example in blue.
access
Retrieves the access type of the expression. The expression
may be a variable (scalar), a table cell, or a table column.
Variables Global Context
Applies To
Services
Tables
Columns
Variables
Use this syntax when in the global context.

access Service_Name.Scalar_Name
access Cli.InactivityTimeOut
Variables Service Contexts

Use this syntax when in a service context.
access Scalar_Name
access InactivityTimeOut
Table Cell or Column Properties Global Context

Use this syntax when in the global context. The Index parameter is the first column of the table.
access Service_Name.Table_Name[Index=key].Column_Name
access Hoc.DnsServersInfo[Priority=2].IpAddress
access Service_Name.Table_Name.Column_Name
27
access Hoc.DnsServersInfo.IpAddress
Table Cell or Column Properties Service Contexts

Use this syntax when in a service context. The Index parameter is the first column of the table.
access Table_Name[Index=key].Column_Name
access DnsServersInfo[Priority=2].IpAddress
access Table_Name.Column_Name
access DnsServersInfo.IpAddress
alias / unalias
The alias function allows you to create a keyboard shortcut, an
abbreviation, a mean of avoiding typing a long command
sequence. You can assign an alias to services, scalars, tables,
and commands. You cannot currently assign an alias to
columns.
Applies To
Services
Tables
Columns
Variables
Once an alias has been added, you can use it in place of the
entity name when typing commands. You can delete an alias
with the unalias command.
Note: When naming an alias, you cannot use an existing macro name, service name, nor MIB object name
from the same context.
You can see the list of available aliases by typing the following command from anywhere in the CLI:
Conf.Alias
This will return a table similar to the following:

_______________________________________________
| Name
| Entity
| Type | Context |
|__________|___________________|______|_________|
| TimeOut | InactivityTimeOut | 200 | Cli
|
|__________|___________________|______|_________|

alias Service_Name.Scalar_Name=aliasName
alias Cli.InactivityTimeOut=timeout
unalias aliasName
unalias timeout

alias Scalar_Name = aliasName
alias InactivityTimeOut=timeout
unalias aliasName
unalias timeout
Tables or Columns Global Context

alias Service_Name.Table_Name=aliasName
alias Hoc.DnsServersInfo.IpAddress
alias Service_Name.Table_Name.Column_Name=aliasName
alias Hoc.DnsServersInfo.IpAddress = IPAddress

unalias aliasName
unalias IPAddress
28
Tables or Columns Service Contexts

alias Table_Name = aliasName
alias DnsServersInfo.IpAddress
alias Table_Name.Column_Name=aliasName
alias DnsServersInfo.IpAddress=IPAddress
unalias aliasName
unalias IPAddress
cd
Changes context (global or service context).
Applies To
Enter into a Context Global Context
Services
Tables
Columns
Variables

cd Service_Name
cd Hoc
Enter into a Context Service Contexts

cd Service_Name
cd Hoc
Get Back to the Global Context from a Service Context

cd
columnars
Retrieves the columns associated with a table.
Applies To
Table Consultation Global Context
Services
Tables
Columns
Variables

columnars Service_Name.Table_Name
columnars Hoc.DnsServersInfo
Table Consultation Service Contexts

columnars Table_Name
columnars DnsServersInfo
commands
Retrieves the commands associated with a service or a table.
Applies To
Service Consultation Global Context
Services
Tables
Columns
Variables

commands Service_Name
commands Bni
Service Consultation Service Contexts

29

commands

commands Service_Name.Table_Name
commands Hoc.DnsServersInfo

commands Table_Name
commands DnsServersInfo
defval
Retrieves the default value of the expression. The expression
may be a variable (scalar), a table column, or a table cell).
Applies To
Services
Tables
Columns
Variables

defval Service_Name.Scalar_Name
defval Cli.InactivityTimeOut

defval Scalar_Name
defval InactivityTimeOut
Cell or Column Properties Global Context

defval Service_Name.Table_Name[Index=key].Column_Name
defval Hoc.DnsServersInfo[Priority=2].IpAddress
defval Service_Name.Table_Name.Column_Name
defval Hoc.DnsServersInfo.IpAddress
Cell or Column Properties Service Contexts

defval Table_Name[Index=key].Column_Name
defval DnsServersInfo[Priority=2].IpAddress
defval Table_Name.Column_Name
defval DnsServersInfo.IpAddress
dump
Displays the units whole configuration on screen.
dump
Applies To
Services
30
Tables
Columns
Variables
get
Retrieves the value of the expression. The expression may be a
variable (scalar), a table, a table row, a table column, or a table
cell). Note that entering the get command is optional.
Applies To
Services
Tables
Columns
Variables
Variable Consultation Global Context

Service_Name.Scalar_Name
Cli.InactivityTimeOut
get Service_Name.Scalar_Name
get Cli.InactivityTimeOut
Variable Consultation Service Contexts

Scalar_Name
InactivityTimeout
get Scalar_Name
get InactivityTimeOut

Service_Name.Table_Name
Hoc.DnsServersInfo
get Service_Name.Table_Name
get Hoc.DnsServersInfo

Table_Name
DnsServersInfo
get Table_Name
get DnsServersInfo
Column Consultation Global Context

Service_Name.Table_Name.Column_Name
Hoc.DnsServersInfo.Priority
get Service_Name.Table_Name.Column_Name
get Hoc.DnsServersInfo.Priority
Column Consultation Service Contexts

Table_Name.Column_Name
DnsServersInfo.Priority
get Table_Name.Column_Name
get DnsServersInfo.Priority
Row Consultation Global Context

Service_Name.Table_Name[Index=key]
Hoc.DnsServersInfo[Priority=2]
get Service_Name.Table_Name[Index=key]
get Hoc.DnsServersInfo[Priority=2]
31
Row Consultation Service Contexts

Table_Name[Index=key]
DnsServersInfo[Priority=2]
get Table_Name[Index=key]
get DnsServersInfo[Priority=2]
Cell Consultation Global Context

Service_Name.Table_Name[Index=key].Column_Name
Hoc.DnsServersInfo[Priority=2].IpAddress
get Service_Name.Table_Name[Index=key].Column_Name
get Hoc.DnsServersInfo[Priority=2].IpAddress
Cell Consultation Service Contexts

Table_Name[Index=key].Column_Name
DnsServersInfo[Priority=2].IpAddress
get Table_Name[Index=key].Column_Name
get DnsServersInfo[Priority=2].IpAddress
help
Retrieves the documentation related to the expression. This
keyword is case sentitive.
You can have access to general or contextual help. You can
access the general help by typing the help keyword. You can
access the contextual help by typing the help keyword followed
by the name of the expression.
Applies To
Services
Tables
Columns
Variables

help Service_Name
help Bni

help

help Service_Name.Table_Name
help Hoc.DnsServersInfo
help Service_Name.Table_Name.Column_Name
help Hoc.DnsServersInfo.IpAddress

help Table_Name
help DnsServersInfo
help Table_Name.Column_Name
32
help DnsServersInfo.IpAddress
Commands Global Context

help Service_Name.Command
help Scm.LockConfig
Commands Service Contexts

help Command
help LockConfig
indexes
Retrieves the indexes associated with the expression of a table.
The expression may be the table itself or one of its columns.
Applies To
Services
Tables
Columns
Variables

indexes Service_Name.Table_Name
indexes Hoc.DnsServersInfo

indexes Table_Name
indexes DnsServersInfo

indexes Service_Name.Table_Name.Column_Name
indexes Hoc.DnsServersInfo.IpAddress

indexes Table_Name.Column_Name
indexes DnsServersInfo.IpAddress
keys
Retrieves the keys associated with the expression of a table. The
expression may be the table itself or one of its columns.
Applies To
Services
Tables
Columns
Variables

keys Service_Name.Table_Name
keys Hoc.DnsServersInfo

keys Table_Name
keys DnsServersInfo
33

keys Service_Name.Table_Name.Column_Name
keys Hoc.DnsServersInfo.IpAddress

keys Table_Name.Column_Name
keys DnsServersInfo.IpAddress
logs off
Stops to display of Syslog traces.
logs off
Applies To
Services
Tables
Columns
Variables
logs on
Displays Syslog traces as soon as they are sent.
The traces displayed are the notifications coming from the
services, the diagnostic traces and the Signaling Logs.
logs on
Applies To
Services
Tables
Columns
Variables
ls
Retrieves the list of services available in a global context and the
objects of the service on a service context.
Applies To
Services
Tables
Columns
Variables

ls Service_Name
ls Bni

ls

ls Service_Name.Table_Name
ls Hoc.DnsServersInfo

ls Table_Name
ls DnsServersInfo
34
name
Retrieves the name of the expression. This keyword is case
sensitive. You must type the exact name after the name key.
Variables Configuration Global Context
Applies To
Services
Tables
Columns
Variables

name Service_Name.Scalar_Name
name Cli.InactivityTimeOut
Variables Configuration Service Contexts

name Scalar_Name
name InactivityTimeOut
Table Configuration Global Context

name Service_Name.Table_Name
name Hoc.DnsServersInfo
name Service_Name.Table_Name.Column_Name
name Hoc.DnsServersInfo.IpAddress
Table Configuration Service Contexts

name Table_Name
name DnsServersInfo
name Table_Name.Column_Name
name DnsServersInfo.IpAddress
Command Execution Global Context

name Service_Name.Command
name Scm.LockConfig
Command Execution Service Contexts

name Command
name LockConfig
objects
Retrieves the objects associated with the expression of a
service.
Applies To
Services
Tables
Columns
Variables

objects Service_Name
objects Bni
35

objects
Table consultation Global Context

objects Service_Name.Table_Name
objects Hoc.DnsServersInfo
Table consultation Service Contexts

objects Table_Name
objects DnsServersInfo
PCapture
Starts a network capture. Typing ctrl+c stops immediately a running capture command and displays statistics.
Supported parameters can be found by typing "help pcapture".
The Telnet and SSH ports are automatically filtered out. The host addresses are not converted to names to
avoid DNS lookups. The protocol and port numbers are not converted to names either.
Use this syntax.
pcapture [options] [expression]
pcapture -raw -c 50 port 161
Options:
-c 'count': Exit after receiving 'count' packets.
-e: Print the link-level header on each dump line.
-p: Don't put the interface into promiscuous mode.
-raw: Raw packets are output (unreadable output, must be redirected to file or Wireshark)
-D: Print the list of network interfaces available on the system and on which pcapture can
capture packets.
-i 'if': Listen on interface 'if'. Can be any of the interfaces returned by option -D or can be set to
'any'. 'any' will listen on all interfaces but not in promiscuous mode.
-S: Print absolute, rather than relative, TCP sequence numbers.
-T 'expression': Force packets selected by 'expression' to be interpreted of the specified type.
Supported types are rtp, rtcp, snmp, tftp.
Expression:
Selects which packets will be dumped. If no expression is given, all packets on the net will
be dumped. Otherwise, only packets for which expression is 'true' will be dumped. For the
expression syntax, see pcap-filter(7).
It is possible to route the capture to Wireshark to have a remote live capture. From the remote PC (Windows
or Linux), type the following command:
plink.exe -pw "" public@10.4.127.128 "pcapture -raw port 161" | wireshark -k -i -
This example connects by using plink (from putty) in SSH to the unit 10.4.127.128 by using the username
"public" and an empty password. It would capture the SNMP packets.
For more information in the pcapture command, please refer to the following page: http://www.tcpdump.org/
pcap3_man.html.
ping (IPv4)
Executes a ping command using IPv4 with the arguments and the target host provided by the user.
Use this syntax when using the ping command:
ping [-c COUNT -s SIZE -q] host_name
ping -c 3 -s 300 -q 192.168.0.25
36
The supported ping arguments are:
-c COUNT: Stops the ping after it has sent COUNT packets.

-s SIZE: Sends SIZE data bytes in packets (default = 56).
-q: Shows information only at the start and when finished.
Typing Ctrl+c immediately stops a running ping command and displays statistics.
ping (IPv6)
Executes a ping command using IPv6 with the arguments and the target host provided by the user.
Use this syntax when using the ping command:
ping [-c COUNT -s SIZE -q] host_name
ping -c 3 -s 300 -q 192.168.0.25
The supported ping arguments are:
-c COUNT: Stops the ping after it has sent COUNT packets.

-s SIZE: Sends SIZE data bytes in packets (default = 56).
-q: Shows information only at the start and when finished.
Typing Ctrl+c immediately stops a running ping command and displays statistics.
scalars
Retrieves the scalars associated with the expression of a
service.
Applies To
Services
Tables
Columns
Variables

scalars Service_Name
scalars Bni

scalars
set
Assigns a constant value to the expression. The expression may
be a variable (scalar) or a table cell.
Applies To
Services
Tables
Columns
Variables

Service_Name.Scalar_Name = constant
Cli.InactivityTimeOut = 25
set Service_Name.Scalar_Name = constant
set Cli.InactivityTimeOut = 25

Scalar_Name = constant
InactivityTimeOut = 25
set Scalar_Name = constant
set InactivityTimeOut = 25
Cell Configuration Global Context
37
Service_Name.Table_Name[Index=key].Column_Name=Value
Hoc.DnsServersInfo[Priority=2].IpAddress=192.168.0.10
set Service_Name.Table_Name[Index=key].Column_Name=Value
set Hoc.DnsServersInfo[Priority=2].IpAddress=192.168.0.10
Cell configuration Service Contexts

Table_Name[Index=key].Column_Name=Value
DnsServersInfo[Priority=2].IpAddress=192.168.0.10
set Table_Name[Index=key].Column_Name=Value
set DnsServersInfo[Priority=2].IpAddress=192.168.0.10
show
Retrieves the value of the expression.
Applies To
Variable Consultation Global Context
Services
Tables
Columns
Variables

show Service_Name.Scalar_Name
show Cli.InactivityTimeOut
Variable Consultation Service Contexts

show Scalar_Name
show InactivityTimeOut

show Service_Name.Table_Name
show Hoc.DnsServersInfo

show Table_Name
show DnsServersInfo

show Service_Name.Table_Name.Column_Name
show Hoc.DnsServersInfo.IpAddress

show Table_Name.Column_Name
show DnsServersInfo.IpAddress
Row Consultation Global Context

show Service_Name.Table_Name[Index=key]
show Hoc.DnsServersInfo[Priority=2]
38
Row Consultation Service Contexts

show Table_Name[Index=key]
show DnsServersInfo[Priority=2]
Cell Consultation Global Context

show Service_Name.Table_Name[Index=key].Column_Name
show Hoc.DnsServersInfo[Priority=2].IpAddress
Cell Consultation Service Contexts

show Table_Name[Index=key].Column_Name
show DnsServersInfo[Priority=2].IpAddress
sysinfo
Displays the current unit status. The information displayed is:
System Description
Serial Number
Firmware Version
Host Name
Mac Address
System Uptime
System Time
Snmp Port
Installed Hardware Information (Name, description, location).
tables
Retrieves the tables associated with the a service.
Applies To
Services
Tables
Columns
Variables

tables Service_Name
tables Bni

tables
type
Retrieves the type of the data of the expression. The expression
may be a variable (scalar), a table column, or a table cell.
Applies To
Services
Tables
Columns
Variables

type Service_Name.Scalar_Name
type Cli.InactivityTimeOut
39

type Scalar_Name
type InactivityTimeOut
Cell or Column Properties Global Context

type Service_Name.Table_Name[Index=key].Column_Name
type Hoc.DnsServersInfo[Priority=2].IpAddress
type Service_Name.Table_Name.Column_Name
type Hoc.DnsServersInfo.IpAddress
Cell or Column Properties Service Contexts

type Table_Name[Index=key].Column_Name
type DnsServersInfo[Priority=2].IpAddress
type Table_Name.Column_Name
type DnsServersInfo.IpAddress
Command Execution
This section describes the syntax to use to execute a MIB command.
Global Context
Service_Name.Command arg1=value1 -b arg2=[value2 value3 value4]
SipEp.InsertGateway Name=test
Service Context
Command arg1=value1 -b arg2=[value2 value3 value4]
InsertGateway Name=test
40
H A P T E R
Web Interface Configuration

The Mediatrix unit contains an embedded web server to set parameters by using the HTTP or HTTPS protocol.
Standards Supported
RFC 1945: Hypertext Transfer Protocol - HTTP/1.0
RFC 2616: Hypertext Transfer protocol - HTTP/1.1.
This chapter describes the following:
Introduction to the Mediatrix unit web pages.

How to access the web interface and description of the various menus available.
How to submit changes.
Introduction
The web interface may be used to:
View the status of the Mediatrix unit.

Set the uplink parameters of the Mediatrix unit.
Perform a firmware update, configuration scripts download, or configuration backup/restore.
Set numerous parameters of the Mediatrix unit.
All of the parameters in the web interface may also be configured via SNMP. See Chapter 48 - SNMP
Configuration on page 575 for more details.
To configure the web-based configuration service:

1.
In the webMIB, locate the serverGroup folder.
2.
Define the HTTP mode(s) to which the Web server should listen in the httpMode variable.
web.httpMode="Value"
where Value may be as follows:

Table 39: HTTP Modes
Value
Mode
Description
100
Secure
The Web server only accepts requests using HTTPS. Requests using HTTP
are ignored. This is the default value.
200
Unsecure The Web server only accepts requests using HTTP. Requests using HTTPS
are ignored.
300
Both
The Web server accepts requests using HTTP or HTTPS.
If you are using HTTPS (either in Secure mode or Both mode), the web server needs a valid
server certificate with server authentication extended key usage installed on the Mediatrix unit.
See Chapter 47 - Certificates Management on page 569 for more details.
Accessing the web pages via HTTPS adds additional delay since encryption is used. To access the
unit via HTTPS, your browser must support RFC 2246 (TLS 1.0).
Note that the web server does not listen to the configured modes when the management interface
is down or a configuration error occurred (e.g., missing or invalid certificate for HTTPS mode) while
setting up the web server.
41
Chapter 3 - Web Interface Configuration
3.
Introduction
Set the TCP port on which the web service listens for HTTP requests in the serverPort variable.
web.serverPort="Value"
4.
Set the port on which the web service listens for HTTPS requests in the secureServerPort
variable.
web.secureServerPort="Value"
5.
Define the allowed cipher suites for the network security settings to which the Web server should
listen when using the HTTPS mode in the httpsCipherSuite variable.
Any connection attempts to the web server using a cipher that is not allowed by the cipher suite will
result in a failure to establish the connection.
web.httpsCipherSuite="Value"

Table 40: HTTPS Cipher Suite Values and Parameters
Value Parameter
100
200
42
CS1
CS2
Description
The Web server only accepts requests using cipher suites:
TLS_DHE_RSA_WITH_AES_256_CBC_SHA
TLS_DHE_DSS_WITH_AES_256_CBC_SHA
TLS_RSA_WITH_AES_256_CBC_SHA
TLS_DHE_RSA_WITH_3DES_EDE_CBC_SHA
TLS_DHE_DSS_WITH_3DES_EDE_CBC_SHA
TLS_RSA_WITH_3DES_EDE_CBC_SHA
TLS_RSA_WITH_RC4_128_SHA
TLS_RSA_WITH_RC4_128_MD5
TLS_DHE_RSA_WITH_DES_CBC_SHA
TLS_RSA_WITH_DES_CBC_SHA
TLS_DHE_RSA_EXPORT_WITH_DES40_CBC_SHA
TLS_DHE_DSS_EXPORT_WITH_DES40_CBC_SHA
TLS_RSA_EXPORT_WITH_DES40_CBC_SHA
TLS_RSA_EXPORT_WITH_RC4_40_MD5
This represents a secure configuration using SHA-1. Web server only

accepts requests using cipher suites:
Introduction
Table 40: HTTPS Cipher Suite Values and Parameters (Continued)

Value Parameter
300
CS3
Description
This represents the most secure configuration using SHA-2. Only the most
secure cipher suites are allowed when using this value.
TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384
TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA384
TLS_DHE_RSA_WITH_AES_256_GCM_SHA384
TLS_DHE_RSA_WITH_AES_256_CBC_SHA256
TLS_ECDH_RSA_WITH_AES_256_GCM_SHA384
TLS_ECDH_RSA_WITH_AES_256_CBC_SHA384
TLS_RSA_WITH_AES_256_GCM_SHA384
TLS_RSA_WITH_AES_256_CBC_SHA256
Table 41: Cipher Suites Configuration Values

Value
Meaning
100
CS1
200
CS2
300
CS3
43
Introduction
HTTP User-Agent Header Format


by using the CLI
You can define the text to display in the HTTP User-Agent header. You can use macros to include information
specific to the unit.
You can also define the same information in the SIP User-Agent header. See SIP User-Agent Header Format
on page 334 for more details.
To set the HTTP User-Agent header format:

1.
In the hocMIB, set the HTTP User-Agent header format in the httpUaHeaderFormat variable.
hoc.httpUaHeaderFormat="Value"
where Value may contain any text, as well as one or more of the following macros:
Table 42: Macros Supported
Macro
Description
%version%
Application version.
%mac%
MAC address.
%product% Product name.

%profile%
Profile.
%%
Insert the % character.
For instance, the default value is:

%product%/v%version% %profile%
44
Using the Web Interface

Media5 recommends that you use the latest version of the Microsoft
properly access the web interface.
Internet Explorer web browser to
To use the web interface configuration:

1.
In your web browsers address field, type the IP address of the Mediatrix unit LAN interface (if you
have performed a partial reset, this is 192.168.0.10).
Figure 8: Login Window
2.
Enter the proper user name and password.

The user name and password are case sensitive hence they must be entered properly. Default
factory values are:
User Name: admin
Password: administrator
You can also enter the user name public and no password.
3.
Click Login.
The Information web page displays. It stays accessible for as long as the Internet browser used to
access the Mediatrix unit web interface is opened.
Figure 9: Information Web Page
45
The Installed Hardware section (Mediatrix 3000 Series only) lists the cards installed in the Mediatrix
unit:
Table 43: Hardware Codes Description
Card Number
Mediatrix 3301-010
Quantity Description
1
FXS/FXO
card
Mediatrix 3308, Mediatrix 3716, Mediatrix 3731,

or Mediatrix 3741
FXS/FXO
card
Mediatrix 3316
E1/T1 card

Mediatrix 3731, or Mediatrix 3734
E1/T1 card
Mediatrix 3532 or Mediatrix 3632
BRI card
Mediatrix 3404, Mediatrix 3734, or Mediatrix

3741
BRI card
Mediatrix 3408
2 FXS/6
FXO card

or Mediatrix 3742
2 FXS/6
FXO card
Mediatrix 3216
Mediatrix 3301-020
Mediatrix 3301-060
Mediatrix 3301-080
4.
Products
Click Sign Out to end your Mediatrix web session.

The Login Window web page displays.
Menu Items
The Menu frame is displayed at the top of the browser window. It contains management links that allow you
to display web pages in the Content frame. The management links available vary depending on the Mediatrix
unit you are using.
Table 44: Menu Frame Links
Link
Description
Information: Displays, in read-only format, the status of the Mediatrix unit.
Services: Allows you to start/stop the services running on the Mediatrix unit. See
Chapter 4 - Services on page 55 for more details.
Hardware: Allows you to configure the hardware installed on the Mediatrix unit.
See Chapter 5 - Hardware Parameters on page 61 for more details.
System
Endpoints: Allows you to configure the administrative state of the Mediatrix units
ports. See Chapter 6 - Endpoints State Configuration on page 69 for more
details.
Syslog: Allows you to configure the Mediatrix unit to properly handle syslog
messages and notification messages. See Chapter 7 - Syslog Configuration on
page 73 for more details.
Events: Allows you to associate a NOTIFICATION message and how to send it
(via syslog or via a SIP NOTIFY packet). See Chapter 8 - Events Configuration
Local Log: Displays local log status and entries of your Mediatrix unit. See
Chapter 9 - Local Log on page 83 for more details.
46
Table 44: Menu Frame Links (Continued)

Link
Description
Status: Displays, in read-only format, the network parameters status of the
Mediatrix unit.
Host: Allows you to configure the host parameters of the Mediatrix unit. See
Chapter 11 - Host Parameters on page 91 for more details.
Interfaces: Allows you to configure the uplink information used by the Mediatrix
unit. See Chapter 12 - Interface Parameters on page 101 for more details.
VLAN: Allows you to create and manage VLANs on the Mediatrix unit. See
Chapter 13 - VLAN Parameters on page 115 for more details.
QoS: Allows you to configure packets tagging sent from the Mediatrix unit. See
Chapter 14 - Local QoS (Quality of Service) Configuration on page 117 for more
details.
Network
Local Firewall: Allows you to configure the local firewall service of the Mediatrix
unit. See Chapter 15 - Local Firewall Configuration on page 123 for more
details.
IP Routing: Allows you to configure the IP routing parameters of the Mediatrix
unit. See Chapter 16 - IP Routing Configuration on page 129 for more details.
Network Firewall: Allows you to configure the network firewall service of the
Mediatrix unit. See Chapter 17 - Network Firewall Configuration on page 137 for
more details.
NAT: Allows you to configure the NAT service of the Mediatrix unit. See Chapter
18 - NAT Configuration on page 143 for more details.
DHCP Server: Allows you to configure the the embedded DHCP server of the
Mediatrix unit. See Chapter 19 - DHCP Server Settings on page 151 for more
details.
Status: Allows you to view the status of the Mediatrix unit POTS parameters. See
Chapter 20 - POTS Configuration on page 161 for more details.
POTS
Config: Allows you to configure the POTS parameters of the Mediatrix unit. See
Chapter 20 - POTS Configuration on page 161 for more details.
FXS Config: Allows you to configure the FXS parameters of the Mediatrix unit.
See Chapter 20 - POTS Configuration on page 161 for more details.
FXO Config: Allows you to configure the FXO parameters of the Mediatrix unit.
See Chapter 20 - POTS Configuration on page 161 for more details.
47

Link
Description
Status: Allows you to view the status of the Mediatrix unit ISDN parameters. See
Chapter 21 - ISDN Configuration on page 179 for more details.
Primary Rate Interface: Allows you to configure the PRI ISDN parameters of the
Mediatrix unit. See PRI Configuration on page 186 for more details.
Basic Rate Interface: Allows you to configure the BRI ISDN parameters of the
Mediatrix unit. See BRI Configuration on page 197 for more details.
ISDN
Interop: Allows you to set parameters to properly work, communicate, or connect

with specific ISDN devices. See Interop Parameters Configuration on page 208
for more details.
Timer: Allows you to set timer parameters. See ISDN Timers Configuration on
Services: Allows you to set the ISDN optional services. See Services
IP Sync: Allows you to configure the Mediatrix unit in order to control the IP media
synchronization using clock reference signals sent over IP. See IP Sync
Status: Allows you to view the status of the Mediatrix unit R2 parameters. See
Chapter 22 - R2 CAS Configuration on page 223 for more details.
R2: Allows you to configure the parameters related to R2. See R2 Channel
Associated Signaling on page 226 for more details.
Signaling: Allows you to configure the R2 signaling parameters of the Mediatrix
unit. See R2 Signaling Variants on page 230 for more details.
Timers: Allows you to configure the R2 timers of the Mediatrix unit. See R2
Timers Variants on page 234 for more details.
R2
Digit Timers: Allows you to configure the R2 digit timers of the Mediatrix unit. See
R2 Digit Timers Variants on page 238 for more details.
Link Timers: Allows you to configure the R2 link timers of the Mediatrix unit. See
R2 Link Timers Variants on page 240 for more details.
Tones: Allows you to configure the R2 CAS tones of the Mediatrix unit. See R2
Tones Variants on page 242 for more details.
48

Link
Description
Status: Allows you to view the status of the Mediatrix unit E&M parameters. See
Chapter 23 - E&M CAS Configuration on page 255 for more details.
R2: Allows you to configure the parameters related to E&M. See E&M Channel
Associated Signaling on page 258 for more details.
Signaling: Allows you to configure the E&M signaling parameters of the Mediatrix
unit. See E&M Signalling Variants on page 262 for more details.
Timers: Allows you to configure the E&M timers of the Mediatrix unit. See E&M
Timers Variants on page 265 for more details.
E&M
Digit Timers: Allows you to configure the E&M digit timers of the Mediatrix unit.
See E&M Digit Timers Variants on page 267 for more details.
Link Timers: Allows you to configure the E&M link timers of the Mediatrix unit. See
E&M Link Timers Variants on page 269 for more details.
Tones: Allows you to configure the E&M CAS tones of the Mediatrix unit. See
E&M Tones Variants on page 270 for more details.
Gateways: Allows you to add and remove SIP gateways in the Mediatrix unit.
See Chapter 24 - SIP Gateways on page 279 for more details.
Servers: Allows you to configure the SIP server and SIP user agent parameters
of the Mediatrix unit. See Chapter 25 - SIP Servers on page 283 for more
details.
Registrations: Allows you to configure the registration parameters of the
Mediatrix unit. See Chapter 26 - SIP Registration on page 291 for more details.
SIP
Authentication: Allows you to configure authentication parameters of the

Mediatrix unit. See Chapter 27 - SIP Authentication on page 303 for more
details.
Transport: Allows you to configure the SIP transport parameters of the Mediatrix
unit. See Chapter 28 - SIP Transport Parameters on page 307 for more details.
Interop: Allows you to configure the SIP interop parameters of the Mediatrix unit.
See Chapter 29 - Interop Parameters on page 315 for more details.
Misc: Allows you to configure interoperability features of the Mediatrix unit. See
Chapter 30 - Miscellaneous SIP Parameters on page 337 for more details.
Codecs: Allows you to configure the voice and data codec related parameters of
the Mediatrix unit. See Chapter 31 - Voice & Fax Codecs Configuration on
Media
Security: Allows you to properly configure the security parameters of the

Mediatrix unit. See Chapter 32 - Security on page 381 for more details.
RTP Stats: Allows you to read and configure the RTP statistics collected by the
Mediatrix unit. See Chapter 33 - RTP Statistics Configuration on page 385 for
more details.
Misc: Allows you to configure parameters that apply to all codecs. See Chapter
34 - Miscellaneous Media Parameters on page 391 for more details.
49

Link
Description
DTMF Maps: Allows you to configure the various DTMF maps of the Mediatrix
unit. See Chapter 35 - DTMF Maps Configuration on page 405 for more details.
Call Forward: Allows you to configure three types of Call Forward. See Chapter
36 - Call Forward Configuration on page 413 for more details.
Services: Allows you to configure the Mediatrix unit subscriber services. See
Chapter 37 - Telephony Services Configuration on page 421 for more details.
Tone Customization: Allows you to override the pattern for a specific tone
defined for the selected country. See Chapter 38 - Tone Customization
Parameters Configuration on page 447 for more details.
Telephony
Music on Hold: Allows you to configure the Music on Hold service of the
Mediatrix unit. See Chapter 39 - Music on Hold Parameters Configuration on
Misc: Allows you to configure the country in which the Mediatrix unit is located.
See Chapter 40 - Country Parameters Configuration on page 455 for more
details.
Status: Allows you to view the current status of the call routing service. See
Chapter 42 - Call Router Configuration on page 467 for more details.
Call Router
Route Config: Allows you to configure the call routing service of the Mediatrix
unit. See Chapter 42 - Call Router Configuration on page 467 for more details.
Auto-routing: Allows you to configure the auto-routing feature of the Mediatrix
unit. See Chapter 43 - Auto-Routing Configuration on page 523 for more details.
Configuration Scripts: Allows you to configure the various configuration scripts
parameters of the Mediatrix unit. See Chapter 44 - Configuration Script on
Backup / Restore: Allows you to configure how to backup and restore the
Mediatrix units configuration. See Chapter 45 - Configuration Backup/Restore
Firmware Upgrade: Allows you to configure the various firmware upgrade
parameters of the Mediatrix unit. See Chapter 46 - Firmware Download on
Certificates: Allows you to add and delete security certificates in the Mediatrix
unit. See Chapter 47 - Certificates Management on page 569 for more details.
Management
SNMP: Allows you to configure the SNMP privacy parameters of the Mediatrix
unit. See Chapter 48 - SNMP Configuration on page 575 for more details.
CWMP: Allows you to configure the CWMP parameters of the Mediatrix unit. See
Chapter 49 - CWMP Configuration on page 581 for more details.
Access Control: Allows you to set the Access Control parameters of the
Mediatrix unit. . See Chapter 50 - Access Control Configuration on page 607 for
more details.
File:Allows you to use the units File Manager. See File Manager on page 613
for more details.
Misc: Allows you to set various parameters used to manage the Mediatrix unit.
See Chapter 52 - Miscellaneous on page 617 for more details.
Reboot
50
Allows you to restart the Mediatrix unit.
Submitting Changes
Submitting Changes
When you perform changes in the web interface and click the Submit button, the Mediatrix unit validates the
changes. A message is displayed next to any invalid value. A message is also displayed if a service must be
restarted and a link is displayed at the top of the page. This link brings you to the Services page. In this page,
each service that requires to be restarted has a * beside its name. See Chapter 4 - Services on page 55
for more details.
If you are not able to restart one or more services, click the Reboot link in the top menu. The Reboot page then
opens. You must click Reboot. This restarts the Mediatrix unit. If the unit is in use when you click Reboot, all
calls are terminated.
Where to Go From Here?

If you want to configure the Mediatrix unit to perform a basic call, this usually involves the following:
Table 45: Basic Call Configuration Steps
Action
Description
Configuring the POTS

parameters (Mediatrix
3208 / 3216 / 3308 /
3316 / 3716 / 3731 /
3732 / 3741 / 3742
LP16 / LP24 / 4100 /
C7 Series)
You must minimally configure the FXS and FXO

interfaces so that they can send and receive
calls.
Chapter 20 - POTS
Configuring the ISDN

Interface (Mediatrix
3000 / 4400 Series)
You must minimally configure the ISDN

interface so that it can send and receive calls.
This means setting the Endpoint Type and
Connection Type (BRI models) parameters.
Chapter 21 - ISDN
Configuring the ISDN

power feeding feature
(Mediatrix 4400
Series)
If you have a Mediatrix unit plus model, you may

enable the ISDN power feeding feature to power
your ISDN telephone.
BRI Power Feeding

Configuring the SIP

Endpoint
Configuring the SIP endpoint allows you to

register your ISDN telephone or FXS interfaces
to a SIP server. This includes setting the
following parameters:
Chapter 25 - SIP Servers on

page 283
Registrar Server Host
Proxy Home Domain Host
User Name
Friendly Name
Gateway Name
Where to?
Chapter 26 - SIP Registration

on page 291
Chapter 24 - SIP Gateways on
page 279
Configuring the Call

Router with Routes
You must create routes that will route calls from

ISDN / FXS /FXO to SIP and from SIP to ISDN /
FXS / FXO.
Routes on page 485
Configuration of the
Call Router: Mapping
You must create mappings that will allow you to

properly communicate from ISDN / FXS / FXO
to SIP and from SIP to ISDN / FXS / FXO.
Mappings on page 491
51
Where to Go From Here?
Using Secure Communication

The Mediatrix unit allows you to use a secure communication whenever required. You must set the Mediatrix
unit with security parameters:
Table 46: Secure Communication Steps
Step
5. Transfer a valid CA certificate into the
Mediatrix unit.
Chapter 47 - Certificates Management on

page 569
6. Use secure signalling by enabling the TLS

transport protocol.
Chapter 28 - SIP Transport Parameters on

page 307
7. Use secure media by:
Defining the SRTP/ SRTCP

base port.
Base Ports Configuration on page 401
52
Where to?
Setting the RTP secure mode

to Secure or Secure with
fallback.
Security Parameters on page 381
System Parameters
Page Left Intentionally Blank
H A P T E R
Services
This chapter describes how to view and start/stop system and network parameters of the Mediatrix unit.
Services Table
The Mediatrix unit uses many services grouped in two classes: system and user. You can perform service
commands on user services, but not the system services.
Whenever you perform changes in the various sections of the web interfaces, this usually means that you must
restart a service for the changes to take effect. When a service needs to be restarted, it is displayed in bold
and the message Restart needed is displayed in the Comment column.
If you are not able to restart a service because it is a system service, click the Reboot link in the top menu.
The Reboot page then opens. You must click Reboot. This restarts the Mediatrix unit. If the unit is in use when
you click Reboot, all calls are terminated.
To manage the Mediatrix unit services:

1.
In the web interface, click the System link, then the Services sub-link.
Figure 10: System Services Web Page
55
Chapter 4 - Services
Services Table
The following are the services available. Note that the services available may differ depending on
the Mediatrix unit you are using.
Table 46: Mediatrix unit Services
Service
Description
System Services
Authentication, Authorization
and Accounting (AAA)
Authenticates a user and grants rights to perform specific tasks

on the system.
Certificate Manager (CERT)
Manages certificate files and provides access to these

certificates.
Configuration Manager
(CONF)
Responsible of configuration scripts transfers, as well as

configuration image upload/download for backup/restore of the
unit configuration.
Device Control Manager

(DCM)
Auto-detects and identifies the hardware components of the unit.
Ethernet Manager (ETH)
Configures the system's Ethernet ports parameters.
File Manager (FILE)
Manages the files created with the File transfer protocol.
Firmware Pack Updater

(FPU)
Handles firmware upgrade and downgrade operations.
Host Configuration (HOC)
Configures network parameters that apply to the Mediatrix unit

(not to a specific interface).
Local Quality Of Service

(LQOS)
Configures the packets tagging sent from the Mediatrix unit.
Process Control Manager

(PCM)
Responsible to boot and restart the unit.
Service Controller Manager

(SCM)
Responsible to:
Manage services information.
Offer proxy functionality for service interoperation.

User Services
Basic Network Interface

(BNI)
Configures the IP address and network mask for the Uplink and
LAN1 networks.
Call Routing (CROUT)
Routes calls between interfaces.
Call Detail Record (CDR)
56
Command Line Interface

(CLI)
Allows you user to configure the unit parameters by Serial port

(Mediatrix 3000 Series only), Telnet or SSH.
CPE WAN Management

Protocol (CWMP)
Manages the support of the TR-069 protocol for auto

provisioning.
DHCP Server (Dhcp)
Allows the user to lease IP addresses and send network

configuration to hosts located on any network.
Endpoint Administration
(EpAdm)
Holds basic administration and status at endpoint and unit level.
Endpoint Services (EpServ)
Manages endpoint behaviour and holds configuration

parameters related to endpoints (such as DTMF maps,
telephony services, etc.).
IP Routing (IpRouting)
Allows the user to configure the unit's routing table.
Services Table
Table 46: Mediatrix unit Services (Continued)

Service
2.
Description
IP Synchronization (IpSync)
Controls the IP media synchronization using clock reference

signals sent over IP.
Integrated Services Digital

Network (ISDN)
Configures the Integrated Services Digital Network (ISDN) Basic

Rate Interfaces (BRI) or Primary Rate Interfaces (PRI)
Local Firewall (LFW)
Allows you to filter incoming packets whose final destination is

the unit.
Link Layer Discovery

Protocol (Lldp):
Used by network devices for advertising their identity,

capabilities, and neighbors on a IEEE 802 local area network,
usually wired Ethernet.
Media IP Transport (MIPT)
Holds basic configuration parameters (such as voice/data

codec) and implements basic functionality related to media
stream.
Music on Hold (MOH)
Allows you to configure the Music on Hold parameters.
Network Address Translation

(Nat)
Allows the user to change the source or destination address/port

of a packet.
Network Firewall (Nfw)
Allows the user to filter forwarded packets.
Notifications and Logging

Manager (NLM)
Handles syslog messages and notification messages.
Network Traffic Control (Ntc)
Controls the bandwidth limitation applied to physical network

interfaces.
Plain Old Telephony System

Lines service (POTS)
Holds basic configuration parameters (such as DTMF dialing

delays) and implements basic functionality related to POTS lines
(such as enabling/disabling individual lines).
SIP Endpoint (SipEp)
Manages the behaviour of the system regarding SIP.
SNMP (SNMP)
Accesses internal variables through an SNMP client. It also

handles user authentication.
Telephony Interface (TELIF)
Configures the basic specification of each telephony interface.
Web (WEB)
Allows accessing the unit through web pages, using HTTP.
In the User Service section, select the service startup type of a service in the Startup Type column.
Table 47: Startup Types
Type
Auto
Description
The service is automatically started when the system starts.
Manual The administrator must manually start the service.

You can put only user services in manual startup type. Proceed with caution when setting services
to manual because this could prevent you from successfully contacting the unit.
3.
Select if you want to perform service commands on one or more services in the Action column.
Table 48: Actions
Action
Description
Starts the service.
57
Services Table
Table 48: Actions

Action
Description
Stops the service.
Restarts the service.
When a service needs to be restarted to apply new configuration you have set elsewhere in the web
interface, it is displayed in bold and the message Restart needed is displayed in the Comment
column.
If you stop, start or restart a service, any dependent services are also affected. The tabs of the
services that have been stopped or have never been started because their startup type is manual
are greyed out. Upon clicking these tabs, a list of services that must be restarted is displayed.
4.
Click the Restart Required Services button at the bottom of the page.
Graceful Restart of Services

You can set a delay to allow for telephony calls to be all completed before restarting services that need a
restart.
During that delay, it is impossible to make new calls but calls in progress are not terminated. When all calls
are completed, then the restart is authorized and the services that require a restart are restarted.
You can also set a unit restart grace period when performing a Firmware Upgrade as described in Firmware
Packs Configuration on page 563.
To configure the graceful restart of services:

1.
In the Restart Required Services section, set the Graceful Delay field with the delay (in minutes)
allowed for telephony calls to be all completed.
At the expiration of this delay, the services are forced to restart.
Figure 11: Services Restart Required Services Section
1
2.
Click Restart Required Services to restart only the services that needed a restart for their
configuration to be applied.
If you click Cancel, this cancels the restart during the grace delay period.
Restarting a Service via MIB

If you are using a MIB browser to access the Mediatrix unit configuration via SNMP, you can determine
whether or not a service needs to be restarted by locating the configurationGroup folder of the related service
and checking if the service needs to be restarted in the needRestartInfo variable.
Figure 12: Need Restart Info
If a specific service needs to be restarted, locate the scmMIB, then set the serviceCommandsRestart variable
for this service to restart.
58
Services Table
Figure 13: Restart Service
You can also start a service by setting the serviceCommandsStart variable for this service to Start.
You can also stop a service by setting the serviceCommandsStop variable for this service to Stop.
If you are not able to restart a service because it is a system service, you must restart the Mediatrix unit.
59
60
Services Table
H A P T E R
Hardware Parameters
This chapter describes the hardware installed on your Mediatrix unit.
Note: This web page does not apply to the Mediatrix LP, 4100, C7, Mediatrix 3200, and Mediatrix 3300
Series.
Hardware Card Configuration

The Hardware page will differ depending on the product and model you have. However, this page will allow
you to set a few parameters.
BRI Hardware Configuration

This section applies to the Mediatrix 3404 / 3408 / 3734 / 3741 / 3742 / 4401 / 4402 / 4404 models.
To configure the Mediatrix unit BRI hardware:

1.
In the web interface, click the System link, then the Hardware sub-link.
Figure 14: System Hardware Web Page (Mediatrix 3404 / 3408)
Figure 15: System Hardware Web Page (Mediatrix 4400 Series)
2.
In the BRI Cards Configuration section, select the reference of the clock source in the Clock
Reference drop-down menu.
61
Chapter 5 - Hardware Parameters
A clock reference is needed to package the ISDN voice frames. It can be generated internally or
obtained from an external source (e.g., public ISDN). If you do have an interface connected to a
public ISDN, select the clock reference from this interface. The clock reference provides error-free
ISDN data and fax transmission.
The clock reference may either be provided by the network or by one of the BRI ports of the
Mediatrix unit.
Caution: If the clock reference is not configured according to the network environment, clock slips and
related voice quality degradations may occur.
If you want to configure the clock reference of a specific interface, you must set the Endpoint Type
of this interface to NT. See BRI Configuration on page 197 for more details.
Table 49: BRI Clock Reference
Reference
Description
None
The internal clock does not synchronize with any other source.
Other
Card
The internal clock synchronizes with the other BRI card of the Mediatrix unit. This
interface must be configured in TE mode (Endpoint drop-down menu of the
Interface Configuration section) to provide the clock reference to the other
interfaces.
Note: This choice is not available on the Mediatrix 3404 model.

Bri0 to Bri4 The internal clock synchronizes with the signal provided by the corresponding
interface. This interface must be configured in TE mode (Endpoint drop-down
menu of the Interface Configuration section) to provide the clock reference to the
other interfaces.
Bri1 to Bri4 The internal clock synchronizes with the signal provided by the corresponding
(Mediatrix interface. This interface must be configured in TE mode (Endpoint drop-down
menu of the Interface Configuration section) to provide the clock reference to the
4400
other interfaces.
Series)
If the selected source does not provide a signal, this setting automatically falls back to None (no
synchronization).
3.
Click Submit if you do not need to set other parameters.
PRI Hardware Configuration

This section applies to the Mediatrix 3500 / 3600 Series model.
To configure the Mediatrix unit PRI hardware:

1.
Figure 16: System Hardware Web Page
62
2.
In the PRI Cards Configuration section, select the reference of the clock source in the Clock
A clock reference is needed to package the ISDN or R2 voice frames. It can be generated internally
or obtained from an external source (e.g., public ISDN or R2). If you do have an interface connected
to a public ISDN or R2, select the clock reference from this interface. The clock reference provides
error-free ISDN/R2 data and fax transmission.
The clock reference may either be provided by the network or by one of the PRI ports of the
Mediatrix unit.
Caution: If the clock reference is not configured according to the network environment, clock slips and
related voice quality degradations may occur.
If you want to configure the clock reference of a specific ISDN interface, you must set the Endpoint
Type drop-down menu to NT. See PRI Configuration on page 186 for more details.
Table 50: ISDN PRI Clock Reference
Reference
Description
None
Other
Card
The internal clock synchronizes with the other PRI interface of the Mediatrix unit.
This interface must be configured in TE mode (Endpoint drop-down menu of the
interfaces.
Note: This choice is not available on the Mediatrix 3531, 3621 and 3631 models.
If you want to configure the clock reference of a specific R2 interface, you must set the Clock Mode
drop-down menu to Master. See R2 Channel Associated Signaling on page 226 for more details.
Table 51: R2 Clock Reference
Reference
Description
None
Other
Card
The internal clock synchronizes with the other R2 interface of the Mediatrix unit.
This interface must be configured in Slave mode (Clock Mode-down menu of the R2
Channel Associated Signaling section) to provide the clock reference to the other
interfaces.
Note: This choice is not available on the Mediatrix 3621 and 3631 models.
3.
Select whether the line uses T1 or E1 in the Line Type drop-down menu.
You must restart the unit if you change this setting.
Currently, R2 works only on the E1 line type.
4.
Select whether the unit uses the ISDN or R2 signaling in the Signaling drop down menu.
If you are using ISDN signalling, refer to Chapter 21 - ISDN Configuration on page 179 for more
information on the ISDN parameters to configure.
If you are using R2 signalling, refer to Chapter 22 - R2 CAS Configuration on page 223 for more
information on the R2 parameters to configure.
When changing from R2 to ISDN or ISDN to R2, you must change your routes accordingly. For
instance, if you are in ISDN with a route isdn-Slot2/E1T1, then change to R2, you must change the
route to r2-Slot2/E1T1.
5.
63
BRI Power Feeding Configuration

The ISDN power feeding option is only available on the Mediatrix 4400plus models. It is offered as per ITU-T
I.430 (section 9).
To configure the BRI power feeding:

1.
In the BRI Cards Configuration section, select whether to enable or disable the ISDN power feeding
for a given port of the Mediatrix 4400plus model in the Power Feeding drop-down menu.
The number of interfaces available vary depending on the model you have.
Figure 17: ISDN Basic Rate Interface Web Page
The Status section indicates the state of the ISDN power feeding for a port.
2.
64
On: Power is fed over the ISDN line.
Off: Power is not fed over the ISDN line.
Unavailable: The power feeding feature is unavailable on endpoints of type TE.
Unsupported: The power feeding feature is not physically supported on this port.
Bridging Configuration
Note: This section is available only on the following models:
Mediatrix 4102S
Mediatrix 4104
Mediatrix 4108/4116/4124
Mediatrix LP16 / LP24
Mediatrix C7 Series
For Mediatrix unit models that have two Ethernet ports, you can configure how each port provides a link
interface.
To configure the bridging parameter:

1.
Figure 18: System Hardware Web Page (Mediatrix 4102S shown)
2.
In the Unit Configuration section, set the Eth Ports drop-down menu with the proper behaviour.
Table 52: Bridging Parameters
Parameter
3.
Description
Separate
Each Ethernet port provides an independent link interface. This is the required
configuration for IP Routing.
Bridge
Both Ethernet ports are bridged together and provide a single link interface.

The following message displays:
Note: Your Ethernet configuration has changed. A link interface will be deactivated. Make sure that
your network interfaces are configured accordingly prior to restarting the unit.
4.
In the web interface, click the Network link, then the Interfaces sub-link.
Figure 19: Network Interfaces Web Page
10
5.
Enter the name of the new interface for bridging in the blank field in the bottom left of the window,
then click the
button.
65
The name is case-sensitive. Using the special value All is not allowed.
6.
In the Interface Configuration section, select the link on which to activate the interface in the Link
column.
Select the link associated with the bridge. The name varies depending on the platform used.
7.
Select the configuration source of the interface information in the Type drop-down menu.
Table 53: Interface Configuration Sources
Source
Description
IPv4
DHCP
The IPv4 address and network mask are provided by querying a DHCP server and
using standard DHCP fields or options. Using the DHCP configuration assumes that
you have properly set your DHCP server with the relevant information. DHCP servers
may provide a list of IP configuration parameters to use. See DHCP Server
IPv4
Static
You manually enter the IPv4 address and network mask and they remain the same
every time the Mediatrix unit restarts. Use the static configuration if you are not using a
DHCP server/PPP peer or if you want to bypass it.
IPv4
PPPoE
IPv4 over PPP connection, address and network mask are provided by the PPP peer
using IPCP. PPP peers may provide a list of IP configuration parameters to use. See
PPPoE Configuration on page 107 for more details.
IPv6
AutoConf
IPv6 state-less auto-configuration.
IPv6
Static
every time the Mediatrix unit restarts. Use the IPv6 static configuration if you are not
using IPv6 stateless or stateful auto-configuration or if you want to bypass it.
Note: If no network is configured in IPv6, the unit does not have any IPv6 address, not even the Link-Local
address. When a network is configured in IPv6, the Link-Local (FE80 ::...) address is automatically created
and displayed in the Network Status information.
8.
If the interface configuration source is IPv4 Static or IPv6 Static, enter the address and network
mask (if applicable) of the network interface in the Static IP address field.
9.
If the interface configuration source is IPv4 Static or IPv6 Static, set the Static Default Router field
with the IP address of the default gateway for the network interface.
10.
Define whether or not the Mediatrix unit should attempt to activate the corresponding network
interface in the Activation drop-down menu.
It may not be possible to enable a network interface, for instance if another network interface is
already enabled in the same subnet. The actual status of network interfaces is shown in the Status
page.
Note: The newly created interface will be the only valid interface after the restart, make sure this interface
is Enabled and correctly configured according on the Interface Configuration Source (your network).
11.

The current network interface information is displayed in the Status page. See Interfaces
Configuration on page 102 for more details on network interfaces.
12.
66
Restart the Mediatrix unit to apply the change.
Ring Management
Note: This web page is available only on the Mediatrix 4102S model.

by using the CLI
You can determine how to ring more than one port. You have the following choices:
Table 54: Ring Management Parameters
Parameter
Cascade
Descriprion
The FXS ports are prevented from ringing at the same time in order to reduce the peak
power usage of the device.
Simultaneous All ports are ringing at the same time.
To set the ring management:

1.
In the MbLdpMIB, set the RingManagement variable to the proper value.

MbLdp.RingManagement="Value"

Table 55: Ring Management Values
Value
Description
100
Cascade
200
Simultaneous
67
68
H A P T E R
Endpoints State Configuration

This chapter describes how to set the administrative state of the Mediatrix units endpoints.
Unit Configuration
The unit configuration section allows you to define the administrative state of all the Mediatrix units endpoints.
To set the units endpoints parameters:

1.
In the web interface, click the System link, then the Endpoints sub-link.
Figure 20: System Configuration Endpoints Web Page
2.
In the Unit States section, select a temporary state for all of the units endpoints in the Action
column.
This command locks/unlocks all endpoints of the Mediatrix unit. This state is kept until you modify
it or the unit restarts. It offers the following settings:
Table 56: Action Settings
Setting
3.
Description
Force Lock
Cancels all the endpoints registration to the SIP server. All active calls in
progress are terminated immediately. No new calls may be initiated.
Lock
Cancels all the endpoints registration to the SIP server. Active calls in
progress remain established until normal call termination. No new calls
may be initiated.
Unlock
Registers the endpoints to the SIP server.
If you do not need to set other parameters, click Submit.
Endpoints Configuration
The endpoints configuration allows you to define the administrative state of the Mediatrix units endpoints.
69
Chapter 6 - Endpoints State Configuration
Administration
To set the endpoints parameters:

1.
In the Endpoint States section of the Endpoints page, select the permanent administrative state
each endpoint will have when the Mediatrix unit restarts in the Initial Administrative column.
Figure 21: Endpoint States Section
1
Table 57: Permanent Administrative State Settings

Setting
2.
Description
Unlocked
Registers the endpoint to the SIP server.
Locked
The endpoint is unavailable for normal operation. It cannot be used to

make and/or receive calls.
Select a temporary state for each endpoint in the corresponding Action column.
This command locks/unlocks an endpoint of the Mediatrix unit. This state is kept until you modify it
or the unit restarts. It offers the following settings:
Table 58: Action Settings
Setting
3.
Description
Force Lock
Cancels the endpoint registration to the SIP server. All active calls in
progress are terminated immediately. No new calls may be initiated.
Lock
Cancels the endpoint registration to the SIP server. Active calls in

progress remain established until normal call termination. No new calls
may be initiated.
Unlock
Registers the endpoint to the SIP server.
If you do not need to set other parameters, click Submit.
Administration
The Administration section allows you to define endpoint operational state.
To set administration parameters:

1.
In the Administration section of the Endpoints page, set the Disable Unit (All Endpoints) When No
Gateways Are In State Ready drop-down menu with the proper behaviour.
Figure 22: Administration Section
1
2
Table 59: Unit Operational State Parameters

Parameter
Disable
70
Description
Signaling gateways have no impact on the unit operational
state
Administration
Table 59: Unit Operational State Parameters (Continued)

Parameter
Description
Enable
2.
When all signaling gateways are not ready, the unit

operational state is set to disabled.
Set the Shutdown Endpoint When Operational State is Disable And Its Usage State Is 'idleunusable' drop-down menu with the proper behaviour.
Table 60: Endpoint Shutdown Parameters
Parameter
Description
Enable
When the usage state becomes Idle-unusable and the

operational state becomes Disable, the endpoint is
physically shutdown.
Disable
When an endpoint's usage state becomes Idle-unusable

whatever the value of its operational state, the endpoint
remains physically up but the calls are denied.
The default value is:
3.
Enable for the Mediatrix LP/4100/C7 series
Disable for the Mediatrix 3000 and Mediatrix 4400 series
Unit Shutting Down Behaviour


by using the CLI
You can configure the behaviour of the call permissions when the UnitAdminState is ShuttingDown.
The following parameters are available:
Table 61: Unit Shutting Down Behaviour Parameters
Parameter
Description
BlockNewCalls No new requests are accepted once all activity are terminated. Endpoints cannot make and
receive calls.
AllowNewCalls New requests are accepted until all activities are simultaneously terminated. Endpoints can
make and receive calls.
To set the unit shutting down behaviour:

1.
In the epAdmMIB, locate the UnitConfigGroup folder.
2.
Set the behaviorWhileInUnitShuttingDownState variable with the proper behaviour.

epAdm.behaviorWhileInUnitShuttingDownState="Value"
71
Chapter 6 - Endpoints State Configuration
Administration
where Value may be one of the following:

Table 62: Unit Shutting Down Behaviour Values
Value
72
Meaning
100
BlockNewCalls
200
AllowNewCalls
H A P T E R
Syslog Configuration
This chapter describes how the Mediatrix unit handles syslog messages and notification messages.
For a list and description of all syslog messages and notification messages that the Mediatrix unit may send,
refer to the Notification Reference Guide.
Syslog Daemon Configuration

Standards Supported
RFC 3164: The BSD Syslog Protocol
The Syslog daemon is a general purpose utility for monitoring applications and network devices with the TCP/
IP protocol. With this software, you can monitor useful messages coming from the Mediatrix unit unit. If no
Syslog daemon address is provided by a DHCP server or specified by the administrator, no messages are
sent.
For instance, if you want to download a new firmware into the Mediatrix unit, you can monitor each step of the
firmware download phase. Furthermore, if the unit encounters an abnormal behaviour, you may see accurate
messages that will help you troubleshoot the problem.
The Mediatrix unit supports RFC 3164 as a device only (see definition of device in section 3 of the RFC).
73
Chapter 7 - Syslog Configuration
To configure the Mediatrix unit syslog client:

1.
In the web interface, click the System link, then the Syslog sub-link.
Figure 23: System Syslog Web Page
4
5
2.
Set the static IP address or domain name and port number of the device to use to archive log entries
in the Remote Host field.
Use the special port value zero to indicate the protocol default. For instance, the TFTP default port
is 69 and the HTTP/HTTPS default port is 80.
3.
In the Service Severity section, select the minimal severity to issue a notification message for the
various services in the corresponding drop-down menus.
Any syslog message with a severity value greater than the selected value is ignored. Available
values are:
Table 63: Severity Values
Severity
74
Description
Notification Messages Issued
Disable
N/A
No notification is issued.
Debug
Message describing in detail the unit's

operations.
All notification messages are issued.
Info
Message indicating a significant event for

the unit's normal operations.
Notification messages with severity

Informational and higher are issued.
Warning
Message indicating an abnormal event or

situation that could be potentially risky.
The unit may not be fully operational.

Warning and higher are issued.
Table 63: Severity Values (Continued)

Severity
Description
Notification Messages Issued
Error
Message indicating an abnormal event or Notification messages with severity

situation, the system's operation is
Error and higher are issued.
affected. The unit may not be operational.
Critical
Message indicating a critical event or

situation that requires immediate
attention. The unit is not operational.

Critical are issued.
A higher level mask includes lower level masks, e.g., Warning includes Error and Critical. The
default value is Warning.
4.
In the Technical Assistance Centre section, enable diagnostic traces by setting the Diagnostic
Traces drop-down menu to Enable.
At the request of Media5s Technical Support personnel, enabling these traces will allow Media5 to
further assist you in resolving some issues. However, be advised that enabling this feature issues
a lot of messages to the syslog host. These messages may be filtered by using the Diagnostic
Traces Filter field.
Note: Enabling all the traces could affect the performance of the Mediatrix unit.
5.
If applicable, define the filter applied to diagnostic traces by clicking the Edit button in the Filter field.
The following opens:
Figure 24: Diagnostic Traces Window
You can use the filter to narrow down the number of traces sent at the request of Media5s Technical
Support personnel.
6.
75
Configuring PCM Capture


by using the CLI
Media5 Technical Support personnel may ask you to enable the PCM traces. PCM traces are an efficient tool
to identify problems with:
Echo in your network

DTMF signals
Caller ID signals
Fax signals (or false Fax detection)
Message Waiting Indicator signals
Any other analog or digital signal
The PCM traces are two different RTP streams made specifically to record all analog signals that are either
sent or received on the analog side of the Mediatrix unit. Only the configured port, port #1 and/or #2, are
sending the PCM traces for a maximum of four simultaneous RTP streams.
The RTP streams are sent to a configurable IP address, normally an IP address on your network where it can
be recorded with a packet sniffer (such as Wireshark). Moreover, they are independent from the regular RTP
streams of the VoIP call.
PCM capture supports sending streams with a ptime higher than 20 ms. The Mediatrix unit does not support
sending streams with a ptime of 10 ms.
To enable PCM capture:

1.
In the miptMIB, locate the pcmCaptureGroup folder under the debugGroup folder.
2.
Set the pcmCaptureEnable variable to enable.

mipt.pcmCaptureEnable="1"
3.
Set the unit's endpoint on which the PCM capture must be performed in the pcmCaptureEndpoint
variable.
mipt.pcmCaptureEndpoint="Value"
The format is InterfaceName-Channel#. For digital interfaces (such as ISDN), you must append a
-Channel# for the requested channel.
The list of endpoints is available under EpAdm.EndpointTable. Valid examples (depending of the
platform) are:
PCM capture is to be done on channel #3 of a PRI interface located in slot #2: Slot2/
E1T1-3
PCM capture is to be done on channel #2 of a BRI interface: Bri1-2
PCM capture is to be done on the 16th FXS port: Port16
Note: Note that PCM capture does not support capturing on multiple endpoints simultaneously.
4.
Set the IP address where the captured PCM packets should be sent in the pcmCaptureIpAddr
variable.
mipt.pcmCaptureIpAddr="Value"
76
The PCM traces destination must be set so it can be recorded in a Wireshark capture on your
network, normally sent to the PC doing the capture.
Configuring the Syslog Daemon Application

You must configure the Syslog daemon server to capture those messages. Refer to your Syslog daemons
documentation to learn how to properly configure it to capture messages.
77
78
H A P T E R
Events Configuration
This chapter describes how to associate a NOTIFICATION message and how to send it (via syslog or via a
SIP NOTIFY packet).
For a list and description of all syslog messages and notification messages that the Mediatrix unit may send,
refer to the Notification Reference Guide.
Notification Events
You can configure an event router in order to apply a set of rules to select the proper transport protocol
scheme. A rule entry is made up of three different values: type, criteria and action.
Note that more than one notification may be sent for a single event based on the event router table rules.
To configure notification events:

1.
Ensure that the severity level for all services are set according to the severity level of the notification
messages that are required by the system administrator. See Chapter 7 - Syslog Configuration on
2.
In the web interface, click the System link, then the Events sub-link.
Figure 25: System Events Web Page
8
3
3.
If you want to add a rule entry before an existing entry, locate the proper row in the table and click
the
button of this row.
4.
Set the Activation drop-down menu with the current activation state for the corresponding system
event.
Table 64: Activation Parameters
Parameter
5.
Description
Enable
This action is enabled for this system event.
Disable
This action is disabled for this system event.
Optional: Set the corresponding Criteria field with the expression an event must match in order to
apply the specified action. The expression is based on the event type.
This step is optional because a proper value may be automatically entered by the Mediatrix unit
upon setting the Service (Step 5) and Notification (Step 6) drop-down menus.
79
Chapter 8 - Events Configuration
Notification Events
An event of type notification uses the notification ID as expression criteria. The notification ID is the
combination of the service number key and the message number key separated by a dot. The
information regarding the service and message number key is available in the Notification
Reference Guide document.
Several basic criteria can also be specified on the same line, separated by commas. Criteria can
specify inclusion or exclusion. A group of exclusion criteria can follow the group of inclusion criteria.
The group of exclusion criteria must begin with a hyphen (-).
Matching an inclusion criteria causes the action to be executed unless an exclusion criteria is also
matched. Exclusion criteria have precedence over inclusion criteria.
Spaces are allowed before or after a basic criterion; however, spaces are not accepted within a
basic criterion, i.e. before or after the dot.
Examples:
Service ISDN (number key = 1850)
Message %1$s: Physical link state changed to up (number key = 5)
The corresponding Criteria is: 1850.5

You can also use the special expression All, which means all available services and messages.
Criteria 1850.All,1600.200,1600.W,-1850.500,1600.300
1850.All,1600.200,1600.W are inclusion criteria and -1850.500,1600.300 are exclusion criteria. All
notifications from service 1850, except notification 500, will match the expression. All notifications
from service 1600 with Warning level, except notification 300, will match the expression. Notification
200 from service 1600 will match the expression, no matter the severity level.
6.
In the corresponding Service drop-down menu, select the service for which you want to send
events.
You can use the Suggestion columns drop-down menu to select between suggested values, if any.
7.
In the Notification drop-down menu, select the notification message that you want to send.
8.
In the Action drop-down menu, select the action to apply to the system event if the criteria matches.
The action represents a transport targeted for the event. The format of the event under which the
message is carried is dependent on the protocol in use.
The possible actions are:
Table 65: Action Parameters
Parameter
9.
Description
Send Via
Syslog
The event notification is sent using syslog as transport. See Chapter 7 - Syslog
Send Via
SIP
The event notification is sent using SIP Notify as transport.
Log
Locally
The event notification is logged in Local Log.
Click the Submit button.

The configuration status of the row displays on the right part of the row. It indicates whether the
configuration of the row is valid.
Table 66: Configuration Status Values
Value
80
Description
Valid
The current content of the fields Type, Criteria and Action is valid.
Invalid
The current content of the fields Type, Criteria and Action is not valid.
Notification Events
Table 66: Configuration Status Values (Continued)

Value
Not
Supported
Description
The current content of the fields Type, Criteria and Action is valid but not
supported.
Deleting a Rule
You can delete a rule row from the table in the web interface.
To delete a rule entry:

1.
Click the
button of the row you want to delete.
2.
Monitoring Parameters

by using the CLI
You can set two monitoring parameters for the Notification Events table.
To set monitoring parameters:

1.
In the sipEpMIB, locate the MonitoringGroup folder.
2.
Set the sipNotificationsGateway variable with the SIP gateway used to send SIP NOTIFY
containing the notification events.
sipEp.sipNotificationsGateway="Value"
Value is the name of the SIP gateway from which the NOTIFICATION is sent.
3.
Set the maxNotificationsPerNotify variable with the maximal number of notification events the
device may have to send in one SIP NOTIFY request.
Notifications are sent in XML elements through the SIP NOTIFY's body request.
sipEp.maxNotificationsPerNotify="Value"
Value may be between 1 and 25.
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Chapter 8 - Events Configuration
82
Notification Events
H A P T E R
Local Log
This chapter describes local log status and entries generated by the Notifications and Logging Manager (NLM)
for your Mediatrix unit.
Local Log Status and Entries

You can display, clear and refresh local log status and entries.
To manage local log status and entries:

1.
In the web interface, click the System link, then the Local Log sub-link.
Figure 26: System Local Log Web Page
The following is the Local Log Status information displayed.

Table 67: Local Log Status Parameters
Parameter
Description
Maximum Number of Entries
Maximum number of entries that the local log can contain. When
adding a new entry while the local log is full, the oldest entry is
erased to make room for the new one.
Number of Error Entries
Current number of error entries in the local log.
Number of Critical Entries
Current number of critical entries in the local log.
The following is the Local Log Entries information displayed.

Table 68: Local Log Entries Parameters
Parameter
Description
Local Time
Local date and time at which the log entry was inserted. Format is
YYYY-MM-DD HH:MM:SS.
Severity
Severity of the log entry.
Service Name
Textual identifier of the service that issued the log entry.
Service Key
Numerical identifier of the service that issued the log entry.
Message Key
Numerical identifier of the notification message.
83
Chapter 9 - Local Log
Local Log Status and Entries
Table 68: Local Log Entries Parameters

Parameter
Message Content
84
Description
The readable content of the log message.
2.
Click Clear Local Log to clear all log entries.
3.
Click Refresh Local Log to refresh the log entries display.
Network Parameters
H A P T E R
10
IPv4 vs. IPv6

This chapter describes the differences between IPv4 and IPv6 addressing.
Introduction
IPv6 (Internet Protocol version 6) is the successor to the most common Internet Protocol today (IPv4). This is
largely driven by the fact that IPv4s 32-bit address is quickly being consumed by the ever-expanding sites and
products on the internet. IPv6s 128-bit address space should not have this problem for the foreseeable future.
IPv6 addresses, in addition to being longer, are distinguished from IPv4 addresses by the use of colons ":",
e.g., 2001:470:8929:4000:201:80ff:fe3c:642f. An IPv4 address is noted by 4 sets of decimal numbers
separated by periods ".", e.g., 192.168.10.1.
Please note that IPv6 addresses should be written between [ ] to allow port numbers to be set. For instance:
[fd0f:8b72:5::1]:5060.
IPv4 vs. IPv6 Availability

The Mediatrix unit fully supports IPv4 IP addresses, as well as IPv6 IP addresses in some of its features. The
following table lists all the network related features of the Mediatrix unit with their availability in IPv4 and IPv6.
Table 69: IPv4 vs. IPv6 Availability
Feature
IPv4
IPv6
Backup/Restore transfer
Configuration file transfer
Embedded DHCP server
Firmware Transfer
IP Routing
IP Sync
Link Layer Discovery Protocol (LLDP) QoS settings
Local Firewall (LFW)
Network Address Translation (NAT)
Network Configuration (IP addresses, DNS and SNTP servers)
Network Firewall (NFW)
Online Certificate Status Protocol (OCSP)
Remote Authentication Dial In User Service (Radius )
SIP signaling and media transport
Simple Network Management Protocol (SNMP)
87
Chapter 10 - IPv4 vs. IPv6
IPv6 Scope Identifier
Table 69: IPv4 vs. IPv6 Availability (Continued)

Feature
IPv4
IPv6
TR-069
WEB Configuration
If you configure the Mediatrix unit with IPv6 addresses, then decide to go downgrade to a firmware version
that does not support IPv6, all IPv6 networks are deleted.
Please note that IPv6 addresses should be written between [ ]. For instance: [fd0f:8b72:5::1].

When using an IPv6 address starting with "FE80::" (IPv6 link-local addresses), there must be additional
information: the IPv6 scope identifier (this represents the network link that will be used to contact the IPv6 linklocal address). The format is "[IPv6 link-local%ScopeIdentifier]".
When Contacting the unit using its IPv6 link-local Address

On Windows, the scope identifier is represented by an interface number. The interface number can be
determined through the command line of Windows.
Go to Start -> Run and type cmd to enter the command prompt.
At the command prompt, type ipconfig and find the IPv6 address. Appended to the end of this
will be a "%x" where x is the interface number.
To contact the IPv6 link-local IPv6 address "fe80::201:80ff:fe3c:642f", you would use:
[fe80::201:80ff:fe3c:642f%4]
On Linux, the scope identifier may be the link name or the interface number. The interface number can be
determined through the Linux command line.
To contact the IPv6 link-local IPv6 address "fe80::201:80ff:fe3c:642f", you would use:
[fe80::201:80ff:fe3c:642f%2] or [fe80::201:80ff:fe3c:642f%eth0]
88
When Configuring the Mediatrix unit to use an IPv6 link-local Address

In that case, the scope identifier represents the "link" in Network/Interfaces.
For instance, if you want your unit to contact a server with the address IPv6 link-local
"fe80::201:80ff:fe3c:642f", you must check on which network link the server is available. Some units have
"wan" or "lan". Lets say it is on the "wan" link. The IP address whoud then become
"[fe80::201:80ff:fe3c:642f%wan]".
89
Chapter 10 - IPv4 vs. IPv6
90
H A P T E R
11
Host Parameters
This chapter describes how to set the host information of the Mediatrix unit:
General Configuration (automatic configuration interface)

Host name and domain name.
Default gateway parameters.
DNS parameters.
SNTP client parameters.
Time parameters.
General Configuration
The General Configuration section allows you to configure the networks that will provide the automatic
configuration (host name, default gateway, DNS servers and SNTP servers) used by the Mediatrix unit.
Automatic configuration may be provided via IPv4 (DHCPv4) and/or via IPv6 (stateless auto-configuration and
DHCPv6).
To set the general configuration:

1.
In the web interface, click the Network link, then the Host sub-link.
Figure 27: Network Host Web Page
2
3
2.
Set the Automatic IPv4 config source network drop-down menu with the IPv4 network interface that
provides the automatic configuration.
3.
Set the Automatic IPv6 config source network drop-down menu with the IPv6 network interface that
provides the automatic configuration.
4.

The current automatic configuration interface is displayed in the Status page.
Host Configuration
The Host Configuration section allows you to configure the host name and domain name of the Mediatrix unit.
91
Chapter 11 - Host Parameters
Host Configuration
To set the host configuration:

1.
In the Host Configuration section of the Host page, select the configuration source of the domain
name information in the Domain Name Configuration Source drop-down menu.
Figure 28: Host Name Configuration Section
2
3
4
Table 70: Host Name Configuration Sources

Source
Description
Automatic The domain name is automatically obtained from the network. The value obtained
IPv4
depends on the connection type of the automatic network interface (see General
Configuration on page 91) if any. Using the automatic configuration assumes that
you have properly set your network server with the relevant information.
Note: Some Uplink connection types (for example Static and PPPoE) cannot obtain
domain name information from the network, and therefore lead to no domain name
being applied to the system.
Automatic The domain name is automatically obtained from the IPv6 network defined in the
IPv6
Automatic IPv6 config source network drop-down menu.
Static
You manually enter the domain name and it remains the same every time the
Mediatrix unit restarts. Use the static configuration if you are not using a network
server or if you want to bypass it.
When switching from the Static to Automatic IPv4 or Automatic IPv6 configuration source, the last
value correctly obtained from the network (if any) is applied to the system.
Static Configuration Source Only
2.
Set the systems domain name in the Domain Name field.

A domain name is a name of a device on the Internet that distinguishes it from the other systems
on the network. For instance: example.com.
3.
Set the system's host name in the Host Name field.

The host name is the unique name by which the device is known on a network. It may contain any
of the following characters:
A to Z and a to z letters
0 to 9 digits
-._~
!$&'()*+=
Certain restrictions apply to this name:
4.
The host name must be shorter than 64 characters.
The host name must not start with a period.
The host name must not contain double quotes, semicolons, curly braces, spaces, and
commas.
The host name must not contain the following characters: :/?#[@

The current domain name is displayed in the Status page.
92
Default Gateway Configuration

The default gateway (also known as default router) is the gateway to which the Mediatrix unit sends packets
when all other internally known routes have failed.
To set the default gateway configuration:

IPv4 Configuration
1.
In the Default Gateway Configuration IPv4 section of the Host page, select the IPv4 configuration
source of the default gateway information in the Configuration Source drop-down menu.
Figure 29: Default Gateway Configuration Section
1
2
3
4
Table 71: Default Gateway Configuration Sources

Source
Description
Automatic The default gateway is automatically obtained from the network. The value obtained
IPv4
Note: Some Uplink connection types (for example Static) cannot obtain default
gateway information from the network, and therefore lead to no default gateway being
applied to the system.
Static
You manually enter the IP address of the default gateway and it remains the same
every time the Mediatrix unit restarts. Use the static configuration if you are not using
a network server or if you want to bypass it.
When switching from the Static to Automatic configuration source, the last value correctly obtained
from the network (if any) is applied to the system.
IPv4 Static Configuration Source Only
2.
If the default gateway configuration source is Static, enter the static default gateway address in the
IP address field.
This can be an IP address or domain name. The default value is 192.168.10.10.
IPv6 Configuration
3.
In the Default Gateway Configuration IPv6 section of the Host page, select the IPv6 configuration
source of the default gateway information in the Configuration Source drop-down menu.
Table 72: IPv6 Default Gateway Configuration Sources
Source
Description
Automatic The default gateway name is automatically obtained from the IPv6 network defined
IPv6
in the Automatic IPv6 config source network drop-down menu.
Static
You manually enter the IPv6 address of the default gateway and it remains the same
93
DNS Configuration
When switching from the Static to Automatic IPv6 configuration source, the last value correctly
obtained from the network (if any) is applied to the system.
4.
If the default gateway configuration source is Static, enter the static default gateway IPv6 address
in the IP address field.
This can be an IP address or domain name.
5.

The current default gateway address is displayed in the Status page.
DNS Configuration
Standards Supported
RFC 1034: Domain Names - Concepts and Facilities
RFC 1035: Domain Names - Implementation and

Specification
RFC 1886: DNS Extensions to support IP version 6
RFC 2181: Clarifications to the DNS Specification
You can use up to four Domain Name Servers (DNS) to which the Mediatrix unit can connect. The DNS
servers list is the ordered list of DNS servers that the Mediatrix unit uses to resolve network names. DNS
query results are cached on the system to optimize name resolution time.
To set the DNS configuration:

1.
In the DNS Configuration section of the Host page, select the configuration source of the DNS
information in the Configuration Source drop-down menu.
Figure 30: DNS Configuration Section
1
2
Table 73: DNS Configuration Sources

Source
Description
Automatic The DNS servers are automatically obtained from the network. The value obtained
IPv4
Note: Some Uplink connection types (for example Static) cannot obtain DNS
information from the network, and therefore lead to no DNS servers being applied to
the system.
Automatic The DNS servers are automatically obtained from the IPv6 network defined in the
IPv6
Automatic IPv6 config source network drop-down menu.
Static
You manually enter up to four DNS servers IP addresses and they remain the same
values correctly obtained from the network (if any) are applied to the system.
94
SNTP Configuration
2.
3.
If the DNS configuration source is Static, enter up to four static DNS addresses in the following
fields:
Primary DNS
Secondary DNS
Third DNS
Fourth DNS

The current list of DNS servers is displayed in the Status page.
SNTP Configuration
Standards Supported
RFC 2030: Simple Network Time Protocol (SNTP) Version 4

for IPv4, IPv6 and OSI
bootp-dhcp-option-88
The Simple Network Time Protocol (SNTP) enables the notion of time (date, month, time) into the Mediatrix
unit. SNTP is used to synchronize a SNTP client with a SNTP or NTP server by using UDP as transport. It
updates the internal clock of the unit to maintain the system time accurate. It is required when dealing with
features such as the caller ID.
The Mediatrix unit implements a SNTP version 3 client.
Note: The Mediatrix unit hardware does not include a real time clock. The unit uses the SNTP client to get
and set its clock. As certain services need correct time to work properly (such as HTTPS), you should
configure your SNTP client with an available SNTP server in order to update and synchronise the local clock
at boot time.
To set the SNTP client of the Mediatrix unit:

1.
In the SNTP Configuration section of the Host page, select the configuration source of the SNTP
Figure 31: SNTP Configuration Section
1
2
3
Table 74: SNTP Configuration Sources

Source
Description
Automatic The SNTP parameters are automatically obtained from the network. The value
IPv4
obtained depends on the connection type of the automatic network interface (see
General Configuration on page 91) if any. Using the automatic configuration
assumes that you have properly set your network server with the relevant
information.
Note: Some Uplink connection types (for example Static and PPPoE) cannot obtain
SNTP information from the network, and therefore lead to no SNTP parameters being
applied to the system.
Automatic The SNTP parameters are automatically obtained from the IPv6 network defined in
IPv6
the Automatic IPv6 config source network drop-down menu.
95
Time Configuration
Table 74: SNTP Configuration Sources (Continued)

Source
Static
Description
You manually enter the values and they remain the same every time the Mediatrix
unit restarts. Use the static configuration if you are not using a network server or if
you want to bypass it.
values correctly obtained from the network (if any) are applied to the system.
2.
3.
If the SNTP configuration source is Static, enter up to four static SNTP server IP addresses or
domain names and port numbers in the following fields:
Primary SNTP
Secondary SNTP
Third SNTP
Fourth SNTP
Set the synchronization information:

Table 75: SNTP Synchronization Information
Field
Description
Synchronisation Period
Time interval (in minutes) between system time

synchronization cycles. Each time this interval expires, a
SNTP request is sent to the SNTP server and the result is
used to set the system time. The maximum value is set to
1 440 minutes, which corresponds to 24 hours.
Synchronisation Period on Error Time interval (in minutes) between retries after an
unsuccessful attempt to reach the SNTP server. The
maximum value is set to 1 440 minutes, which corresponds to
24 hours.
4.

The current SNTP host is displayed in the Status page.
Time Configuration
Standards Supported
bootp-dhcp-option-88
You can define the current system date and time configured in the unit by specifying in which time zone the
unit is located.
If the time seems not valid, verify the SNTP configuration in SNTP Configuration on page 95.
To set the time of the Mediatrix unit:

1.
In the Time Configuration section of the Host page, enter a valid string in the Static Time Zone field.
Figure 32: Time Configuration Section
1
The format of the string is validated upon entry. Invalid entries are refused. The default value is:
EST5DST4,M4.1.0/02:00:00,M10.5.0/02:00:00
96
Time Configuration
A POSIX string is a set of standard operating system interfaces based on the UNIX operating
system. The format of the IEEE 1003.1 POSIX string is defined in the bootp-dhcp-option-88 Internet
draft as:
STDOFFSET[DST[OFFSET],[START[/TIME],END[/TIME]]]
Refer to the following sub-sections for explanations on each part of the string.
2.

The current system time is displayed in the Status page.
STD / DST
Three or more characters for the standard (STD) or alternative daylight saving time (DST) time zone. Only STD
is mandatory. If DST is not supplied, the daylight saving time does not apply. Lower and upper case letters are
allowed. All characters are allowed except digits, leading colon (:), comma (,), minus (-), plus (+), and ASCII
NUL.
OFFSET
Difference between the GMT time and the local time. The offset has the format h[h][:m[m][:s[s]]]. If no offset is
supplied for DST, the alternative time is assumed to be one hour ahead of standard time. One or more digits
can be used; the value is always interpreted as a decimal number.
The hour value must be between 0 and 24. The minutes and seconds values, if present, must be between 0
and 59. If preceded by a minus sign (-), the time zone is east of the prime meridian, otherwise it is west, which
can be indicated by the preceding plus sign (+). For example, New York time is GMT 5.
START / END
Indicates when to change to and return from the daylight saving time. The START argument is the date when
the change from the standard to the daylight save time occurs; END is the date for changing back. If START
and END are not specified, the default is the US Daylight saving time start and end dates. The format for start
and end must be one of the following:
n where n is the number of days since the start of the year from 0 to 365. It must contain the
leap year day if the current year is a leap year. With this format, you are responsible to
determine all the leap year details.
Jn where n is the Julian day number of the year from 1 to 365. Leap days are not counted. That
is, in all years including leap years February 28 is day 59 and March 1 is day 60. It is
impossible to refer to the occasional February 29 explicitly. The TIME parameter has the same
format as OFFSET but there can be no leading minus (-) or plus (+) sign. If TIME is not
specified, the default is 02:00:00.
Mx[x].y.z where x is the month, y is a week count (in which the z day exists) and z is the day
of the week starting at 0 (Sunday). For instance:
M10.4.0
is the fourth Sunday of October. It does not matter if the Sunday is in the 4th or 5th week.
M10.5.0
is the last Sunday of October (5 indicates the last z day). It does not matter if the Sunday is in the
4th or 5th week.
M10.1.6
is the first week with a Saturday (thus the first Saturday). It does not matter if the Saturday is in the
first or second week.
The TIME parameter has the same format as OFFSET but there can be no leading minus (-) or plus
(+) sign. If TIME is not specified, the default is 02:00:00.
97
Additional Parameters
Example
The following is an example of a proper POSIX string:
Standard
time zone
Offset
Month, Week, and Day

to start the Daylight
Saving Time
Month, Week, and Day

to stop the Daylight
Saving Time
EST5DST4,M4.0.0/02:00:00,M10.5.0/02:00:00
Daylight
Saving Time
time zone
Offset
Time to stop
the Daylight
Saving Time
Time to start the

Daylight Saving
Time
The following are some valid POSIX strings:

Table 76: Valid POSIX Strings
Time Zone
POSIX String
Pacific Time (Canada & US)
PST8PDT7,M3.2.0/02:00:00,M11.1.0/02:00:00
Mountain Time (Canada & US)
MST7MDT6,M3.2.0/02:00:00,M11.1.0/02:00:00
Central Time (Canada & US)
CST6CDT5,M3.2.0/02:00:00,M11.1.0/02:00:00
Eastern Time Canada & US)
EST5EDT4,M3.2.0/02:00:00,M11.1.0/02:00:00
Atlantic Time (Canada)
AST4ADT3,M3.2.0/02:00:00,M11.1.0/02:00:00
GMT Standard Time
GMT0DMT-1,M3.5.0/01:00:00,M10.5.0/02:00:00
W. Europe Standard Time
WEST-1DWEST-2,M3.5.0/02:00:00,M10.5.0/03:00:00
China Standard Time
CST-8
Tokyo Standard Time
TST-9
Central Australia Standard Time
CAUST-9:30DCAUST-10:30,M10.5.0/02:00:00,M3.5.0/02:00:00
Australia Eastern Standard Time
AUSEST-10AUSDST-11,M10.5.0/02:00:00,M3.5.0/02:00:00
UTC (Coordinated Universal Time)
UTC0

by using the CLI
Configuring DNS Records Randomization

You can define how the DNS A/AAAA records are accessed from the devices internal DNS cache using the
DnsCacheRecordsRandomization variable.
98
The following values are available:

Table 77: DNS Cache Records Randomization Values
Value
Description
Enable
When DNS A/AAAA records are accessed from the cache, they are sent to requesting
service in a randomized order.
Disable
When DNS A/AAAA records are accessed from the cache, they are sent to requesting
service in the same order they were originally received from the network. This is the default
value.
To configure DNS Cache records randomization:

1.
In the hocMIB, set the DnsCacheRecordsRandomization variable.

hoc.DnsCacheRecordsRandomization=Value

Table 78: DNS Cache Records Randomization Values
Value Meaning
0
Disable
Enable
Configuring Pre-resolved Static FQDNs

You can configure up to 10 pre-resolved FQDNs. The StaticHosts table allows configuring FQDNs with static
IP addresses. When a device attempts to reach a FQDN configured in this table, the static IP addresses will
be used instead of resolving the FQDN.
Table 79: Static Host Command Parameters
Parameter
Description
Name
Name (FQDN) of the static host. This name must be unique across the table.
The name only accepts valid FQDNs as defined by RFC 3986 (Uniform Resource Identifier
(URI): Generic Syntax). In addition, strict validation is applied, i.e. the suggested syntax
defined in RFC 1035 is enforced.
IpAddresses
List of static IP addresses associated with the FQDN specified in the StaticHosts.Name
variable.
This list contains numerical IPv4 or IPv6 addresses. IP addresses MUST be separated by
a comma (,).
Index
Index in the table. A value of zero (default) causes automatic selection of the largest
current index value + 1. If the index value already exists in the table, the insertion is
refused. This parameter is optional.
To insert a new static host:

1.
You can use one of the following lines in the CLI or a configuration script:
hoc.InsertStaticHost Index=value Name="hostname" IpAddresses=address,address1
hoc.InsertStaticHost Name="hostname" IpAddresses=address,address1
where:
value can be an integer. This is an optional parameter.
hostname is a unique valid FQDN as define by RFC 3986.
99
address and address1 are numerical IPv4 or IPv6 addresses separated by a comma.
To delete a static host:

1.
In the hocMIB, delete the host name using the Delete command.
You can also use one of the following lines in the CLI or a configuration script:
hoc.StaticHosts.Delete[Index=value]=Delete
where value can be an integer.
Updating the "sysname" or "syslocation"

You can specify the name and location of the Mediatrix unit. This information is for display purposes only and
does not affect the behavior of the unit.
To set the sysname and syslocation parameters:

1.
In the hocMIB, set the system name in the systemName variable.

hoc.systemName="Value"
The value of this variable is also returned by the "sysName" object in SNMPv2-MIB.
2.
Set the system location in the systemLocation variable.

hoc.systemLocation="Value"
The value of this variable is also returned by the "sysLocation" object in SNMPv2-MIB.
100
H A P T E R
12
Interface Parameters
This chapter describes how to set the interfaces of the Mediatrix unit:
How to reserve an IP address in a network server.

Link Connectivity Detection
Partial Reset
Managing interfaces.
PPPoE parameters.
LLDP Configuration
Ethernet Link Configuration
DHCP Server Configuration
Ethernet Connection Speed
Configuring a MTU Value
Reserving an IP Address
Before connecting the Mediatrix unit to the network, Media5 strongly recommends that you reserve an IP
address in your network server if you are using one for the unit you are about to connect. This way, you
know the IP address associated with a particular unit.
Network servers generally allocate a range of IP addresses for use on a network and reserve IP addresses
for specific devices using a unique identifier for each device. The Mediatrix unit unique identifier is its media
access control (MAC) address. You can locate the MAC address as follows:
It is printed on the label located on the bottom side of the unit.

It is stored in the Device Info page of the web interface.
You can take one of the telephones connected to the Mediatrix unit and dial *#*1 on the keypad.
The MAC address of the Mediatrix unit will be stated. This applies to Mediatrix units with FXS
interfaces.
Media5 recommends to reserve an IP address with an infinite lease for each Mediatrix unit on the network.
Link Connectivity Detection

Each Ethernet port of the Mediatrix unit is associated with an Ethernet link. This information is available in the
Ethernet Ports Status section of the Network / Status page. A link has connectivity if at least one of its port
status is not disconnected.
The link connectivity is periodically polled (every 500 milliseconds). It takes two consecutive detections of the
same link state before reporting a link connectivity transition. This avoids reporting many link connectivity
transitions if the Ethernet cable is plugged and unplugged quickly.
Partial Reset
When a partial reset is triggered, the Rescue interface is configured and enabled with:
its hidden IPv4 link configuration values
101
Chapter 12 - Interface Parameters
Interfaces Configuration
its hidden IPv4 address configuration

an IPv6 link-local address on all network links
Hidden values are set by the unit's profile.

Just before the Rescue is configured, all IPv4 network interfaces that could possibly conflict with the Rescue
interface are disabled.
If the BNI Service is stopped when the partial reset occurs, it is started and the above configuration is applied.
Standards Supported
IEEE 802.1Q Virtual Bridged Local Area Networks
RFC 2460: IPv6
RFC 2464: Transmission of IPv6 Packets over Ethernet

Networks
RFC 4193: Unique_local_address
RFC 4291: IP Version 6 Addressing Architecturea
RFC 4443: ICMPv6
RFC 4861: IPv6_neighbor_discovery
RFC 4862: IPv6_stateless_autoconf
a. Site-local address are deprecated.
The Interface Configuration section allows you to add and remove up to 48 network interfaces. By default, this
section contains the following network interfaces:
The Uplink interface, which defines the uplink information required by the Mediatrix unit to
properly connect to the WAN. The Uplink network interface is the IP interface that encapsulates
the following link interface (WAN connection):
eth5 (Mediatrix 3000 Series models)
eth1 (Mediatrix 4400 Series models)
eth1 (Mediatrix LP/4100/C7 Series models), wan for the Mediatrix 4102S
By default, this interface uses the IPv4 DHCP connection type.
The Rescue interface, which defines the address and network mask to use to contact the
Mediatrix unit after a partial reset operation. You cannot delete this interface. See Partial
Reset on page 15 for more details.
The LAN interface IPv4 address and network mask.
The current status of the network interfaces is displayed in the Status page. It allows you to know which
interfaces are actually enabled. Enabled networks are activated when their configured link gets connectivity
and are deactivated as soon as the link connectivity is lost. See Link Connectivity Detection on page 101 for
more details.
The Interfaces Status section of the Status page displays the status of all currently enabled network interfaces,
including interfaces with an invalid configuration or waiting for a response.
When configuring network interfaces, Media5 recommends to have a syslog client properly configured and
enabled in order to receive any message related to the network interfaces behaviour. The interface used to
access the syslog client must also be properly enabled. See Chapter 7 - Syslog Configuration on page 73
for more details on enabling a syslog client.
Caution: Use extreme care when configuring network interfaces, especially when configuring the network
interface used to contact the unit for management. Be careful never to disable or delete the network interface
used to contact the unit. Also be careful to always set the units management interface to be an interface
that you can contact.
102
Note: When performing a partial reset (see Partial Reset on page 15 for more details), the management
interface used for SNMP, CLI and WEB is automatically set to the Rescue interface. In that case, you must
change the Mediatrix unit system management network interface to something other than Rescue. Note that
you must be able to contact the interface you select.
To configure interfaces parameters:

1.
In the web interface, click the Network link, then the Interfaces sub-link.
Figure 33: Network Interfaces Web Page
2.
If you want to add a new interface, enter its name in the blank field in the bottom left of the window,
then click the
button.
The name is case-sensitive. Using the special value All is not allowed.
You can use the following ASCII codes in the network interface name:
49
50
51
52
53
54
55
56
57
65
66
67
68
69
70
71
72
73
74
75
76
1
2
3
4
5
6
7
8
9
A
B
C
D
E
F
G
H
I
J
K
L
77 M
78 N
79 O
80 P
81 Q
82 R
83 S
84 T
85 U
86 V
87 W
88 X
89 Y
90 Z
95 _, underscore
97 a
98 b
99 c
100 d
101 e
102 f
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
g
h
i
j
k
l
m
n
o
p
q
r
s
t
u
v
w
x
y
z
A valid network interface name must be compliant with the following rules:
It must start with a letter
It cannot contain characters other than letters, numbers, underscores.
If your Mediatrix unit contains an invalid interface name created in a previous firmware version
without the validation feature, the invalid interface name will be modified everywhere it appears on
the first restart and a syslog notification will be sent.
You can also delete an existing network interface by clicking the corresponding
cannot delete the Rescue interface.
3.
button. You
In the Interface Configuration section, select the link on which to activate the interface in the Link
column.
103
Mediatrix 3000 Series: You can select between the eth1-4 and eth5 interfaces, as well
as any defined VLANs.
Mediatrix LP/4100/C7 Series: You can select between the eth1 and eth2 interfaces, as
well as any defined VLANs.
Mediatrix 4400 Series: You can select between the eth1 interface and any defined
VLANs.
A VLAN is listed with the following syntax:

Link.VLAN ID
For instance, if you have added VLAN 20 on the interface eth5, it is listed as follows:
eth5.20
Figure 34: VLAN Example
4.
Select the configuration source of the interface information in the Type drop-down menu.
Table 80: Interface Configuration Sources
Source
Description
IPv4
DHCP
The IPv4 address and network mask are provided by querying a DHCP server and
using standard DHCP fields or options. Using the DHCP configuration assumes that
you have properly set your DHCP server with the relevant information. DHCP servers
may provide a list of IP configuration parameters to use. See DHCP Server
IPv4
Static
every time the Mediatrix unit restarts. Use the static configuration if you are not using a
DHCP server/PPP peer or if you want to bypass it.
IPv4
PPPoE
IPv4 over PPP connection, address and network mask are provided by the PPP peer
using IPCP. PPP peers may provide a list of IP configuration parameters to use. See
PPPoE Configuration on page 107 for more details.
IPv6
AutoConf
IPv6 state-less auto-configuration. See IPv6 Autoconfiguration Interfaces on

IPv6
Static
every time the Mediatrix unit restarts. Use the IPv6 static configuration if you are not
using IPv6 stateless or stateful auto-configuration or if you want to bypass it.
Note: If no network is configured in IPv6, the unit does not have any IPv6 address, not even the Link-Local
address. When a network is configured in IPv6, the Link-Local (FE80 ::...) address is automatically created
and displayed in the Network Status information.
5.
If the interface configuration source is IPv4 Static or IPv6 Static, enter the address and network
mask (if applicable) of the network interface in the Static IP address field.
6.
If the interface configuration source is IPv4 Static or IPv6 Static, set the Static Default Router field
with the IP address of the default gateway for the network interface.
7.
Define whether or not the Mediatrix unit should attempt to activate the corresponding network
interface in the Activation drop-down menu.
page.
104
8.

The current network interface information is displayed in the Status page.
Table 81: Network Interface Status
Status
Description
Disabled
The interface is not operational because it is explicitly disabled or the link

interface is unavailable.
Invalid Config
The interface is not operational because its configuration is not valid.
Network Conflict
The interface is configured with an IP address that is already used on the

network.
Link Down
The interface is configured with a link that has no connectivity.
Waiting
Response
The interface is not operational because a response from a peer or server

is required.
Active
The interface is operational.
IPv6 Autoconfiguration Interfaces

When the Type drop-down menu is set to IPv6 Auto-Conf, the network interface is an IPv6 over Ethernet
connection with IP parameters obtained by stateless auto-configuration or stateful (DHCPv6) configuration.
Autoconfiguration of IPv6 address is first initiated using state-less autoconfiguration. Stateful
autoconfiguration is initiated only if one of the following conditions is met:
The router explicitly required stateful autoconfiguration by setting the managed or other flag
of the router advertisement.
No router advertisement was received after 3 router solicitations. RFC 4861 defines the
number of router solicitations to send and the 4 seconds interval between the sent router
solicitations.
Stateless Autoconfiguration
All IPv6 addresses present in the router advertisements are applied to the network interface. Each IPv6
address is assigned a network name based on the configured network name with a suffix in the following
format: ConfiguredNetworkName-XX-Y.
XX is the address scope
GU (Global Unique)
UL (Unique Local)
LL (Link-Local)
Y is a unique ID for the address scope.
Spanning Tree Protocol vs Stateless Autoconfiguration

Many network switches use the Spanning Tree Protocol (STP) to manage Ethernet ports activity. STP uses a
detection timeout before a router advertisement is sent to the Mediatrix unit. The default value for this timeout
is usually 30 seconds. However, when the unit wants to get an IPv6 address in Stateless autoconfiguration,
this timeout is too long and the unit falls into Stateful Autoconfiguration mode before it receives the router
advertisement. This results in the unit receiving a DHCPv6 address.
To solve the issue, check if the default STP detection timeout value in your router can be modified. If so, set
it to a value of 8 s or less. If you cannot modify the timeout value, Media5 recommends to disable the Spanning
Tree Protocol on the network to which the unit is connected.
105
Rescue Interface Configuration
Stateful Autoconfiguration
Stateful autoconfiguration is managed by DHCPv6. The DHCPv6 lease is negotiated according to RFC 3315
with the limitations listed in section 1.5. DHCPv6 may be used to obtain the following information (depending
on the router advertisement flags):
IPv6 addresses (when the router advertisement managed flag is set)

Other configuration (when the router advertisement other flag is set)
If only the other flag is set in the router advertisement, the DHCPv6 client only sends an information request
to the DHCPv6 server, otherwise it sends a DHCPv6 solicit message. If the flags change over time, only the
transitions from not set to set are handled.
Network Interface Priority


by using the CLI
You can prioritize the network interfaces of the Mediatrix unit. In case of address conflicts between two or more
network interfaces, the network interface with the highest priority will remain enabled and the other interfaces
will be disabled. If the priority is the same, only the first enabled network interface will be able to use the IP
address. When a conflict ends, all network interfaces concerned automatically return to an operational state.
The actual status of network interfaces is displayed in the Status web page.
To set the network interface priority:

1.
In the ethMIB, set the networkInterfacesPriority variable with the proper value for the
corresponding interface.
eth.networkInterfacesPriority="Value"
where Value may be any number between 0 and 100.
Rescue Interface Configuration

You can define whether or not the Mediatrix unit should attempt to activate the rescue network interface.
Caution: Please be careful when using this section.
To enable/disable the Rescue interface:

1.
In the Rescue interface section, define whether or not the Mediatrix unit should attempt to activate
the corresponding network interface in the Activation drop-down menu.
Figure 35: Rescue Interface Configuration Section
1
page.
2.
106
PPPoE Configuration
PPPoE Configuration
Standards Supported
RFC 1332 The PPP Internet Protocol Control Protocol

(IPCP)
RFC 1334 PPP Authentication Protocolsa
RFC 1661 The Point-to-Point Protocol (PPP)
RFC 1877 PPP Internet Protocol Control Protocol

Extensions for Name Server Addressesb
RFC 1994 Challenge Handshake Authentication Protocol

(CHAP)
RFC 2516 A Method for Transmitting PPP Over Ethernet

(PPPoE)
a. Section 2 (PAP), section 3 is obsoleted by RFC 1994

b. Supported except for sections 1.2 and 1.4
The PPPoE Configuration section applies only if you have selected the PPPoE connection type in the Interface
Configuration section of the web page.
To configure PPPoE parameters:

1.
In the PPPoE Configuration section, set the name of the service requested to the access
concentrator (AC) when establishing the next PPPoE connection in the Service Name field.
Figure 36: PPPoE Configuration Section
1
2
3
This is used as the Service-Name field of the packet broadcasted to the access concentrators. See
RFC 2516 section 5.1 for details.
The field may be set with any string of characters, with a maximum of 255 characters.
If you leave this field empty, the Mediatrix unit looks for any access concentrator.
2.
3.
Select the authentication protocol to use for authenticating the system to the PPP peer in the
Protocol drop-down menu.
PAP: Use the Password Authentication Protocol.
CHAP: Use the Challenge Handshake Authentication Protocol.
Set the PPP user name and password that identify the system to the PPP peer during the
authentication process in the User Name and Password fields.
Caution: The User Name and Password fields are not accessible if you have the User or Observer access
right. See Users on page 607 for more details.
When connecting to an access concentrator, it may request that the Mediatrix unit identifies itself
with a specific user name and password.
There are no restrictions, you can use any combination of characters.
4.

The current PPPoE information is displayed in the Status page.
107
PPPoE Configuration
PPP Negotiation
When the Mediatrix unit restarts, it establishes the connection to the access concentrator in conformance with
the RFCs listed in PPPoE Configuration on page 107.
When establishing a PPP connection, the Mediatrix unit goes through three distinct phases:
Discovery phase
Authentication phase
Network-layer protocol phase
Discovery Phase
The Mediatrix unit broadcasts the value of the Service Name field.
The access concentrator with a matching service name answers the Mediatrix unit.
If no access concentrator answers, this creates a PPPoE failure error.

If more than one access concentrators respond to the discovery, the Mediatrix unit tries to
establish the PPP connection with the first one that supports the requested service name.
Authenthication Phase
If the access concentrator requests authentication, the Mediatrix unit sends the ID/secret pair configured in
the User Name and Password fields. If the access concentrator rejects the authentication, this creates an
authentication failure error.
Network-Layer Protocol Phase

The Mediatrix unit negotiates an IP address. The requested IP address is the one from the last successful
PPPoE connection. If the Mediatrix unit never connected by using PPPoE (or after a factory reset), it does not
request any specific IP address.
DHCP Client Identifier Presentation


by using the CLI
You can define the method to use to present the value of the Client Identifier (Option 61) field through a DHCP
request. The following values are available:
Table 82: DHCP Client Identifier Presentation Parameters
Parameter
Description
Disabled
The Client Identifier option is not presented in a DHCP request.
MacAscii
The Client Identifier value is presented as the client MAC address in ASCII format. The MAC
address is represented in lowercase.
MacBinary
The Client Identifier value is presented as the client MAC address in binary format.
To define the DHCP client identifier presentation:

1.
In the bniMIB, locate the DhcpClientGroup folder.
2.
Set the dhcpClientIdentifierPresentation variable with the proper presentation.

bni.dhcpClientIdentifierPresentation="Value"
108
LLDP Configuration

Table 83: DHCP Client Identifier Presentation Values
Value
Meaning
100
Disabled
200
MacAscii
300
MacBinary
LLDP Configuration
The Link Layer Discovery Protocol (LLDP) service is used by network devices for advertising their identity,
capabilities, and neighbors on a IEEE 802 local area network, usually wired Ethernet.
The LLDP Configuration section allows you to configure parameters related to LLDP.
To configure LLDP parameters:

1.
In the LLDP Configuration section, set the network interface name on which LLDP should be
enabled in the Network Interface drop-down menu.
Figure 37: LLDP Configuration Section
1
2
3
LLDP cannot be activated on multiple network interfaces simultaneously.

2.
Select the address type to populate the chassis ID device identifier in the Chassis ID drop-down
menu.
Table 84: Chassis ID Parameters
Parameter
MAC Address
Description
The MAC address.
Network Address The IP address (or 0.0.0.0 if DHCP is not obtained yet).
3.
Select whether to enable the LLDP-MED protocol override of the VLAN ID, User Priority and
DiffServ values in the Override Network Policy drop-down menu.
Table 85: Override Network Policy Parameters
Parameter
Description
Enable
The service listens for LLDP advertisements, and overrides the previously
configured VLAN ID, User Priority and DiffServ with the values received.
Disable
The service only publishes its characteristics and configurations by LLDP,

and does not override anything.
The LLDP-MED (Media Endpoint Discovery) protocol is an enhancement of LLDP.

4.

The current LLDP information is displayed in the Status page.
109

Standards Supported
IEEE 802.1X-2001 Port Based Network Access Controla
IEEE 802.1X-2004 Port Based Network Access Control
a. Partially compliant (Section 5.1.d): Supports supplicant role only with EAP-TLS authentication.
The Ethernet Link Configuration section allows you to configure the MTU as well as IEEE 802.1X
authentication.
To configure Ethernet link parameters:

1.
In the Ethernet Link Configuration section, set the MTU field of a specific Ethernet link with a proper
value.
Figure 38: Ethernet Link Configuration Section
1
The Maximum Transmission Unit (MTU) is a parameter that determines the largest packet than can
be transmitted by an IP interface (without it needing to be broken down into smaller units). Each
interface used by TCP/IP may have a different MTU value specified. See Appendix C - Maximum
Transmission Unit (MTU) on page 661 for more details on MTU.
The range is from 576 to 1500 bytes. All VLAN connections use the MTU size configured on their
related Ethernet link.
Note: The MTU value applied for a PPPoE connection is the smallest of the value negotiated with the server
and the value configured here.
2.
Define the IEEE 802.1x authentication protocol activation to use for a specific Ethernet link in the
corresponding 802.1x Authentication drop-down menu.
802.1X Authentication is a tag optionally added to the Ethernet frame header to specify the support
of the IEEE 802.1X Authentication. It allows getting authorization and access to secured network(s).
Table 86: 802.1x Authentication Parameters
Parameter
3.
Description
Disable
The IEEE 802.1x authentication protocol is disabled on the Ethernet link interface.
Enable
The IEEE 802.1x authentication protocol using the EAP-TLS authentication

method is enabled on the Ethernet link to get an access, through an IEEE 802.1x
EAP-TLS authenticator (such as an IEEE 802.1x capable network device), to
secured network(s). The Ethernet link interface remains always 'UP' whatever the
result of the IEEE 802.1x authentication.
Set the username used to authenticate each Ethernet link interfaces during the IEEE 802.1x EAPTLS authentication process in the corresponding EAP Username field.
This parameter is used only when the IEEE 802.1x authentication is enabled (802.1x Authentication
drop-down menu set to Enabled).
4.
110
Define the IEEE 802.1x level of validation used by the device to authenticate the IEEE 802.1x EAPTLS peer's certificate.
EAP 802.1x Configuration
This parameter also controls the criteria used to select the host certificate sent during the
authentication handshakes.
Table 87: 802.1x Certificate Validation Parameters
5.
Parameter
Description
No
Validation
No validation is performed on the peers certificate. Authentication with the peer is

attempted even if the system time is not synchronized. If more than one host
certificate is configured for an EAP-TLS usage, the one with the latest expiration
date is used.
Trusted
And Valid
Allow a connection to the network by validating if the authentication peers

certificate is trusted and valid. The IEEE 802.1x authentication is attempted only if
the system time is synchronized. If more than one host certificate is configured for
an EAP-TLS usage, the one that is currently valid and with the latest expiration
date is used.

The current status of the network interfaces is displayed in the Status page. It allows you to know
which interfaces are actually enabled.
Table 88: Ethernet Link Interface State
State
Description
Disconnected
The link interface is physically disconnected.
Up
The link interface is physically connected and considered as usable by network

interface(s).
EAP 802.1x Configuration

The EAP 802.1x Configuration section allows you to set the IEEE 802.1x version to be used by the unit.
To configure the IEEE 802.1x version parameter:

1.
In the EAP 802.1x Configuration section, set the IEEE 802.1x version from the EAP 802.1x Version
drop-down menu.
Figure 39: EAP 802.1x Configuration Section
1
Table 89: EAP 802.1x Version Parameters

Parameter
2.
Description
Version 2001
Version 2004
111

Standards Supported
RFC 2131 Dynamic Host Configuration Protocola
RFC 2132 DHCP Options and BOOTP Vendor Extensionsb
RFC 3315: DHCPv6c
a. Supports the client side of the protocol

b. Only sections 3.3, 3.5, 3.8 and 8.3
c. Supports the client side of the protocol
Note: This section applies only if you are using the DHCP connection type (Interfaces Configuration on
page 102).
DHCP servers generally allocate a range of IP addresses for use on a network and reserve IP addresses for
specific devices using a unique identifier for each device. The Mediatrix unit unique identifier is its media
access control (MAC) address.
You can locate the MAC address as follows:
on the label located on the bottom side of the unit.

in the System > Information web page
you can dial the following digits on a telephone connected to the Mediatrix unit:
*#*1
The Mediatrix unit answers back with its MAC address. This applies to units with FXS interfaces.
See General POTS Configuration on page 162 for more details.
Media5 recommends to reserve an IP address with an infinite lease for each Mediatrix unit on the network.
DHCP Negotiation
The DHCP lease is negotiated according to RFC 2131 (supports the client side of the protocol) and RFC 2132
(only sections 3.3, 3.5, 3.8 and 8.3). The following parameters are set
Table 90: DHCP Parameters
DHCP Parameter
Host Name (option 12)
Value
Set according to the Host Name parameter of the Network > Host
page (Host Configuration on page 91). This option cannot be empty
according to RFC 2132. If the Host Name parameter is empty, the
DHCP option 12 is not sent.
Vendor Class Identifier (option 60) Set according to the System Description parameter of the System >
Information page.
Client identifier (option 61)
112
Set according to MAC Address parameter of the System >

Information.


by using the CLI
You can set the speed and duplex of the Ethernet connection of the Mediatrix unit. The following values are
available:
Table 91: Ethernet Ports Speed and Duplex Supported
Parameter
Description
Auto
Automatic negociation of speed and duplex.
Half10
10 Mbit/s Half-duplex.
Full10
10 Mbit/s Full-duplex.
Half100
100 Mbit/s Half-duplex.
Full100
100 Mbit/s Full-duplex.
A half-duplex connection refers to a transmission using two separate channels for transmission and reception,
while a full-duplex connection refers to a transmission using the same channel for both transmission and
reception.
If unknown, set the variable to Auto so that the Mediatrix unit can automatically detect the network speed.
Caution: Whenever you force a connection speed / duplex mode, be sure that the other device and all
other intermediary nodes used in the communication between the two devices have the same configuration.
See Speed and Duplex Detection Issues on page 114 for more details.
The current speed and duplex configuration is displayed in the Network > Status page under the Ethernet
Ports Status section.
To set the Ethernet connection speed and duplex:

1.
In the ethMIB, locate the portsTable folder.
2.
Set the portsSpeed variable with the proper Ethernet speed and duplex.
eth.portsSpeed="Value"

Table 92: Ethernet Ports Speed and Duplex Values
Value
Meaning
100
Auto
200
Half10
300
Full10
400
Half100
500
Full100
113
Speed and Duplex Detection Issues

There are two protocols for detecting the Ethernet link speed:
An older protocol called parallel detection.

A more recent protocol called auto-negotiation (IEEE 802.3u).
The auto-negotiation protocol allows to detect the connection speed and duplex mode. It exchanges
capabilities and establishes the most efficient connection. When both endpoints support the auto-negotiation,
there are no problems. However, when only one endpoint supports auto-negotiation, the parallel detection
protocol is used. This protocol can only detect the connection speed; the duplex mode cannot be detected. In
this case, the connection may not be established.
The Mediatrix unit has the possibility to force the desired Ethernet link speed and duplex mode by disabling
the auto-negotiation and selecting the proper setting. When forcing a link speed at one end, be sure that the
other end (a hub, switch, etc.) has the same configuration. To avoid any problem, the link speed and duplex
mode of the other endpoint must be exactly the same.
114
H A P T E R
13
VLAN Parameters
This chapter describes how to create and manage dynamic VLANs on the Mediatrix unit.
VLAN Configuration
A virtual LAN is a network of computers that behave as if they are connected to the same wire even though
they may actually be physically located on different segments of a LAN. You can add VLANs on the Ethernet
links of the Mediatrix unit. You can currently add or manage up to a maximum of 16 VLANs.
Caution: When working with VLANs, take care not to cut your access to the unit, for instance by putting
the Uplink on a VLAN to which your PC does not have access and then setting the management interface
to Uplink.
To add a VLAN:
1.
In the web interface, click the Network link, then the VLAN sub-link.
Figure 40: Network VLAN Web Page
2.
Select the Ethernet link over which the VLAN interface is built in the Link drop-down menu.
3.
Set the VLAN ID used by the VLAN interface in the Id field.

This is a 12 bit field in the 802.1Q tag carrying an ID that differentiates frames containing this ID
from frames containing different IDs or no 802.1Q tag at all.
To systems supporting Ethernet 802.1Q, frames containing the same VLAN ID are considered as
belonging to the same virtual LAN, and frames containing different IDs are considered as not
belonging to the same virtual LAN, even though they use the same physical LAN.
4.
Click on the
button.
5.
Set the default user priority value the interface uses when tagging packets in the Default User
Priority field.
You can also set specific service class values in the Quality of Service page. See Chapter 14 Local QoS (Quality of Service) Configuration on page 117 for more details.
6.

You can also delete an existing VLAN by clicking the corresponding
button.
Once you have added a VLAN, you must select this VLAN on an interface to activate it. You can do
so in the Link column of the Interface Configuration section in the Network > Interfaces page
(Interfaces Configuration on page 102). The VLAN is listed with the following syntax:
Link.VLAN ID
For instance, if you have added VLAN 20 on the interface eth5, it is listed as follows:
eth5.20
115
Chapter 13 - VLAN Parameters
VLAN Configuration
Figure 41: VLAN Example
116
H A P T E R
14
Local QoS (Quality of Service)

Configuration
This chapter describes how to configure the local QoS parameters. The local QoS tags packets sent from the
Mediatrix unit. It does not process nor classify packets coming from the network.
Introduction
QoS (Quality of Service) features enable network managers to decide on packet priority queuing. The Dgw
v2.0 application supports the Differentiated Services (DS) field and 802.1q taggings.
The Dgw v2.0 application supports the Real Time Control Protocol (RTCP), which is used to send packets to
convey feedback on quality of data delivery.
The Dgw v2.0 application does not currently support the Voice Band Data service class. It also does not
support RSVP (Resource Reservation Protocol).
Differentiated Services (DS) Field

Standards Supported
RFC 2475: An Architecture for Differentiated Services
Differentiated Services (DiffServ, or DS) is a protocol for specifying and controlling network traffic by class so
that certain types of traffic for example, voice traffic, which requires a relatively uninterrupted flow of data,
might get precedence over other kinds of traffic.
What are Differentiated Services?

Differentiated Services avoids simple priority tagging and depends on more complex policy or rule
statements to determine how to forward a given network packet. An analogy is made to travel services, in
which a person can choose among different modes of travel train, bus, airplane degree of comfort,
the number of stops on the route, standby status, the time of day or period of year for the trip, and so
forth.
For a given set of packet travel rules, a packet is given one of 64 possible forwarding behaviors known
as per hop behaviors (PHBs). A six-bit field, known as the Differentiated Services Code Point (DSCP), in
the Internet Protocol header specifies the per hop behavior for a given flow of packets. The DS field
structure is presented below:
0
1
2
3
4
5
6
7
+---+---+---+---+---+---+---+---+
| DSCP
| CU
|
+---+---+---+---+---+---+---+---+
MSB
LSB
DSCP: Differentiated Services CodePoint.
CU: Currently Unused. The CU bits should always be set to 0.
For both signalling and media packets, the DSCP field is configurable independently. The entire DS field
(TOS byte) is currently configurable.
117
Chapter 14 - Local QoS (Quality of Service) Configuration
Differentiated Services (DS) Field
DiffServ replaces the first bits in the ToS byte with a differentiated services code point (DSCP). It uses the
existing IPv4 Type of Service octet.
It is the network administrators responsibility to provision the Mediatrix unit with standard and correct values.
To configure the Mediatrix unit DiffServ value:

1.
In the web interface, click the Network link, then the QoS sub-link.
Figure 42: Network QoS Web Page
2
3
2.
Set the default Differentiated Services value used by the unit for all generated packets in the Default
DiffServ (IPv4) field.
You can override this value by setting specific service class values. See Specific Service Class
This 8-bit value is directly set in the TOS field (2nd byte) of the header of transmitted IPv4 packets,
allowing you to use either DiffServ or TOS mapping.
The DiffServ value is 1 octet scalar ranging from 0 to 255. The DSCP default value should be
101110. This results in the DS field value of 10111000 (184d). This default value would result in a
value of 101 precedence bits, low delay, high throughput, and normal reliability in the legacy IP
networks (RFC 791, RFC 1812). Network managers of legacy IP networks could use the abovementioned values to define filters on their routers to take advantage of priority queuing. The default
value is based on the Expedited Forwarding PHB (RFC 2598) recommendation.
Note: RFC 3168 now defines the state in which to set the two least significant bits in the TOS byte. On the
other hand, this RFC only applies to TCP transmissions and the bits are thus set to 0 in the Mediatrix unit.
This has the following effects:
The TOS values for UDP packets are the same as in the MIB.
The TOS values for TCP packets are equal to the closest multiple of 4 value that is not greater than the
value in the MIB.
You can find references on DS field under the IETF working group DiffServ. For more information,
please refer to the following RFC documents:
3.
Definition of the Differentiated Services Field (DS Field) in the IPv4 and IPv6 Headers
(RFC 2474)
An Architecture for Differentiated Services (RFC 2475)
Assured Forwarding PHB Group (RFC 2597)
An Expedited Forwarding PHB (RFC 2598)
Set the Default Traffic Class value used by the unit for all generated IPv6 packets in the Default
Traffic Class (IPv6) field.
Specific service class values may be set in the Service Classes table. See Specific Service Class
The 8-bit Traffic Class field in the IPv6 header is available for use by originating nodes and/or
forwarding routers to identify and distinguish between different classes or priorities of IPv6 packets.
4.
118
IEEE 802.1q
IEEE 802.1q
The 802.1q standard recommends the use of the 802.1q VLAN tags for Ethernet frames traffic prioritization.
VLAN tags are 4-byte headers in which three bits are reserved for priority indication. The values of the priority
bits shall be provisioned.
The 802.1q standard comprises the 802.1p standard.
It is the network administrators responsibility to provision the Mediatrix unit with standard and correct values.
To enable the IEEE 802.1q user priority configuration:

1.
In the Ethernet 802.1Q Tagging Configuration section of the QoS page, select Enable in the Enable
column for each interface on which you want to enable user priority tagging.
Figure 43: Ethernet 802.1Q Tagging Configuration Section
1
The VLAN ID part of the 802.1Q tag is always set to 0.

2.
Set the default user priority value each interface uses when tagging packets in the Default User
Priority column.
You can override each value by setting specific service class values. See Specific Service Class
The user priority is a 3 bit field in the 802.1Q tag that carries a priority value ranging from 0 to 7 and
may be used by switches to prioritize traffic. The 802.1q default priority value should be 6 for both
signalling and media packets.
3.
Specific Service Class Configuration

You can override the default value set in the DiffServ and 802.1q sections for each service class of the
Mediatrix unit:
Signalling
Voice
T.38
IP Sync (IP Sync is not available in IPv6)
To set specific service class values:

1.
In the Service Class Configuration section of the QoS page, set a specific DiffServ value for each
class in the DiffServ (IPv4) column.
Figure 44: Service Class Configuration Section
1
See Differentiated Services (DS) Field on page 117 for more details.
2.
Set the Default Traffic Class value used in IPv6 packets for each class in the Traffic Class (IPv6)
column.
119
Network Traffic Control Configuration
The 8-bit Traffic Class field in the IPv6 header is available for use by originating nodes and/or
forwarding routers to identify and distinguish between different classes or priorities of IPv6 packets.
3.
Set a specific user priority for each class in the User Priority column.
See IEEE 802.1q on page 119 for more details.
4.

You can apply a bandwidth limitation on the network interfaces. The limitations are applied on raw data on the
physical link and not only on the payload of the packets. All headers, checksums and control bits (TCP, IP,
CRC, etc.) are considered in the actual bandwidth.
A bandwidth limitation is applied on a physical link and not on a high-level network interfaces. All high-level
network interfaces (including VLANs) using the same physical link are affected by a configured limitation. This
limitation is applied egress only (outgoing traffic).
If the NTC service is stopped, this section is not displayed in the QoS page. See Chapter 4 - Services on
page 55 on information on how to start the service. Starting the NTC service enables Traffic Shaping even if
bandwidth limitation is disabled.
Bandwidth limitation is an average of the amount of data sent per second. It is thus normal that the unit sends
a small burst of data after a period of silence.
Note that the NTC service sends packets on the physical link according to their respective priorities as
described below. Lower priority packets are dropped first.
Table 93: Physical Link Priorities
Priority
Description
Highest priority. Packets originating from the unit with 802.1p priority set to 7.
Packets originating from the unit with 802.1p priority set to 6.
Lowest priority. Packets originating from another link interface (routed packets).
Packets that exceed the defined bandwidth are eventually dropped (when the buffers are exceeded). This
implies that data bursts can suffer a slight amount of packet loss. The different codecs configured and the
desired number of simultaneous channels should be taken into account when choosing a bandwidth limit to
prevent call drops, choppy voice or inconstant ptime. The NTC service can impact the execution of other
processes if the number of packets to process is too high. (High traffic and/or low limit).
To set network traffic control parameters:

1.
In the Network Traffic Control Configuration section of the QoS page, set the corresponding Egress
Limit field with the egress bandwidth limitation for the selected link interface.
The range is from 64 to 40960 kilobits per second.
The value 0 means no bandwidth limitation and no prioritization.
120
This value must be set according to the upstream bandwidth limit of the network on this link. Set to
0 (disable) if the network bandwidth exceeds 40960 kbps or if it exceeds the effective limit of this
device.
Figure 45: Network Traffic Control Configuration Section
1
2.
121
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H A P T E R
15
Local Firewall Configuration

This chapter describes how to configure the local firewall parameters.
Setting the default policy

Creating/editing a firewall rule
Moving a firewall rule
Deleting a firewall rule
Disabling the local firewall
Managing the Local Firewall

The local firewall allows you to dynamically create and configure rules to filter packets. The traffic is analyzed
and filtered by all the rules configured.
Note: The Mediatrix units local firewall settings do not support IPv6. See IPv4 vs. IPv6 on page 87 for
more details.
Since this is a local firewall, rules apply only to incoming packets with the unit as destination.
Incoming packets for an IP communication established by the unit are always accepted (Example : If the
Mediatrix unit sends a DNS request, the answer will be accepted).
Rules priority is determined by their position in the table.
The maximum number of rules allowed in the configuration is 20.
Caution: Enabling the local firewall and adding rules has an impact on the Mediatrix units overall
performance as the firewall requires additional processing. The more rules are enabled, the more overall
performance is affected. Furthermore, Media5 recommends to use a 30 ms packetization time when the
firewall is enabled (instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels
available on the unit, especially with the Mediatrix 3632 / 4404 / 4124 / LP24 models.
Partial Reset
When a partial reset is triggered and the firewall is enabled, the configuration is rolled back if it was being
modified. A new rule is then automatically applied in the firewall to allow access to the 'Rescue' interface.
However, if the firewall is disabled, the configuration is rolled back but no rule is added.
Setting the Default Policy

The default policy defines the action the Mediatrix unit must take when a packet does not match any rule.
123
Chapter 15 - Local Firewall Configuration
To set the default policy:

1.
In the web interface, click the Network link, then the Local Firewall sub-link.
Figure 46: Network Local Firewall Web Page
2.
In the Local Firewall Configuration section, define the Default Policy drop-down menu.
Table 94: Default Policy Parameters
Parameter
Description
Accept
Lets the packet through.
Drop
Drops the packet without any notification.
Caution: Make sure there are some rules with the Action parameter set to Accept in the local firewall
BEFORE applying changes that set the default policy to Drop. If you do not comply with this warning, you
will lose contact with the unit and a partial or factory reset will be required.
Setting the default policy to Drop or adding a rule automatically enables the local firewall. Enabling
the local firewall may have a negative impact on performance.
Creating/Editing a Firewall Rule

The web interface allows you to create a firewall rule or modify the parameters of an existing one.
To create or edit a firewall rule:

1.
In the Local Firewall Rules section of the Local Firewall page, do one of the following:
If you want to add a rule before an existing entry, locate the proper row in the table and
click the
button of this row.
If you want to add a rule at the end of the existing rows, click the
bottom right of the section.
button at the
Figure 47: Local Firewall Rules Section

2
Note: When you add a new rule, edit an existing rule, or delete a rule, you can see a yellow Yes in the
Config Modified section at the top of the window. It warns you that the configuration has been modified but
not applied (i.e., the Firewall section of the Status page differs from the Local Firewall). The Local Firewall
sub-menu is a working area where you build up a local firewall configuration. While you work in this area,
the configured parameters are saved but not applied (i.e., they are not used to filter incoming packets). The
yellow Yes flag warns you that the configuration has been modified but is not applied.
124
2.
Set the current activation state for this rule in the corresponding Activation drop-down menu.
Table 95: Firewall Rule Activation State Parameters
Parameter
Description
Enable
This rule is active in the firewall.
Disable
This rule is not in the firewall.
Only enabled rules may be applied to the firewall.

3.
Enter the source address of the incoming packet in the corresponding Source Address field.
Use one of the following syntax:
Table 96: Source Address Parameters
Parameter
Description
address[/mask]
Can either be a network IP address (using /mask) or one of the host

IP addresses. The mask must be a plain number specifying the
number of binary 1s at the left side of the network mask (a mask of
24 specifies a network mask of 255.255.255.0).
networkInterfaceName
/
The value must already exist in the Interface Configuration table

(see Interfaces Configuration on page 102 for more details). The
interface name is case sensitive, hence it must be entered properly.
If the specified network interface is disabled or removed, the rule is
automatically disabled thus removed from the firewall. When the
network interface is enabled or added back, the rule is automatically
enabled and applied in the firewall.
Note: It is mandatory to use the suffix / to indicate that the network

address of this interface is used instead of the host address.
Leaving the default empty string matches any address.
4.
Enter the source port of the incoming packet in the corresponding Source Port field.
You can enter a single port or a range of ports. In the case of a range of ports, use the following
format:
port[-port]
Leaving the default empty string means that no filtering is applied on the source port, thus matching
any port.
This parameter is only effective when the Protocol drop-down menu is set to TCP or UDP (see Step
7).
5.
Enter the destination address of the incoming packet in the corresponding Destination Address
field.
Table 97: Source Address Parameters
Parameter
address
Description
Must be one of the host IP addresses. Specifying a network address
is invalid since this is a local firewall.
125
Table 97: Source Address Parameters (Continued)

Parameter
Description
networkInterfaceName
The host address of this interface is used. The value must already
exist in the Interface Configuration table (see Interfaces
Configuration on page 102 for more details). The interface name is
case sensitive, hence it must be entered properly.
If the specified network interface is disabled or removed, the rule is
automatically disabled thus removed from the firewall. When the
network interface is enabled or added back, the rule is automatically
enabled and applied in the firewall.

6.
Enter the destination port of the incoming packet in the corresponding Destination Port field.
You can enter a single port or a range of ports. In the case of a range of ports, use the following
format:
port[-port]
Leaving the default empty string means that no filtering is applied on the destination port, thus
matching any port.
7).
7.
Select the protocol of the incoming packet to filter in the corresponding Protocol drop-down menu.
Table 98: Firewal Rule Protocol Parameters
Parameter
8.
Description
All
Matches packets using any protocols.
TCP
Matches only TCP packets.
UDP
Matches only UDP packets.
ICMP
Matches only ICMP packets.
Select the action to take in the corresponding Action field.

Table 99: Firewal Rule Action Parameters
Parameter
Description
Accept
Reject
Sends back an ICMP port unreachable in response to the matched packet. The
packet is then dropped.
Drop
Note that if a connection is already established before creating a rule that rejects it, this connection
stays active despite the rule applied.
9.
Click the Apply button to activate the enabled rules.

The current enabled rules applied are displayed in the Network > Status web page, Firewall section,
which contains the active configuration in the firewall. You can also see that the yellow Config
Modified Yes flag is cleared.
Moving a Firewall Rule

The firewall rules sequence is very important because rules priority is determined by their position in the table.
If you want the unit to try to match one rule before another one, you must put that rule first.
126
Disabling the Local Firewall
To move a rule up or down:

1.
Either click the
or
arrow of the rule you want to move until the entry is properly located.
2.
Click the Apply button to update the Network > Status web page.
Deleting a Firewall Rule

You can delete a rule from the table in the web interface.

1.
Click the
button of the rule you want to move.
2.

When the local firewall is enabled, it has an impact on the Mediatrix units overall performance as the firewall
requires CPU power. You can disable the firewall if you do not need it, thus not impacting performance.
To disable the firewall:
1.
In the Local Firewall Configuration section, set the default policy to Accept with no rules in the local
firewall.
2.
Restart the Mediatrix unit.
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128
H A P T E R
16
IP Routing Configuration
This chapter describes how to configure the IP Routing parameters of the Mediatrix unit.
IPv4 Forwarding
Creating/editing an IP routing rule
Moving an IP routing rule
Deleting an IP routing rule
IP routing examples
Managing IP Routing
The IP Routing service allows the Mediatrix unit to perform advanced routing based on the packets criteria
(source IP address and source Ethernet link), which allows the packet to be forwarded to a specific network.
You can create up to four advanced IP routes.
Note: The Mediatrix units IP Routing settings do not support IPv6. See IPv4 vs. IPv6 Availability on
Packets matching a list of criteria should1 use advanced IP routes instead of routes present in the main routing
table of the unit.
IP Routing works together with the following services:
Network Firewall (Chapter 17 - Network Firewall Configuration on page 137)

NAT (Chapter 18 - NAT Configuration on page 143)
DHCP server (Chapter 19 - DHCP Server Settings on page 151)
Network Traffic Control (Network Traffic Control Configuration on page 120)
These services must be properly configured.

When the IP Routing service is started, IP routing is activated even if there is no configured rule (the Mediatrix
unit will forward received packets). If the IP Routing service is stopped, IP forwarding is disabled, this tab is
greyed out and the parameters are not displayed. See Chapter 4 - Services on page 55 on information on
how to start the service.
Caution: Enabling the IP routing service and adding rules has an impact on the Mediatrix units overall
performance as IP routing requires additional processing. The more rules are enabled, the more overall
performance is affected. Furthermore, Media5 recommends to use a 30 ms packetization time when IP
routing is enabled (instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels
available on the unit, especially with the Mediatrix 3632 / 4404 / 4104 models.
IPv4 Forwarding
IPv4 forwarding allows you to control the IPv4 forwarding feature and the Advanced IP Routes. When set to
Enabled, IPv4 Forwarding is enabled and the Advanced IP Routes are applied. When set to Disabled, IPv4
Forwarding is disabled and the Advanced IP Routes are not applied (the Advanced IP Routes section of the
IP Routing page is disabled).
1. A packet matching a route uses the custom routing table first and then the main routing table if no route in the custom routing table was
able to send the packet to the desired destination IP address.
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Chapter 16 - IP Routing Configuration
Managing IP Routing
To manage IPv4 forwarding:

1.
In the web interface, click the Network link, then the IP Routing sub-link.
2.
In the IP Routing Configuration section of the IP Routing page, define whether or not IPv4
forwarding is enabled by setting the IPv4 Forwarding drop-down menu accordingly.
Figure 48: IPv4 Forwarding Configuration Section
3.
Click the Submit & Apply button to update the Network > Status web page.
Creating/Editing an IP Routing Rule

The web interface allows you to create a routing rule or modify the parameters of an existing one.
To create or edit a routing rule:

1.
In the Advanced IP Routes section of the IP Routing page, do one of the following:
click the
button of this row.
button at the
Note: When you add a new rule, edit an existing rule or delete a rule, you can see a yellow Yes in the Config
Modified section at the top of the window. It warns you that the configuration has been modified but not
applied (i.e., the Advanced IP Routes section of the Status page differs from the IP Routing page). The IP
Routing sub-menu is a working area where you build up a routing configuration. While you work in this area,
the configured parameters are saved but not applied (i.e., they are not used to route packets). The yellow
Yes flag warns you that the configuration has been modified but is not applied.
Figure 49: Advanced IP Routes Section
2
2.
Set the required state for this rule in the corresponding Activation drop-down menu.
Table 100: IP Routing Rule Activation Parameters
Parameter
Description
Enable
Activates this route.
Disable
Does not activate this route.
Only enabled rules may be applied to the routing table.

3.
130
Enter the source IP address criteria an incoming packet must have to match this rule in the Source
Address field.
Managing IP Routing
Use the following syntax:

Table 101: Source Address Syntax
Syntax
Description
Can either be a network IP address (using /mask) or one of the
host IP addresses. The mask must be a plain number specifying
the number of binary 1s at the left side of the network mask (a
mask of 24 specifies a network mask of 255.255.255.0). For
instance:
address[/mask]
networkInterfaceName[/]
192.168.0.11
192.168.1.0/24

(see Interfaces Configuration on page 102 for more details).
The interface name is case sensitive, hence it must be entered
properly.
If the specified network interface is disabled or removed, the rule
is automatically disabled thus removed from the NAT. When the
network interface is enabled or added back, the rule is
automatically enabled and applied. For instance:
Lan1/ (Lan1 network address)
Note: It is mandatory to use the suffix / to indicate that the

network address of this interface is used instead of the host
address.
When left empty, any source address matches this rule.
4.
Enter the source link criteria an incoming packet must have to match this rule in the Source Link
field.
When left empty, packets received on any link match this rule.
5.
Select the network on which the packet is forwarded in the Forward to Network drop-down menu.
6.
Click the Submit & Apply button to activate the enabled rules.
The current applied rules applied are displayed in the Network > Status web page, Advanced IP
Routes section, which contains the active configuration of the custom routing tables. You can also
see that the yellow Config Modified Yes flag is cleared.
Note: You can revert back to the configuration displayed in the Status web page at any time (including the
disabled rules) by clicking the Rollback button at the bottom of the page. All modified settings in the IP
Routing page will be lost.
Moving an IP Routing Rule

The IP routing rules sequence is very important because only one forwarding rule is applied on a packet. Rules
priority is determined by their position in the table. If you want the unit to try to match one rule before another
one, you must put that rule first.

1.
Either click the
or
2.
Deleting an IP Routing Rule

131
Managing IP Routing

1.
Click the
2.
Static IPv4 Routes

You can add or delete static IPv4 routes in the Mediatrix unit. A "static" route means that the route is configured
manually by the administrator. It can be configured through two different methods: through unit provisioning
or through a DHCP server (DHCPv4 Classless Static Route Option on page 133).
To manage static IPv4 routes:

1.
In the Static IP Routes section of the IP Routing page, do one of the following:
If you want to add a route, click the
button at the bottom of the section.
If you want to delete an existing route, click the

move.
button of the route you want to
Figure 50: Static IP Routes Section

2
This section is not available if IPv4 forwarding is disabled.

2.
Specify the destination IP address criteria that an outgoing packet must have to match this route in
the corresponding Destination field.
The supported format for the destination is:
IP address[/mask]
When specifying a network as a destination, it is mandatory to use the "/" format.

The mask must be a plain number specifying the number of binary 1s at the left side of the network
mask (a mask of 24 specifies a network mask of 255.255.255.0). For instance:
3.
192.168.1.5 specifies an IP address as the destination.
192.168.1.0/24 specifies a network address as the destination.
Select the output link (interface) name in the corresponding Link drop-down menu.
When left empty, the link is selected automatically according to the information already present in
the routing table.
4.
Define the IP address of the gateway used by the route in the corresponding Gateway field.
5.
The current routes available are displayed in the Network > Status web page, IPv4 Routes section.
This section identifies the entity that installed the route.
Table 102: IPv4 Routes Protocol
Protocol
132
Description
Dhcp
The route was installed dynamically by the DHCP protocol.
Static
The route was installed by the administrator of the unit.
Kernel
The route was installed by the operating system.
Other
The route was installed by another entity.
Managing IP Routing
DHCPv4 Classless Static Route Option

Standards Supported
RFC 3442: The Classless Static Route Option for Dynamic

Host Configuration Protocol (DHCP) version 4

by using the CLI
You can define whether or not the Classless Static Route Option is enabled. Static routes can be configured
through the Classless Static Route Option for DHCPv4 (option 121) defined in RFC 3442.
If a static route to 0.0.0.0/0 is received through option 121 while a default router is also specified (see Default
Gateway Configuration on page 93 for more details), the route received through option 121 has priority.
Table 103: DHCPv4 Classless Static Route Option Parameters
Parameter
Description
Request
The device requests the Classless Static Route Option 121.
None
Routes received from the DHCP server are ignored.
To define whether or not the Classless Static Route Option is enabled:

1.
2.
Set the dhcpClientClasslessStaticRouteOption variable with the proper behaviour.

bni.dhcpClientClasslessStaticRouteOption="Value"

Table 104: DHCPv4 Classless Static Route Option Values
Value
Meaning
100
None
200
Request
DHCPv4 User Class Route Option

Standards Supported
RFC 3004 -The User Class Option for DHCP

by using the CLI
You can define a list of user classes to enable the User Class Route Option. The list of user classes is sent
using option 77. Hexadecimal values are supported using the \xXX format where XX is the hexadecimal
value. When the variable is empty, user class option is not sent.
To define a list of user classes:
1.
2.
Set the dhcpClientUserClass variable with the list of user classes.

133
Network Configuration Examples
bni.dhcpClientUserClass="Value"
where Value may be one or more user classes.

User Class items are separated by a comma and items must not be empty.

The following are two examples of advanced IP routing that can be accomplished with the Mediatrix unit.
Forward Packets from the Lan1 Network to the Uplink Network with NAT
1.
Create an IP routing rule so that the packets are routed (Managing IP Routing on page 129).
Source IP: Lan1/

Remove this criterion if you want to forward all packets received on the lan link.
2.
Source Link: lan2
Destination Network: Uplink
Click Submit & Apply.
Create a NAT rule so that the forwarded packets going on the Uplink network use the correct source
IP address (Creating/Editing a Source NAT Rule on page 143).
3.
4.
Type: SNAT
Source IP: Lan1/
Protocol: All
New Address: Uplink
Create a Network Firewall rule to let established or related packets go through the unit (if the default
policy is not set to Accept) (Managing the Network Firewall on page 137).
Connection State: Established or Related
Action: Accept
Create a Network Firewall rule to let the packets pass from the Lan1 network to the Uplink network
(if the default policy is not set to Accept). All response packets will be accepted by the previous rule
(Managing the Network Firewall on page 137).
Source IP: Lan1/

Use additional rules or set the default policy to Accept if you want to forward packets
received on the lan link with a source address that does not match the Lan1 subnet.
Connection State: New
Action: Accept
Configure Port Forwarding for a Web Server Located on the LAN

1.
Make sure the IP Routing service is started (to activate IP forwarding).
2.
Create a NAT rule (Creating/Editing a Destination NAT Rule on page 147).

This will change the destination of an HTTP packet originally destined to the Mediatrix unit with the
IP:Port of the Web server on the LAN side (to make sure the unit does not process the packet but
forwards it on the Lan1 network).
Type: DNat
2. The source link name may vary depending on the unit model you have.
134
3.
Destination IP: Uplink
Destination Port: 8080
Protocol: TCP
New Address: 192.168.0.11:80 (IP:Port of the Web server on the LAN side)
Create a NAT rule (Creating/Editing a Source NAT Rule on page 143).

This will change the source IP address of the packet before it is sent on the Lan1 network (to make
sure the Web browser can reply correctly to the request).
4.
5.
Type: SNat
Destination IP: 192.168.0.11
Protocol: TCP
New Address: Lan1
Create a Network Firewall rule to let established or related packets go through the unit (if the default
policy is not set to Accept) (Managing the Network Firewall on page 137).
Connection State: Established or Related
Action: Accept
Create a Network Firewall rule to let the packets pass from the Uplink network to the Lan1 network
(if the default policy is not set to Accept). All response packets will be allowed by the previous rule
(Managing the Network Firewall on page 137).
Destination IP: 192.168.0.11
Protocol: TCP
Action: Accept
135
136
H A P T E R
17
Network Firewall Configuration

This chapter describes how to configure the network firewall parameters.
Setting the default policy

Creating/editing a firewall rule
Moving a firewall rule
Deleting a firewall rule
Disabling the network firewall
Managing the Network Firewall

The network firewall allows dynamically creating and configuring rules to filter packets forwarded by the unit.
Since this is a network firewall, rules only apply to packets forwarded by the unit. The traffic is analyzed and
filtered by all the rules configured.
Note: The Mediatrix units network firewall settings do not support IPv6. See IPv4 vs. IPv6 Availability on
If no rule matches the incoming packet, the default policy is applied. A rule's priority is determined by its index
in the table.
Rules using Network Names are automatically updated as the associated IP addresses and network mask are
modified.
If the Network Firewall service is stopped, all forwarded traffic is accepted, this tab is greyed out and the
parameters are not displayed. See Chapter 4 - Services on page 55 on information on how to start the
service.
The maximum number of rules allowed in the configuration is 20.
Caution: Enabling the network firewall and adding rules has an impact on the Mediatrix units overall
performance as the firewall requires additional processing. The more rules are enabled, the more overall
performance is affected. Furthermore, Media5 recommends to use a 30 ms packetization time when the
firewall is enabled (instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels
available on the unit, especially with the Mediatrix 3632 / 4404 / 4104 models.
Setting the Default Policy

The default policy defines the action the Mediatrix unit must take when a forwarded packet does not match
any rules.
137
Chapter 17 - Network Firewall Configuration
To set the default policy:

1.
In the web interface, click the Network link, then the Network Firewall sub-link.
Figure 51: Network Network Firewall Web Page
2.
In the Network Firewall Configuration section, define the default policy in the Default Policy dropdown menu.
Table 105: Default Policy Parameters
Parameter
Description
Accept
Drop
Setting the default policy to Drop or adding a rule automatically enables the network firewall.
Enabling the network firewall may have a negative impact on performance.
Creating/Editing a Network Firewall Rule

The web interface allows you to create a network firewall rule or modify the parameters of an existing one.
To create or edit a network firewall rule:

1.
In the Network Firewall Rules section of the Network Firewall page, do one of the following:
click the
button of this row.
button at the
Figure 52: Network Firewall Rules Section

2
applied (i.e., the Firewall section of the Status page differs from the Network Firewall page). The Network
Firewall page is a working area where you build up a network firewall configuration. While you work in this
area, the configured parameters are saved but not applied (i.e., they are not used to filter packets). The
yellow Yes flag warns you that the configuration has been modified but is not applied.
138
2.
Table 106: Firewall Rule Activation Parameters
Parameter
Description
Enable
This rule is active in the firewall.
Disable
This rule is not in the firewall.
Only enabled rules may be applied to the firewall.

3.
Enter the source address of the incoming packet in the corresponding Source Address or Interface
field.
Syntax
address[/mask]
networkInterfaceName/
Description
Network IP address (using /mask). The mask must be a plain
number specifying the number of binary 1s at the left side of the
network mask (a mask of 24 specifies a network mask of
255.255.255.0). For instance:
192.168.0.11
192.168.1.0/24

interface name is case sensitive, hence it must be entered
properly.
is automatically disabled thus removed from the firewall. When
the network interface is enabled or added back, the rule is
automatically enabled and applied in the firewall. For instance:

address.
4.
You can enter a single port or a range of ports. This field supports the following syntax:
port[-port]
any port.
7).
5.
Enter the destination address of the incoming packet in the corresponding Destination Address or
Interface field.
139

Syntax
Description
Network IP address (using /mask). The mask must be a plain
number specifying the number of binary 1s at the left side of the
network mask (a mask of 24 specifies a network mask of
255.255.255.0). For instance:
address[/mask]
192.168.0.11
192.168.1.0/24

interface name is case sensitive, hence it must be entered
properly.

is automatically disabled thus removed from the firewall. When
the network interface is enabled or added back, the rule is
automatically enabled and applied in the firewall. For instance:

address.
6.
port[-port]
matching any port.
7).
7.
Select the protocol of the incoming packet to filter in the corresponding Protocol drop-down menu.
Table 109: Firewall Rule Protocol Parameters
Parameter
8.
Description
All
TCP
UDP
ICMP
Set the corresponding Connection State drop-down menu with the connection state associated with
the incoming packet.
The connection state can be one of the following:
Table 110: Connection State Parameters
State
140
Description
All
Match packets in any state.
New
Match packets that are not part of an existing connection.
Disabling the Network Firewall
Table 110: Connection State Parameters (Continued)

State
Description
Established Or Related Match packets that are part of an existing connection.

9.
Select the action to take in the corresponding Action field.

Table 111: Network Firewall Rule Action Parameters
Parameter
Description
Accept
Reject
Sends back an ICMP port unreachable in response to the matched packet. The
packet is then dropped.
Drop
Note that if a connection is already established before creating a rule that rejects it, this connection
stays active despite the rule applied.
10.

The current enabled rules applied are displayed in the Network > Status web page, which contains
the active configuration in the network firewall. You can also see that the yellow Config Modified
Yes flag is cleared.
Note: You can revert back to the configuration displayed in the Network > Status web page at any time
(including the disabled rules) by clicking the Rollback button at the bottom of the page. All modified settings
in the Network > Network Firewall page will be lost.
Moving a Network Firewall Rule

The firewall rules sequence is very important because only one network firewall rule is applied on a packet.
Rules priority is determined by their position in the table. If you want the unit to try to match one rule before
another one, you must put that rule first.

1.
Either click the
or
2.
Deleting a Network Firewall Rule


1.
Click the
2.

When the network firewall is enabled, it has an impact on the Mediatrix units overall performance as the
firewall requires additional processing. You can disable the firewall if you do not need it, thus not impacting
performance. To disable the network firewall, you must stop the NFW service in the System > Services page.
See Chapter 4 - Services on page 55 for more details on how to stop a service. All forwarded traffic is allowed
when the network firewall service is stopped.
141
142
H A P T E R
18
NAT Configuration
This chapter describes how to configure the NAT parameters of the Mediatrix unit.
Creating/editing a Source NAT

Creating/editing a Destination NAT
Moving a NAT rule
Deleting a NAT rule
Introduction
Network Address Translation (NAT, also known as network masquerading or IP masquerading) rewrites the
source and/or destination addresses/ports of IP packets as they pass through a router or firewall. It is most
commonly used to connect multiple computers to the Internet (or any other IP network) by using one IP
address. This allows home users and small businesses to cheaply and efficiently connect their network to the
Internet. The basic purpose of NAT is to multiplex traffic from the internal network and present it to the Internet
as if it was coming from a single computer having only one IP address.
The Mediatrix units NAT service allows the dynamic creation and configuration of network address translation
rules. Depending on some criteria, the packet matching the rule may see its source or destination address
modified.
There are two types of NAT rules:
Source rules: They are applied on the source address of outgoing packets.
Destination rules: They are applied on the destination address of incoming packets.
A rule's priority is determined by its index in the Source NAT or Destination NAT tables.
If the NAT service is stopped, this tab is greyed out and the parameters are not displayed. See Chapter 4 Services on page 55 on information on how to start the service.
The maximum number of rules allowed in the configuration is 10 of each Source NAT and Destination NAT.
Caution: Adding source or destination NAT rules has an impact on the Mediatrix units overall performance
as the NAT requires additional processing. The more rules are enabled, the more overall performance is
affected. Furthermore, Media5 recommends to use a 30 ms packetization time when the NAT is enabled
(instead of a 20 ms ptime, for instance) in order to simultaneously use all the channels available on the unit,
especially with the Mediatrix 3632 / 4404 /4104 models.
Partial Reset
When a partial reset is triggered, the configuration is rolled back if it was being modified.
A new rule is then automatically applied in the source and in the destination NAT tables to prevent incorrect
rules from blocking access to the unit. If those rules are not the first priority, they are raised. If there are no
rules in the tables, the new rules are not added since there are no rules to override.
Creating/Editing a Source NAT Rule

SNAT rules are executed after the routing decision, before the packet leaves the unit.
The web interface allows you to create a source NAT rule or modify the parameters of an existing one. The
following parameters must all match to apply a SNAT rule to a packet:
Source Address
143
Chapter 18 - NAT Configuration
Source Port
Destination Address
Destination Port
Protocol
When the above parameters all match, then a new source IP address/port is applied to the packet.
To create or edit a source NAT rule:

1.
In the web interface, click the Network link, then the NAT sub-link.
Figure 53: Source Network Address Translation Rules Section
2.
In the Source Network Address Translation Rules section of the NAT page, do one of the following:
click the
button of this row.
button at the
applied (i.e., the Network Address Translation section of the Status page differs from the NAT page). The
NAT page is a working area where you build up a NAT configuration. While you work in this area, the
configured parameters are saved but not applied (i.e., they are not used in the NAT). The yellow Yes flag
warns you that the configuration has been modified but is not applied.
3.
Table 112: Source NAT Rule Activation Parameters
Parameter
Description
Enable
This SNAT rule is enabled.
Disable
This SNAT rule is disabled.
Only enabled rules may be applied to the Source NAT.

4.
144

Syntax
Description
the number of binary 1s at the left side of the network mask (a
instance:
address[/mask]
192.168.0.11
192.168.1.0/24

(see Interfaces Configuration on page 102 for more details).
The interface name is case sensitive, hence it must be entered
properly.
automatically enabled and applied in the NAT. For instance:
Lan1 (Lan1 IP address)

5.
port[-port]
any port.
7).
6.
field.
Table 114: Destination Address Syntax
Syntax
address[/mask]
Description
the number of binary 1's at the left side of the network mask (a
instance:
192.168.0.11
192.168.1.0/24
145
Table 114: Destination Address Syntax (Continued)

Syntax
Description
Configuration on page 102 for more details). The interface name
is case sensitive, hence it must be entered properly.

automatically enabled and applied in the Source NAT. For
instance:

address.
7.
You can enter a single port or a range of ports. This field supports the following format:
port[-port]
matching any port.
7).
8.
Select the protocol of the incoming packet to NAT in the corresponding Protocol drop-down menu.
Table 115: Source NAT Rule Protocol Parameters
Parameter
9.
Description
All
TCP
Matches only TCP packet.
UDP
ICMP
Enter the new address applied to the source of the packet in the New Address field.
Table 116: New Address Syntax
Syntax
address[:port]
10.
Description
Any IP address. When specifying a port number, it is mandatory to
have the protocol set to TCP or UDP.

The current enabled rules applied are displayed in the Network > Status web page, Network
Address Translation section, which contains the active configuration in the NAT. You can also see
that the yellow Config Modified Yes flag is cleared.
Note: You can revert back to the configuration displayed in the Status web page at any time (including the
disabled rules) by clicking the Rollback button at the bottom of the page. All modified settings in the NAT
page will be lost.
146
Creating/Editing a Destination NAT Rule

The web interface allows you to create a Destination NAT rule or modify the parameters of an existing one.
This creates a rule that allows remote computers (e.g., public machines on the Internet) to connect to a specific
computer within the private LAN, depending on the port used to connect. A destination NAT is also known as
port forwarding or virtual server.
DNAT rules are executed before the routing decision, as the packet enters the unit. Therefore it is important
to configure the Network Firewall (Chapter 17 - Network Firewall Configuration on page 137) with respect to
the DNAT rules. An example of this would be port forwarding where the DNAT changes the routed address of
a packet to a new IP address/port. The Network Firewall must also accept connection to this IP/port in order
for the port forwarding to work.
The following parameters must all match to apply a DNAT rule to a packet:
Source Address
Source Port
Destination Address
Destination Port
Protocol
When the above parameters all match, then a new destination IP address/port is applied to the packet.
To create or edit a Destination NAT rule:

1.
In the Destination Network Address Translation Rules section of the NAT page, do one of the
following:
click the
button of this row.
button at the
Figure 54: Destination Network Address Translation Rules Section

2
Note: When you add a new rule, edit an existing rule, or delete a rule, you can see a yellow Yes in the
Config Modified section at the top of the window. It warns you that the configuration has been modified but
not applied (i.e., the Network Address Translation section of the Status page differs from the NAT page).
The NAT page is a working area where you build up a NAT configuration. While you work in this area, the
configured parameters are saved but not applied (i.e., they are not used in the NAT). The yellow Yes flag
warns you that the configuration has been modified but is not applied.
2.
Table 117: Destination NAT Rule Activation Parameters
Parameter
Description
Enable
This DNAT rule is enabled.
Disable
This DNAT rule is disabled.
Only enabled rules may be applied to the Destination NAT.

3.
147

Syntax
address[/mask]
Description
instance:
192.168.0.11
192.168.1.0/24
automatically enabled and applied in the Destination NAT. For
instance:

address.
4.
port[-port]
any port.
7).
5.
field.
Table 119: Destination Address Syntax
Syntax
address[/mask]
148
Description
instance:
192.168.0.11
192.168.1.0/24
Table 119: Destination Address Syntax (Continued)

Syntax
Description
automatically enabled and applied in the Destination NAT. For
instance:
Lan1 (Lan1 IP address)

6.
port[-port]
matching any port.
7).
7.
Select the protocol of the incoming packet to NAT in the corresponding Protocol drop-down menu.
Table 120: Destination NAT Rule Protocol Parameters
Parameter
8.
Description
All
TCP
UDP
ICMP
Enter the new address of the packet in the New Address field.
Table 121: New Address Syntax
Syntax
address[:port]
9.
Description
Any IP address. When specifying a port number, it is mandatory to
have the protocol set to TCP or UDP.

The current enabled rules applied are displayed in the Network > Status web page, Network
Address Translation section, which contains the active configuration in the NAT. You can also see
Note: You can revert back to the configuration displayed in the Network > Status web page at any time
(including the disabled rules) by clicking the Rollback button at the bottom of the page. All modified settings
in the Network > NAT page will be lost.
149
Moving a NAT Rule
Moving a NAT Rule

The NAT rules sequence is very important because only one SNAT rule or one DNAT rule is applied on a
packet. Rules priority is determined by their position in the table. If you want the unit to try to match one rule
before another one, you must put that rule first.

1.
Either click the
or
2.
Deleting a NAT Rule


1.
Click the
2.
Disabling the NAT

When the NAT is enabled, it has an impact on the Mediatrix units overall performance as the NAT requires
additional processing. You can disable the NAT if you do not need it, thus not impacting performance. To
disable the NAT, you must stop the NAT service in the System > Services page. See Chapter 4 - Services
on page 55 for more details on how to stop a service.
150
H A P T E R
19
DHCP Server Settings

This chapter describes how to configure the embedded DHCP server of the Mediatrix unit.
Standards Supported
RFC 2131: Dynamic Host Configuration Protocol, section 2

(server side)
RFC 2132: DHCP Options and BOOTP Vendor Extensions

(sections 3.3, 3.5, 3.8, 3.17, 8.3 and 8.5)
Introduction
The Mediatrix unit contains an embedded DHCP server that allocates IP addresses and provides leases to
the various subnets that are configured. These subnets could have PCs or other IP devices connected to the
units LAN Ethernet connectors. These devices could be any combination of switches, PCs, IP phones, etc.
If the DHCP service is stopped, this tab is greyed out and the parameters are not displayed. See Chapter 4 Services on page 55 on information on how to start the service.
Note: The Mediatrix units DHCP server settings do not support IPv6. See IPv4 vs. IPv6 on page 87 for
more details.
Subnet Server
The DHCP server manages the hosts network configuration on a given subnet. Each subnet can be seen as
having a distinct DHCP server managing it, which is called a subnet server. To activate a subnet server for a
given network interface, the name of that network interface and the name of the subnet configuration must
match (the names are case sensitive). Only one subnet can be defined per network interface. The network
interface can be a physical interface or a logical interface (e.g., sub-interface using VLAN).
Leases
In order to assign leases, the subnet server draws from an IP address pool (or subnet scope) defined by a
start address and an end address. The subnet mask assigned to hosts is taken directly from the network
interface. All hosts on the same subnet share the same configuration. The maximum number of hosts
supported on a subnet is 254.
You can reserve IP addresses for specific hosts that are designated by their MAC address. Those addresses
are then removed from the pool of IP addresses that can be leased. Once a lease is assigned, it is removed
from the pool of IP addresses that can be leased for as long as the host keeps it.
Configuration Parameters
When an address is leased to a host, several network configuration parameters are sent to that host at the
same time according to the options found in the DHCP request. You can modify the configuration source of a
parameter. The following are the possible configuration sources:
Table 122: Parameter Configuration Sources
Source
Description
Static
The parameter is defined as a static parameter locally.
Automatic
The parameter is obtained from the network configured in the Automatic Configuration
Interface drop-down menu of this subnet (DHCP Basic Configuration on page 153).
151
Chapter 19 - DHCP Server Settings
Introduction
Table 122: Parameter Configuration Sources (Continued)

Source
Description
Host Configuration The parameter is obtained from the host configuration.

Host Interface
The parameter is obtained from the network interface matching the subnet.
The following table lists the configuration parameters and their available configuration sources:
Table 123: Optional Parameter and Possible Configuration Sources
Configuration Sources
Parameter Name
Static
Automatic
Host Config
Host Interface
Domain Name
Lease time
Default gateway
List of DNS servers
List of NTP servers
List of NBNS servers
Default vs. Specific Configurations

You can use two types of configuration:
Default configurations that apply to all the subnets of the Mediatrix unit.
Specific configurations that override the default configurations.
You can define specific configurations for each subnet in your Mediatrix unit. For instance, you
could define a lease time for all the subnets of the Mediatrix unit and use the specific configuration
parameters to set a different value for one specific subnet.
The parameters available differ according to the subnet you have selected. The Default subnet has less
parameters than the specific subnets available on the Mediatrix unit.
152
DHCP Basic Configuration
DHCP Basic Configuration

The basic configuration parameters are available only on the specific subnets configuration.
To set the DHCP server basic parameters:

1.
In the web interface, click the Network link, then the DHCP Server sub-link.
2.
Select a specific subnet in the Select Subnet drop-down menu at the top of the window.
You have the choice between Default (applies to all subnets) and specific subnets.
3.
In the DHCP Server Configuration section of the DHCP Server page, enable the DHCP server by
selecting Enable in the DHCP Server Enable drop-down menu.
Figure 55: DHCP Server Configuration General Parameters
3
4
5
4.
Set the start and end IP addresses of the subnet range in the Start IP Address and End IP Address
fields.
These are the addresses that the DHCP server offers to the subnets of the Mediatrix unit. The
Mediatrix unit can offer up to 254 addresses. These addresses must be within the network
interfaces subnet or the subnet server will have an invalid configuration status.
5.
Set the Automatic Configuration Interface drop-down menu with the network interface that provides
the automatic configuration (e.g.: DNS servers, NTP server, etc.) to all parameters of this subnet
that use the "Automatic" configuration source.
6.
Lease Time (Option 51)

The Mediatrix unit DHCP server offers a lease time to its subnets. You can use a default lease time for all
subnets or define a lease time specific to one or more subnets.
To set the DHCP server lease time parameters:

1.
In the Lease Time (Option 51) sub-section of the DHCP Server Configuration section, define
whether or not you want to override the lease time set in the Default configuration in the Subnet
Specific drop-down menu.
This menu is available only in the specific subnets configuration.
Figure 56: DHCP Server Configuration Lease Time Option
1
2
2.
Define the lease time (in seconds) given by the Mediatrix unit DHCP server in the Lease Time field.
3.
153
Domain Name (Option 15)

The Mediatrix unit DHCP server offers a domain name to its subnets. You can use a default domain name for
all subnets or define a domain name specific to one or more subnets.
To set the DHCP server domain name parameters:

1.
In the Domain Name (Option 15) sub-section of the DHCP Server Configuration section, enable the
domain name (option 15) by selecting Enable in the Enable Option drop-down menu.
Figure 57: DHCP Server Configuration Domain Name Option
1
3
4
2.
Define whether or not you want to override the domain name parameters set in the Default
configuration in the Subnet Specific Value drop-down menu.
3.
If the domain name option is enabled, select the configuration source of the domain name
Table 124: Domain Name Configuration Sources
Source
Description
Host
Configuration
The domain name is the one used by the unit.
Static
You manually enter a domain name.

4.
If the configuration source is Static, enter the static default domain name for all subnets in the
Domain Name field.
5.
Default Gateway (Option 3)

The Mediatrix unit DHCP server offers a default gateway (also called default router) to its subnets.
Note: The default gateway parameters are not available in the Default interface. You must access the
specific subnets configuration to set its parameters.
To set the DHCP server default gateway parameters:

1.
In the Default Gateway (Option 3) sub-section of the DHCP Server Configuration section, enable
the default gateway (option 3) by selecting Enable in the Enable Option drop-down menu
Figure 58: DHCP Server Configuration Default Gateway Option
1
2
3
154
DNS (Option 6)
2.
Select the configuration source of the default gateway information in the Configuration Source dropdown menu.
Table 125: Default Gateway Configuration Sources
Source
Description
Host Interface
The default gateway is the host address within the client's subnet.
Static
You manually enter the value.

3.
If the configuration source is Static, enter the default gateway host name or IP address of the
subnet in the Default Gateway field.
4.
DNS (Option 6)
The Mediatrix unit DHCP server offers up to four DNS addresses to its subnets. You can use the default DNS
addresses for all subnets or define static DNS addresses specific to one or more subnets.
To set the DHCP server DNS parameters:

1.
In the DNS (Option 6) sub-section of the DHCP Server Configuration section, enable the DNS
servers (option 6) by selecting Enable in the Enable Option drop-down menu
Figure 59: DHCP Server Configuration DNS Option
1
3
4
2.
Define whether or not you want to override the default values in the Subnet Specific drop-down
menu.
3.
Select the configuration source of the DNS information in the Configuration Source drop-down
menu.
Table 126: DNS Configuration Sources
Source
Description
Host
Configuration
The DNS servers are obtained from the host configuration.
Automatic
The DNS servers are automatically obtained from the network configured in the
Automatic Configuration Interface drop-down menu of this subnet (DHCP
Basic Configuration on page 153).
Static

4.
If the configuration source is Static, enter the static addresses of up to four DNS servers in the
following fields:
155
5.
NTP (Option 42)
Primary DNS
Secondary DNS
Third DNS
Fourth DNS
NTP (Option 42)

The Mediatrix unit DHCP server offers the addresses of up to four NTP (Network Time Protocol) servers to its
subnets. You can use the default NTP addresses for all subnets or define static DNS addresses specific to
one or more subnets.
To set the DHCP server NTP parameters:

1.
In the NTP (Option 42) sub-section of the DHCP Server Configuration section, enable the NTP
Figure 60: DHCP Server Configuration NTP Option
1
2
3
4
2.
menu.
3.
Select the configuration source of the NTP information in the Configuration Source drop-down
menu.
Table 127: NTP Configuration Sources
Source
Description
Host
Configuration
The NTP servers are obtained from the host configuration.
Automatic
The NTP servers are automatically obtained from the network configured in the
Automatic Configuration Interface drop-down menu of this subnet (DHCP
Basic Configuration on page 153).
Static

4.
5.
156
If the configuration source is Static, enter the static addresses of up to four NTP servers in the
following fields:
Primary NTP
Secondary NTP
Third NTP
Fourth NTP
NBNS (Option 44)
NBNS (Option 44)

The NetBIOS Name Server (NBNS) protocol, part of the NetBIOS over TCP/IP (NBT) family of protocols, is
implemented in Windows systems as the Windows Internet Name Service (WINS). By design, NBNS allows
network peers to assist in managing name conflicts.
The Mediatrix unit DHCP server offers up to four NBNS addresses to its subnets. You can use the default
NBNS addresses for all subnets or define static NBNS addresses specific to one or more subnets.
To set the DHCP server NBNS parameters:

1.
In the NBNS (Option 44) sub-section of the DHCP Server Configuration section, enable the NBNS
Figure 61: DHCP Server NBNS Option
1
2
3
2.
menu.
3.
4.
Enter the static addresses of up to four NBNS servers in the following fields:
Primary NBNS
Secondary NBNS
Third NBNS
Fourth NBNS
DHCP Static Leases Configuration


by using the CLI
The embedded DHCP server leases addresses to the hosts that request it. The address is assigned to a host
for a configurable amount of time (as defined in Lease Time (Option 51) on page 153). The DHCP server
can service all subnets on which it is enabled.
To define DHCP leases offered by the Mediatrix unit:
1.
In the web interface, click the System link, then the DHCP Leases sub-link.
2.
In the Static Leases Configuration section, if applicable, delete an existing reserved IP address by
selecting Delete in the Action drop-down next to an existing lease.
3.
If applicable, add a new lease by entering the MAC address of the device and the IP address you
want to reserve for it, then click Submit.
157
DHCP Static Leases Configuration
The static IP address is added to the Static Leases Configuration section, but not to the Current
Leases section.
4.
158
POTS Parameters
H A P T E R
20
POTS Configuration
This chapter describes how to configure the POTS (Plain Old Telephony System) line service, which allows
you to configure the analog specification of each line, as well as gateways-specific parameters.
Note: This web page is available only on the following models:

Mediatrix 3208 / 3216
Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
POTS Status
The POTS parameters are displayed in the POTS / Status page.
Line Status
The Line Status table lists the link state of the FXS lines.
Figure 62: POTS Status Web Page
The State column may have one of the following values:
Idle: The line is available

In Use: The line is currently used
Disabled: The line is disabled
Bypass: The line is on bypass
Down: The power of the line is down
161
Chapter 20 - POTS Configuration
General POTS Configuration
FXO Line Status

Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742Mediatrix C730 / C733 / C731
The FXO Line Status table lists the link state updated by the link state verification mechanism.
Figure 63: POTS FXO Line Status Table
You can enable the line state verification in FXO Line Verification on page 176.
The unit does not automatically detect when a previously connected port has changed to the disconnected
status. The unit only detects the change of status when it attempts to use the port or after a restart.
The unit automatically detects within seconds when a disconnected port becomes connected.
Unknown: The line fault detection is disabled.

Up: When last polled, the line was connected.
Down: When last polled, the line was disconnected.

The General Configuration section allows you to select the detection/generation method of caller ID.
To configure the general POTS parameters:

1.
In the web interface, click the POTS link, then the Config sub-link.
Figure 64: POTS Web Page
2
4
2.
Select the detection/generation method of caller ID in the Caller ID customization drop-down menu.
This allows selecting the detection/generation method of caller ID. See Caller ID Information on
Table 128: Caller ID Parameters
Parameter
Country
162
Description
Uses the default caller ID of the country defined in the Country section of the
Telephony > Misc page (Country Configuration on page 455).
Table 128: Caller ID Parameters (Continued)

Parameter
3.
Description
EtsiDtmf
ETSI 300 659-1 (DTMF string sent between the first and second ring).
EtsiFsk
ETSI 300 659-1 (FSK (V.23) sent between the first and second ring).
Select the caller ID transmission method in the Caller ID Transmission drop-down menu.
It allows selecting the transmission type of the caller ID.
Table 129: Caller ID Transmission Parameters
Parameter
Description
Country
Uses the default caller ID of the country defined in the Country section of the
Telephony > Misc page (Country Configuration on page 455).
First Ring
The caller ID is sent after the first ring.
Ring Pulse
The caller ID is sent between a brief ring pulse and the first ring.
Line Reversal
Ring Pulse
The caller ID is sent between a brief ring pulse and the first ring on an inverted
polarity line.
DT-AS
The caller ID is sent after the dual tone alerting state tone.
Line Reversal
DT-AS
The caller ID is sent after the dual tone alerting state tone on an inverted
polarity line.
No Ring Pulse The caller ID is sent before the first ring.

4.
Determine the type of vocal information that can be obtained by dialing a pre-defined digit map in
the Vocal Unit Information drop-down menu.
When entering special characters on your telephone pad, the Mediatrix unit talks back to you with
relevant information.
Table 130: Caller ID Parameters
Parameter
Description
None
The vocal information feature is disabled.
All
Enable all vocal information digit maps.
To access the vocal unit information:

a.
Take one of the telephones connected to the Mediatrix unit.
b.
Dial one of the digits sequence on the keypad.

Table 131: Vocal Unit Information
Digits to Dial
5.
Information Vocally Sent by the Mediatrix unit
*#*0
List of IP addresses of the Mediatrix unit (static or

DHCP).
*#*1
MAC address of the Mediatrix unit.
*#*8
Firmware version number of the Mediatrix unit.
163
Caller ID Information
The caller ID is a generic name for the service provided by telephone utilities that supply information such as
the telephone number or the name of the calling party to the called subscriber at the start of a call. In call
waiting, the caller ID service supplies information about a second incoming caller to a subscriber already busy
with a phone call. However, note that caller ID on call waiting is not supported by all caller ID-capable
telephone displays.
In typical caller ID systems, the coded calling number information is sent from the central exchange to the
called telephone. This information can be shown on a display of the subscriber telephone set. In this case, the
caller ID information is usually displayed before the subscriber decides to answer the incoming call. If the line
is connected to a computer, caller information can be used to search in databases and additional services can
be offered.
The following basic caller ID features are supported:
Date and Time

Calling Line Identity
Calling Party Name
Visual Indicator (MWI)
Caller ID Generation
There are two methods used for sending caller ID information depending on the application and countryspecific requirements:
caller ID generation using DTMF signalling

caller ID generation using Frequency Shift Keying (FSK)
Note: The Dgw v2.0 Application does not support ASCII special characters higher than 127.
The displayed caller ID for all countries may be up to 20 digits for numbers and 50 digits for names.
DTMF Signalling
The data transmission using DTMF signalling is performed during or before ringing depending on the country
settings or endpoint configuration. The Mediatrix unit provides the calling line identity according to the following
standards:
Europe: ETSI 300 659-1 January 2001 (Annex B): Access and Terminals (AT); Analogue
access to the Public Switched Telephone Network (PSTN); Subscriber line protocol over the
local loop for display (and related) services; Part 1: On-hook data transmission.
FSK Generation
Different countries use different standards to send caller ID information. The Mediatrix unit is compatible with
the following widely used standards:
ETSI 300 659-1
Note: The compatibility of the Mediatrix unit is not limited to the above caller ID standards.
Continuous phase binary FSK modulation is used for coding that is compatible with:
164
BELL 202
ITU-T V.23
FXS Configuration
FXS Configuration
The FXS Configuration section allows you to define how a FXS endpoint behaves in certain conditions.
To configure the FXS parameters:

1.
In the web interface, click the POTS link, then the FXS Config sub-link.
Figure 65: FXS Config Web Page
2
4
6
3
5
7
2.
In the FXS Configuration section, set the Line Supervision Mode drop-down menu with the power
drop and line polarity used to signal the state of a line.
Power drop and polarity reversal are also called battery drop and battery reversal.
Table 132: Line Supervision Mode Parameters
Parameter
Description
None
Power drop or polarity reversal is not used to signal the state of the
line.
DropOnDisconnect
Activates the Power Drop on Disconnect feature. A short power drop

is made at the end of a call when the call is disconnected by the
remote party.
The drop duration can be configured in the FXS Power Drop on
Disconnect Duration field (Step 5).
ReversalOnIdle
Activates the Polarity Reversal on Idle feature. The polarity of the line
is initially in reversed state. The polarity of the line returns to the
positive state when the user seizes the line or when the line rings for
an incoming call. The polarity of the line is reversed again when the
call is disconnected.
ReversalOnEstablished Activates the Polarity Reversal on Established option. The polarity of

the line is initially in the positive state. The polarity of the line is
reversed when the call is established and returns to the positive state
when the call is disconnected.
3.
Set the Disconnect Delay field with the value used to determine whether or not call clearing occurs
as soon as the called user is the first to hang up a received call.
This parameter has no effect when you are acting as the calling party.
If you set the value to 0, the call is disconnected as soon as the called user hangs up the call.
If the value is greater than 0, that value is the amount of time, in seconds, the unit waits after the
called user hangs up before signalling the end of the call.
4.
Set the Auto Cancel Timeout field with the time, in seconds, the endpoint rings before the call is
automatically cancelled.
165
FXS Configuration
Setting this variable to 0 disables the timeout. Calls will not be automatically cancelled and will ring
until the party answers.
5.
Set the Inband Ringback drop-down menu to define whether or not the FXS endpoint needs to
generate a ringback for incoming ringing call.
Table 133: Inband Ringback Parameters
Parameter
6.
Description
Disable
The FXS endpoint does not play local ringback to the remote party.
Enable
The FXS endpoint plays local ringback to the remote party via the negotiated
media stream. The local ringback is generated only when the telephone is onhook. The FXS ports never play the local ringback for the call waiting.
Set the Shutdown Behavior drop-down menu with the FXS endpoint behavior when it becomes shut
down.
Table 134: FXS Shutdown Behavior Parameters
Parameter
Description
Disabled
Tone
A disabled tone is played when the user picks up the telephone and the FXS
endpoint is shut down.
Power
Drop
The loop current is interrupted when the FXS endpoint is shut down and no tone is
played when the user picks up the telephone.
A FXS endpoint becomes shut down when the operational state of the endpoint becomes Disabled
and the Shutdown Endpoint When Operational State is 'Disable' And Its Usage State Is 'idleunusable' parameter of the SIP > Endpoints page is set to Enable. See Administration on page 70
for more details.
This parameter is not used by FXS endpoints used for bypass when the Activation column of the
FXS Bypass section is set to Endpoint Disabled. See FXS Bypass on page 169 for more details.
7.
Set the Power Drop on Disconnect Duration field with the power drop duration, in milliseconds, that
is made at the end of a call when the call is disconnected by the remote party.
This value only has an effect when the Line Supervision Mode drop-down menu is set to
DropOnDisconnect.
8.
Set the Service Activation drop-down menu with the method used by the user to activate
supplementary services such as call hold, second call, call waiting, call transfer and conference call.
Table 135: Service Activation Parameters
Parameter
Flash Hook
166
Description
Service activation is performed by flash hook or hanging up.
FXS Country Customization
Table 135: Service Activation Parameters (Continued)

Parameter
Description
Flash Hook And

Digit
Service activation is performed by flash hook, flash hook followed by a digit

or hanging up.
The digit dialed has a different behaviour depending on the current call
context:
One call active with one waiting call or one call on hold:
Flash hook then dial the digit 1 or hang up: Terminate the active call
and switch to the call on hold/waiting.
Flash hook then dial the digit 2: Hold the active call and switch to the
call on hold/waiting.
Flash hook then dial the digit 3: Enter in three-party conference mode.
Flash hook then dial the digit 4: Transfer the call on hold to the active
call (not available on call waiting).
When hanging up in this context, the telephone rings to notify the user
there is still a call on hold.
In three-party conference mode:

Flash hook then dial the digit 1: Exit from three-party conference
mode. The third party remains active and the second party call is
terminated.
Flash hook then dial the digit 2: Exit from three-party conference
mode. The second party remains active and the third party call is
placed on hold.
When hanging up in this context, all calls are finished.
9.

The FXS Country Customization section allows you to override the current default country parameters of
certain features. Refer to Appendix A - Country-Specific Parameters on page 621 for the pre-defined values
for a specific country.
To define the FXS country customization parameters:

1.
In the FXS Country Configuration section, select whether or not you want to override the current
country parameters in the Override Country Customization drop down menu.
This allows overriding FXS related default country settings for the loop current and flash hook
detection features.
Figure 66: FXS Country Customization Section
1
3
Table 136: Line Supervision Mode Parameters

Parameter
Disable
Description
The line uses the default country FXS settings.
167
Table 136: Line Supervision Mode Parameters (Continued)

Parameter
Description
Enable
2.
The line uses the FXS country configuration set in the following
steps.
Set the Country Override Loop Current field with the loop current generated by the FXS port in ma.
When a remote end-user goes on-hook, the Mediatrix unit signals the far end disconnect by
performing a current loop drop (< 1 mA) on the analog line. This current loop drop, also referred to
as Power Denial mode, is typically used for disconnect supervision on analog lines. The Mediatrix
unit maintains a current drop for one second (this value cannot be configured), then a busy tone is
generated to indicate the user to hang up. See the description for the FXS Line Supervision Mode
drop-down menu in FXS Configuration on page 165 for more details.
When one of its analog lines goes off-hook, the Mediatrix unit controls the endpoint in a fixed loop
current mode. When selecting a country (see Country Configuration on page 455 for more details),
each country has a default loop current value. However, you can override this value and define your
own loop current.
Note that the actual measured current may be different than the value you set, because it varies
depending on the DC impedance. Mediatrix LP16/LP24 models: The values available for these
models are between 20 mA and 25 mA. The default value is 22 mA. If you set a value higher than
25 mA, the unit will limit the current to 25 mA.
3.
Set the Country Override Flash Hook Detection Range field.

This is the range in which the hook switch must remain pressed to perform a flash hook.
When selecting a country (see Country Configuration on page 455 for more details), each country
has a default minimum and maximum time value. However, you can override these values and
define your own minimum and maximum time within which pressing and releasing the plunger is
actually considered a flash hook.
The range consists of the minimal delay and maximal delay, in ms, separated by a -. The minimal
value allowed is 10 ms and the maximum value allowed is 1200 ms. The space character is not
allowed.
Flash hook can be described as quickly depressing and releasing the plunger in or the actual
handset-cradle to create a signal indicating a change in the current telephone session. Services
such as picking up a call waiting, second call, call on hold, and conference are triggered by the use
of the flash hook.
A flash hook is detected when the hook switch is pressed for a shorter time than would be required
to be interpreted as a hang-up.
Using the flash button that is present on many standard telephone handsets can also trigger a
flash hook.
4.
Calling Party Name of the Caller ID

Standards Supported
ETSI EN 300659-3a
a. CLIR section
1.
In the potsMIB, specify the Calling Party Name of the caller ID (CLIP) when the calling party is
tagged as private in the FxsCallerIdPrivateCallingPartyName variable.
pots.FxsCallerIdPrivateCallingPartyName="Value"
Value may be any string of characters up to 50 characters.
168
When empty, no Calling Party Name parameter is sent.
FXS Bypass
When set to 'P', no Calling Party Name parameter is sent but a Reason for Absence or
Caller Party Name parameter is sent with the value 0x50 (Private).
FXS Bypass
Mediatrix 3716
Mediatrix 3731
Mediatrix 3741
Mediatrix 4100 Series (except Mediatrix 4102S)
The FXS Bypass section allows you to define whether or not the bypass feature is enabled. This feature allows
its users to maintain telephone services in the event of a power outage or network failure.
During normal operation, the SCN line connected to the Bypass connector is switched out of the circuit through
commuting relays. The Bypass connector can be activated by three different conditions:
When power is removed from the Mediatrix unit.

When the FXS endpoints operational state is disabled and the endpoint is not in use.
When the user signals the configured DTMF map.
If one of these conditions is met, a phone/fax used on a FXS connector that supports the bypass feature is
directly connected to the SCN Bypass line. The FXS connector stays in Bypass connection until:
The error conditions have been cleared.

The device connected to it is on-hook and a delay has elapsed.
Note: For the Mediatrix 3308, Mediatrix 3316, Mediatrix 3731 and Mediatrix 3741 models: If an event that
activates the FXS bypass occurs and the FXO port is in use, the bypass activation waits until the FXO port
is no longer in use. See FXO Configuration on page 173 for more details.
For more information on your models bypass feature and which FXS connectors are available for bypass,
please refer to your models Hardware Configuration Guide.
To define the FXS bypass parameters:

1.
In the FXS Bypass section, specify when the bypass needs to be activated in the corresponding
Activation columns drop-down menu.
Figure 67: FXS Bypass Section
1
Table 137: Activation Parameters

Parameter
Power Off
Description
The bypass is activated only when the unit power is off or the
POTS service is not started.
169
FXS Emergency Call Override
Table 137: Activation Parameters (Continued)

Parameter
Endpoint Disabled
Description
The bypass is activated when:
the operational state of the endpoint is Disable, and
the Shutdown Endpoint When Operational State is

'Disable' And Its Usage State Is 'idle-unusable'
parameter of the SIP > Endpoints page is set to
Enable. See Administration on page 70 for more
details.
The bypass is also activated for the same conditions as the ones
defined in Power Off.
On Demand
The bypass is activated when the user enters the DTMF map
configured in the corresponding Activation DTMF Map field (Step
2).
The bypass is deactivated when the user on hooks the phone for
at least the number of time configured in the corresponding
Deactivation Timeout field (Step 3).
The bypass is also activated for the same conditions as the ones
defined in Power Off.
2.
Set the corresponding Activation DTMF Map field with the DTMFs to signal to enable the bypass.
This field is only used when the corresponding Activation drop-down menu is set to On Demand.
3.
Set the corresponding Deactivation Timeout field with the delay, in seconds, to wait before
deactivating the bypass after an on hook if the bypass is activated on demand.
The delay is restarted after each on hook. The bypass is not deactivated if the delay expires while
the FXS is off hook.
This field is only used when the corresponding Activation drop-down menu is set to On Demand.
Note: For the Mediatrix 3308, Mediatrix 3316, Mediatrix 3731 and Mediatrix 3741 models, the bypass is
automatically disabled after the timeout, whether or not the bypass is in use. Furthermore, if the conditions
that activate the bypass are no longer present, the bypass is disabled.
4.
FXS Emergency Call Override
Note: This feature is available for FXS units only.

This FXS Emergency Call Override feature allows you to override a set of services that are activated during
an emergency call.
Two variables are available:
FxsEmergencyCallOverride : to override or not the services.

FxsEmergencyRingTimeout: to set the period before the phone starts to ring in the event where
the originator of an emergency call hangs-up before the emergency call center disconnects the
call.
The configuration of the Emergency Call Override is only available in the MIB parameters of the Mediatrix unit.
170
FXS Distinctive Ring
You can configure the parameters as follows:

by using the CLI
To set the Emergency Call Override:

1.
In the potsMIB, set the FxsEmergencyCallOverride variable to the proper value, or
2.
In the CLI or a configuration script use:

Pots.FxsEmergencyCallOverride="Value"
Table 138: Fxs Emergency Call Override Parameters
Value
Parameter
Description
100
NoOverride
The set of services for emergency calls remains the same as

configured. This is the default value.
200
NoServices
Ignores any service requiring a flash-hook. Call waiting and all other
related services are deactivated.
300
NoDisconnect
Ignores any service requiring a flash-hook. Call waiting and all other
related services are deactivated AND automatically re-establishes a
call that was disconnected by the originator.
To set the Emergency Ring Timeout Override:

1.
Make sure the FxsEmergencyCallOverride variable is set to NoDisconnect.
2.
In the PotsMIB, set the FxsEmergencyRingTimeout variable to the proper value, or
3.

Pots.FxsEmergencyRingTimeout="Value"
where Value is in milliseconds. The default value is 2000 ms.

This FxsDistinctiveRingId parameter allows you to create a custom distinctive ring. Configuring the custom
distinctive ring allows the administrator to modify the ring pattern. When a pots.fxsDistinctiveRing.RindId is
defined, the corresponding ring pattern is used.
To use the distinctive ringing with the unit, the received SIP INVITE message must contain the Alert-Info
header field with the proper Call Property value.
Example
Alert-Info: <http://127.0.0.1/Bellcore-dr2>
Two variables are used to configure a distinctive ring:
RingId: Identifies the distinctive ring. When the incoming call property distinctive-ring
matches the defined RingId, the corresponding ring pattern is used. Otherwise, the
country ring pattern is used.
Pattern: Describes a tone pattern.
The format of the pattern is as follows:

ring-pattern = [ states-section ]
states-section = on-state-description "," off-state-description [ "," on-state-description
171
"," off-state-description [ "," on-state-description "," off-state-description ] ]

on-state-description = time
off-state-description = time
time = 2*5DIGIT
Table 139: Tag description

Tag
Description
ring-pattern
String describing the pattern to use for the ring. An empty

string means no ring.
states-section
Description of the state of the ring. Up to 3 pairs of states

can be defined. They must be at least one state described if
the ring-pattern is not empty.
on-state-description
Description of a state playing a ring.
off-state-description
Description of a state not playing a ring.
time
The number of time in ms to perform the action of the state.

Range is from 0 to 32767 ms.
Examples:
No ring:""
Bellcore-dr2: "800,400,800,4000"
Bellcore-dr4: "300,200,1000,200,300,4000"
Table 140: Default mapping between call property and ring cadence
Call Property Value
Ring cadence in milliseconds (bold are on,not bold are

off.
//127.0.0.1/Bellcore-dr2
800, 400, 800, 400
400, 200, 400, 200, 800, 4000
300, 200, 1000, 200, 300, 4000
All other value or call properties not present
Countrys normal ring.
The parameters can be set:

by using the CLI
creating a configuration script containing the configuration variables.
To customise a distinctive ring:

1.
In the potsMIB set:
Pots.FxsDistinctiveRingId variable in the FxsDistinctiveRing table
Pots.FxsDistinctivePattern variable in the FxsDistinctiveRing table.
or
2.
Use the CLI or a configuration script:
Pots.FxsDistinctivering[index=value].Pattern="Value"
Index value can vary from 1 to 4.
172
Pots.FxsDistinctivering[index=value].RingId="Value"
FXO Configuration
FXO Configuration
The FXO Config page allows you to configure gateways-specific parameters.

Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix C730 / C733 / C731
The FXO ports use depends on the FXS bypass state (as defined in the FXS Bypass section of the FXS
Configuration page FXS Bypass on page 169):
The FXO port becomes disabled as IP-FXO gateway when the FXS bypass is active. In this
case:
calls from the IP side to the FXO port are discarded
calls coming from the SCN side on the FXO port are routed to FXS port #1.
The FXO port becomes enabled and available for calls when the FXS bypass is disabled.
If an event that activates the FXS bypass occurs and the FXO port is in use, the bypass activation waits until
the FXO port is no longer in use.
FXO Dialing Configuration

The FXO Dialing Configuration section allows you to set dialing parameters.
To set FXO dialing parameters:

1.
In the web interface, click the POTS link, then the FXO Config sub-link.
Figure 68: POTS FXO Config Web Page
2
3
4
2.
Set the Pre Dial Delay field with the delay, in milliseconds, between the time the line is successfully
seized, or dial tone detected, and the moment the destination phone number is dialed.
Some country specifications include a mandatory pre-dial delay. In that case, the highest value
between that and the value set in this field is used.
3.
Set the Dial Tone Detection Mode drop-down menu with the proper behaviour.
173
FXO Configuration
When dial tone detection is enabled, the unit waits for a dial tone on the FXO line before initiating
the dialling sequence. If no dial tone is detected, the line is considered as busy with an incoming
call. This mechanism helps avoid collisions between incoming and outgoing calls.
Table 141: Dial Tone Detection Modes
Parameter
Disable
Description
Dial tone detection is disabled.
Country Tone The unit tries to detect the tone specified for this purpose in the current country's
tone specification. Some country specifications omit this information. In that case,
the unit behaves as if the parameter is set to Disable.
The following table lists the default dial tone detection frequency and cadence for each supported
country.
Table 142: Default Dial Tone Detection
Cadence
Country
Frequency
ON (s)
OFF (s)
Austria
450 Hz
0.5
0.0
Czech Republic
425 Hz
0.3
0.0
France
440 Hz
0.5
0.0
Germany1/2
425 Hz
0.15
0.0
Italy
425 Hz
0.15
0.0
NorthAmerica1
440 Hz
0.5
0.0
Spain
425 Hz
0.15
0.0
Switzerland
425 Hz
0.5
0.0
4.
Set the Dial Tone Detection Timeout field with the value, in milliseconds, indicating how long the
unit waits for a dial tone before considering the line is busy with an incoming FXO call.
5.
FXO Answering Configuration

The FXO Answering Configuration section allows you to define how the FXO line must behave when
answering calls.
To set FXO answering parameters:

1.
In the FXO Answering Configuration section, set the corresponding Wait Before Answering Delay
columns FXO interface field with the waiting period, in milliseconds, before answering an incoming
FXO call.
If this delay expires before the caller ID signal is decoded, the call proceeds without caller ID
information.
If a minimal waiting period is required for the selected country, the highest of both values is used.
Figure 69: FXO Answering Configuration Section
1
2.
174
Set the corresponding Answering On Caller ID Detection columns drop-down menu with the proper
behaviour.
FXO Configuration
This parameter enables answering upon caller ID detection instead of the waiting delay configured
in Step 1.
When enabled, an incoming FXO call is answered on the first occurrence of either:
3.
The reception of the caller ID signal.
The expiration of the delay configured by the Wait Before Answering Delay field and
the country wait before answering delay.
Set the corresponding Wait For Callee To Answer columns drop-down menu with the proper
behaviour.
When the endpoint is set up for automatic call (see Automatic Call on page 423), enabling this
variable makes the endpoint wait until the called the party connection is established before
answering the incoming call.
4.
175
FXO Configuration
FXO Incoming Call Configuration

This section allows you to define how each line behaves when there is an incoming call.
To set FXO incoming call parameters:

1.
In the FXO Incoming Call Behavior section, set the corresponding Not Allowed Behavior columns
drop-down menu with the proper behaviour.
Under certain circumstances (locked port, configuration, etc.), incoming FXO calls are not allowed.
When that is the case, the FXO endpoint behaves in one of the manners below.
Table 143: Not Allowed Behavior Parameters
Parameter
Description
Do Not Answer
The incoming call is left unanswered.
Play Congestion Tone
The incoming call is answered, a congestion tone is played for 10

seconds, and then the call is terminated.
Figure 70: FXO Incoming Call Behavior Section

1
2.
FXO Line Verification

The line state verification mechanism allows the detection of defective or down lines based on the absence of
current when closing the loop. You can view the line state in the FXO Line State table (FXO Line Status on
page 162).
To set FXO line verification parameters:

1.
In the FXO Line Verification section, set the Link State Verification drop-down menu with the proper
behaviour.
Figure 71: FXO Answering Configuration Section
2
3
2.
Set the Link State Verification Timeout field with the value, in milliseconds, indicating how long the
unit waits to successfully take the line before considering the line is defective or down.
3.
FXO Force End of Call

FXO Force end of call, also known as Far End Disconnect, refers to methods for detecting that a remote party
has hung up. If the Far End Disconnect signal is not sent to or properly detected by the Mediatrix unit, the
connection will not be released by the unit, thus freezing the FXO line in the off hook state.
The FXO Force End of Call section allows you to define various methods to detect a far end disconnect.
To set FXO force end of call parameters:

1.
In the FXO Force End of Call section, set the Force End Of Call On Call Failure drop-down menu
with the proper behaviour.
This parameter enables or disables forced-end-of-call on call failure.
The forced end of call on call failure occurs when a SCN caller tries to reach someone on the IP
network and the SCN caller hangs up before reaching the IP callee.
176
FXO Configuration
If the connection is not established, the Mediatrix unit cannot detect that the caller has hung up. You
can configure the Mediatrix unit to disconnect an unsuccessful communication after a specific
number of seconds. This number of seconds is defined in a timeout that starts to count when the
SCN caller contacts the Mediatrix unit. If the connection is not successful, the line is closed when
the call time reaches the defined timeout value.
This feature forcefully terminates a call that stayed in an error state for some time. When the line
falls in an error state where a SIT, a ROH, a BUSY or any error tone is played outbound to the FXO
line, the unit waits for the timeout specified in the Call Failure Timeout field and then hangs up.
Figure 72: FXO Force End of Call Section
1
3
5
2
4
6
7
8
2.
Set the Call Failure Timeout field with the waiting period, in seconds, before terminating a call in an
error state.
3.
Set the Forced End Of Call On Silence Detection Mode drop-down menu with the proper behaviour.
This parameter enables or disables forced-end-of-call on silence detection.
The silence detection feature applies when two parties are in communication and one of them
hangs up. It allows the Mediatrix unit to close a line when no voice activity or silence is detected for
a specified amount of time.
When silence is detected on the IP and/or SCN side for an amount of time specified in a timeout,
the call is terminated. This feature is useful to free resources in the event of an IP network failure
that prevents the end of call to be detected or when the SCN end of call tone was not detected.
Note: The silence detection feature could inadvertently disconnect a communication when one party puts
the other on hold. Using the hold tone can prevent detection of silence when the call is put on hold by the IP
peer. See Tone Override on page 448 for more details.
The current implementation of silence detection relies on the power of the media signal. A silence
is detected if the power level of the media signal is lower than -60 dBm.
This feature forcefully terminates a call that stayed silent for some time. When silence is detected
on the inbound and/or outbound media for an amount of time specified in the Silence Detection
Timeout field, the call is terminated.
Table 144: Forced End of Call on Silence Detection Mode Parameters
Parameter
Disable
Description
Forced-end-of-call on silence detection is disabled.
Inbound And Outbound Silent The call is terminated if both inbound and outbound media are
silent at the same time.
This feature is useful to free resources in the event of a network failure preventing the end-of-call to
be detected or when the FXO end-of-call tone was not detected.
4.
Set the Silence Detection Timeout field with the maximum amount of time, in seconds, that a call
can remain silent, before it is terminated and the line is released.
5.
Set the Forced End Of Call On Tone Detection Mode drop-down menu with the proper behaviour.
177
FXO Configuration
This parameter enables or disables forced-end-of-call upon tone detection. It terminates a call upon
detection of an end-of-call tone on the inbound from the FXO line.
Table 145:
Parameter
Description
Disable
Force-end-of-call upon tone detection is disabled.
Country
Tone
The unit tries to detect the tone specified for this purpose in the current country's
tone specification. Some country specifications omit this information. In that case,
the unit behaves as if the parameter is Disable.
Custom Tone Terminates a call upon detection of a custom tone. See Steps 6-8.
The Mediatrix unit can monitor special tones that indicate the remote SCN user has hung up. You
can use the default value for a selected country or customize a tone. Table 146 lists the default
frequency and cadence detected by supported country.
Table 146: Default Frequency and Cadence Supported
Cadence
Country
Frequency
ON1 (s)
OFF1 (s)
ON2 (s)
OFF2 (s) Repetition
Austria
450 Hz
8.0
0.0
0.0
0.0
Czech Republic
425 Hz
8.0
0.0
0.0
0.0
France
440 Hz
8.0
0.0
0.0
0.0
Germany
425 Hz
8.0
0.0
0.0
0.0
Italy
425 Hz
0.2
0.2
0.6
1.0
North America
440 Hz
8.0
0.0
0.0
0.0
Spain
425 Hz
8.0
0.0
0.0
0.0
Switzerland
425 Hz
8.0
0.0
0.0
0.0
When customizing a tone, Media5 suggests to ask your Central Office about the tone it generates
to indicate a call has been disconnected. You will thus be able to customize your tone according to
this information.
Custom Tone Settings
6.
If you have selected Custom Tone in Step 5, set the Tone Detection Custom Frequency field with
the Frequency, in Hertz (Hz), to detect in the custom cadence.
You can set any value between 350 Hz and 620 Hz.
A customized tone detection can only detect a single frequency. To detect tones made of multiple
frequencies, create the cadence for only one of the frequencies found in the tone.
7.
Set the Tone Detection Custom Cadence field with the cadence to detect.
A cadence is a series of frequencies that are played for a specified time, making up a tone. The
format for a cadence is:
on1,off1,on2,off2,on3,off3
In this string, on and off are numerical values representing the time, in milliseconds, that the
frequency can and cannot be detected, respectively. For instance, 2000, 1000, 2000, 0 is a
cadence in which the frequency plays for 2 seconds, stops for 1 second, and plays for 2 more
seconds. This example is also equivalent to setting the string 2000, 1000, 2000.
You can specify up to three on,off pairs. If you specify less than those six values, 0 values will
be added as necessary. Specifying more than six will only use the six first values.
178
FXO Configuration
A cadence starting with a value of zero (0) is invalid. The first zero (0) found in the string signals the
end of the cadence (i.e. 200, 0, 300 is the same as 200).
To detect a continuous tone, use a single on value, e.g., 200, 0 or 200 (the off time
is 0 ms.). A continuous tone of 200 ms is used if the field is empty.
To detect a tone in which two or more frequencies are used, for instance with a
cadence of 200 ms, 500 ms, 300 ms, 400 ms, in which the respective frequencies
would be 400 Hz, 300 Hz, 400 Hz, 500 Hz, detect a frequency that comes twice or
more in the tone. In the above example, detect the 400 Hz frequency by using the
cadence 200 ms, 400 ms, 300 ms, 500 ms. In this example, the 400 ms and 500 ms
off times represent the times that the 400 Hz frequency cannot be heard, even though
another frequency may be playing.
The on and off values can be from 0 to 32767 ms.
8.
Set the Detection Custom Repetition field with the number of times the custom cadence must be
detected to consider the custom end-of-call tone has been detected.
9.
179
180
FXO Configuration
ISDN Parameters
H A P T E R
21
ISDN Configuration
This chapter describes how to configure the Integrated Services Digital Network (ISDN) Basic Rate Interfaces
(BRI) and/or Primary Rate Interfaces (PRI) parameters of the Mediatrix unit.
Note: This web page is not available on the Mediatrix LP/4100/C7 Series models.
Introduction
ISDN is a set of digital transmission protocols defined by a few international standards body for
telecommunications, such as the ITU-T. One or another of these protocols are accepted as standards by
virtually every telecommunications carrier all over the world.
ISDN replaces the traditional telephone system so that one or two pairs of telephone wires can carry voice and
data simultaneously. It is a fully digital network where all devices and applications present themselves in a
digital form.
ISDN is a User-Network Interface (UNI) signalling protocol with a user and a network side. The user side is
implemented in ISDN terminals (phones, terminal adapters, etc.) while the network side is implemented in the
exchange switches of the network operator. Both sides have different signaling states and messages. The
Mediatrix unit ISDN interfaces can be configured to work as user (TE) or network (NT) interfaces.
Depending on your product, you can configure two types of ISDN interfaces:
The ISDN Basic Rate Interface (BRI) Mediatrix 3404, 3408, 3734, 3741, 3742, and 4400
Series models.
The ISDN Primary Rate Interface (PRI) Mediatrix 3531, 3532, 3621, 3631, 3632, 3731, 3732,
and 3734 models.
ISDN Reference Points

ISDN specifies a number of reference points that define logical interfaces between the various equipment
types on an ISDN access line. The Mediatrix unit supports the following ISDN reference points:
S: The reference point between user terminals and the NT2. This is used in point-to-multipoint
BRI connections.
T: The reference point between NT1 (Modem) and NT2 (PBX) devices. This is used in pointto-point PRI/BRI connections.
All other ISDN reference points are not supported.
179
Chapter 21 - ISDN Configuration
Introduction
Figure 73: ISDN Reference Points

BRI point-to-point
S
T
NT2
TE
NT1
BRI point-to-multipoint (S-bus)
ISDN
S/T
TE
TE
NT1
PRI
S
T
NT2
TE
NT1
Inband Tones Generation

In an ISDN network, most of the call setup tones are played locally by the TE equipment (i.e., telephone
handset), although some require that the tones be played inband by the NT.
When interworking with other networks occurs, the need for the tones to be played inband is more likely to
arise.
The Mediatrix unit may enable inband tones to be played locally, on a per-interface basis. This option is
present when the unit is acting as both the NT and the TE UNI-side. However, in TE mode, only the ringback
tone is played.
The Call Setup tones (dial tone, ringback tone, etc.) are played in the direction where the call has been
initiated. The call disconnection tones are played in both directions, but of course will not arrive to the peer
who disconnected the call.
When an inband tone is played, a Progress Indicator information element (IE) #8 Inband information or
appropriate pattern available is added to the ISDN message corresponding to the call state change, and in a
PROGRESS ISDN message if no state change is occurring.
When an interface is acting as the TE, as soon as the NT advertises that it plays inband tones through a
Progress Indicator IE #8 or #1, the local inband tones generation is disabled for the rest of the call.
Whenever a tone is played inband locally or the ISDN peer advertises that inband informations are available,
the Mediatrix unit is notified. The IP media path can then be opened earlier in the call, and can be closed with
some delay after the call disconnection initiation.
The following table summarizes the inband tones generation behaviour for the NT mode.
Table 144: Inband Tones Generation Behaviour - NT
Inband Tones Generation
Enabled
Signal IE Handling Enabled
Inband Tone Played
No
No
No
No
Yes
Yes
Yes
Dont care
No
The following table summarizes the inband tones generation behaviour for the TE mode.
Table 145: Inband Tones Generation Behaviour - TE
Signal IE
Handling Enabled
No
180
Signal IE
Received
Dont care
Inband Tones
Generation
Enabled
No
NT Peer
Advertised
Inband Tones
Dont care
Inband Tone
Played
No
Setting PRI Hardware Parameters
Table 145: Inband Tones Generation Behaviour - TE (Continued)

Signal IE
Handling Enabled
Inband Tones
Generation
Enabled
Signal IE
Received
NT Peer
Advertised
Inband Tones
Inband Tone
Played
Yes
Yes
No
Dont care
Yes
Yes
No
Dont care
Dont care
No
No
Dont care
Yes
Yes
No
Note that when the PRI/BRI interface Signalling Protocol drop-down is set to QSIG, the Signal IE does not
exist so it has no effect on the inband tones generation. In QSIG, inband tones are played when the inband
tones generation is activated on the incoming side of the call. See PRI Configuration on page 186 or BRI
Signal Handling
The Signal IE is used by the NT ISDN side to tell its TE peers that they must generate an inband tone locally.
Thus, the Signal IEs are sent by the NT only.
When the Signal IE handling is enabled on a given ISDN interface acting as a TE, inband tones are played
towards the IP side when a Signal IE is received. On a NT interface, a Signal IE is inserted in the ISDN
messages sent to the TE when appropriate.
Note that when the Mediatrix 3500 Series signaling protocol is set to NI-2 (National ISDN-2) on that interface,
the Signal IE handling is forced to be enabled for a NT.
Setting PRI Hardware Parameters

You must set hardware-related parameters. You can do so in the System / Hardware page. The Hardware
page differs depending on the product and model you have.
To configure the Mediatrix unit hardware:

1.
2.
If you want to configure the clock reference of a specific interface, you must set the Endpoint Type
drop-down menu to NT. See PRI Configuration on page 186 for more details.
Table 146: Clock Reference
Reference
None
Description
181
ISDN Auto-Configuration
Table 146: Clock Reference (Continued)

Reference
Other
Card
Description
The internal clock synchronizes with the other PRI interface of the Mediatrix unit.
This interface must be configured in TE mode (Endpoint drop-down menu of the
interfaces.
3.
You must restart the unit if you change this setting.
4.
Select the ISDN signaling in the Signaling drop down menu.

instance, if you are in R2 with a route r2-Slot2/E1T1, then change to ISDN, you must change the
route to isdn-Slot2/E1T1.
This parameter is available only for the Mediatrix 3621/3631/3632 models. Other models support
only the ISDN protocol.
5.
ISDN Auto-Configuration
The ISDN Auto-configuration feature allows you to detect and to configure all ISDN interfaces so that the ISDN
link goes up and becomes usable with a minimal user interaction. When launching an auto-configuration
process, it stops automatically when all interfaces have been tested. For each interface, the autoconfiguration process is considered successful when the link becomes up or a failure when all combinations
have been tried without having a link up.
Caution: Launching the auto-configuration may terminate abruptly all ongoing ISDN calls.
Note: Auto-configuration on all ISDN interfaces may take some time to complete. Some of the current ISDN
settings might be replaced by new values.
Please note that some parameters cannot be auto configured. For instance, the clock mode is configured
according to the endpoint type, master for NT and slave for TE.
To launch the auto-configuration process:

1.
In the web interface, click the ISDN link, then the Status sub-link.
Figure 75: ISDN Status Configuration Section
2.
Click the Start Sensing button.

The process starts.
182
Preset
Preset
The ISDN Preset Configuration section allows you to load a set of preset configuration for your ISDN
connections. These preset files are located in the file system's persistent memory. They differ depending on
the Mediatrix unit you are using. Depending on your unit's profile, it may be possible that no preset files are
available.
Using preset files is especially useful for units that do not use the default values provided by Media5 (for
instance, T1 instead of E1 for Mediatrix 3000 units). Please note that only script files work. Any other type of
file present in the file system cannot be run here.
You can also export your current ISDN configuration in a preset. Please note that these user-defined presets
are not kept in the event of a partial or factory reset.
To see the content of the units file system persistent memory, go to File Manager (Chapter 51 - File Manager
on page 613). All installed configuration scripts/images are listed.
To load and execute a preset file:

1.
In the ISDN Status tab, ISDN Preset Configuration section, select one of the available preset files
in the Local Preset drop-down menu.
1
2.
Click Apply.
The configuration is applied.
To export the current ISDN configuration as a preset:

1.
In the ISDN Preset Configuration section, type a name for the preset in the Preset Name field.
2.
Click Save.
The current ISDN configuration is exported. Please note that these user-defined presets are not
kept in the event of a partial or factory reset.
When the clock device is not synchronized, the description value of the file is "Automatically
Generated". When synchronized, the description is "Automatically Genereted on Date/Time". See
the File Manager (Chapter 51 - File Manager on page 613) for more details on how to see and
manage the files in the units file system.
Partial Reset
When a partial reset is triggered, the user-defined presets are deleted.
PRI ISDN Statistics

The Mediatrix unit collects meaningful statistics for each PRI digital card that can be read via the web interface.
These statistics are also available via SNMP and CLI.
183
PRI ISDN Statistics
The Mediatrix unit collects statistics for each of its two cards, if available. Slot 2 and Slot 3 indicate the physical
location of the cards in the unit, Slot 2 being on the left when looking at the rear of the unit. You can click the
Reset Stats button at any time to reset all statistics for the specified interface.
Figure 78: ISDN Statistics Web Page
The following table describes the statistics available.

Table 147: ISDN Statistics Displayed
Statistic
TxFrames
Description
Number of HDLC frames transmitted.
Note: The term frames does not refer to the structure defined in I.431.
RxFrames
Number of HDLC frames received.

TxOctets
Number of octets transmitted. This value is obtained by cumulating the octets

transmitted in the HDLC frames.
RxOctets
Number of octets received. This value is obtained by cumulating the octets received
in the HDLC frames.
FcsErrors
Frame check sequence (FCS) errors indicate that frames of data are being
corrupted during transmission. FCS error count is the number of frames that were
received with a bad checksum (CRC value) in the HDLC frame. These frames are
dropped and not propagated in the upper layers.
This value is available on E1 and T1.
FramesDropped
Number of frames dropped. This value is obtained by cumulating the number of

frames dropped when transferring the data from the framer chip to the device
internal buffer.
OctetsDropped
Number of octets dropped. This value is obtained by cumulating the number of

octets dropped when transferring the data from the framer chip to the device internal
buffer.
184
PRI ISDN Statistics
Table 147: ISDN Statistics Displayed (Continued)

Statistic
NegFrameSlipsTx
Description
A frame is skipped when the frequency of the transmit clock is greater than the
frequency of the transmit system interface working clock based on 2.048 MHz (on
E1) or 1.544 MHz (on T1).
NegFrameSlipsRx
A frame is skipped when the frequency of the received route clock is greater than
the frequency of the receive system interface working clock based on 2.048 MHz (on
E1) or 1.544 MHz (on T1).
PosFrameSlipsTx
A frame is repeated when the frequency of the transmit clock is less than the
frequency of the transmit system interface working clock based on 2.048 MHz (on
E1) or 1.544 MHz (on T1).
PosFrameSlipsRx
A frame is repeated when the frequency of the receive route clock is less than the
frequency of the receive system interface working clock based on 2.048 MHz (on
E1) or 1.544 MHz (on T1).
FramingErrors
The framing error count indicates that a FAS (Frame Alignment Signal) word has
been received with an error.
The FAS-bits are present in every even frame of timeslot 0 on E1.
The FAS-bits are present in ESF format on T1.
CodeViolations
The code violations count indicates that an encoding error on the PCM line has been
detected.
CRCErrors
The CRC errors count is incremented when a multiframe has been received with a
CRC error.
The CRC error count is available in CRC multiframe mode only on E1.
The CRC error count is in ESF format on T1.
EBitErrors
The E-Bit error count gives information about the outgoing transmit PCM line if the
E-bits are used by the remote end for submultiframe error indication. Incrementing is
only possible in the multiframe synchronous state.
Due to signaling requirements, the E-bits of frame 13 and frame 15 of the CRC
multiframe can be used to indicate an error in a received submultiframes:
Submultiframe I status E-bit located in frame 13
Submultiframe II status E-bit located in frame 15
no CRC error : E = 1
CRC error : E = 0
This value is only available in E1.

BlockErrors
The Block Error count is incremented once per multiframe if a multirame has been
received with a CRC error or a bad frame alignment has been detected.
This value is only available for ESF format on T1 only.
185
PRI Configuration
PRI Configuration
Standards Supported
ITU-T Recommendation G.703: Physical/Electrical

Characteristics of Hierarchical Digital Interfaces
ITU-T Recommendation G.704: Synchronous Frame

Structures Used at Primary and Secondary Hierarchy Levels
ITU-T Recommendation I.431: Primary rate user-network

interface - Layer 1 specification
ITU-T Recommendation Q.921: ISDN user-network interface

- Data link layer specification

layer 3 specification for basic call control
This section applies to the following models:
Mediatrix 3531
Mediatrix 3532
Mediatrix 3621
Mediatrix 3631
Mediatrix 3632
Mediatrix 3731
Mediatrix 3732
Mediatrix 3734
The Primary Rate Interface (PRI) port supports 30 x 64 kbit/s B-channels, 1 x 64 kbit/s D-channel and 1 x
synchronization timeslot on a standard E1 (G.704) physical layer. In its T1 version, a PRI interface supports
23 x B-channels and 1 x D-channel, all at a 64 kbit/s rate. E1 is mostly deployed in Europe, while T1 is more
present in North America.
Caution: You can configure ISDN ports while they are active. However they are internally disabled to
modify the configuration and then re-enabled. All active calls on the port are dropped during this process.
Configuration changes should only be performed during planned down times. Most of the ISDN parameters
change require a restart of the ISDN service to be applied.
Caution: The Mediatrix unit PRI ports can be used as a T reference point, but not as U reference points
(2-wire). Never connect a U SCN line or a U TE into the Mediatrix unit PRI ports.
186
PRI Configuration
To configure the PRI parameters:

1.
In the web interface, click the ISDN link, then the Primary Rate Interface sub-link.
Figure 79: ISDN Interface Configuration Section
2
4
6
8
10
12
5
7
9
11
13
14
15
16
17
18
20
19
21
22
24
23
25
26
28
27
29
30
32
31
33
34
36
38
35
37
2.
Select to which interface you want to apply the changes in the Select Interface drop-down menu at
the top of the window.
You can copy the configuration of the selected interface to one or more interfaces of the Mediatrix
unit in the Apply to the Following Interfaces section at the bottom of the page. You can select
specific interfaces by checking them, as well as use the Check All or Uncheck All buttons.
The Mediatrix 3532 and 3632 models have two interfaces.
3.
If applicable, use the Line Type Configure link to set the line type, as described in Setting PRI
Hardware Parameters on page 181.
4.
Select the endpoint type in the Endpoint Type drop-down menu.

Table 148: Endpoint Type
Type
TE
Description
Terminal Equipment.The endpoint emulates the subscriber (terminal) side of the digital
connection. You can connect the SCN to the endpoint.
187
PRI Configuration
Table 148: Endpoint Type (Continued)

Type
NT
Description
Network Termination. The endpoint emulates the central office (network) side of the
digital connection. You can connect a PBX to the endpoint.
The setting used for the Mediatrix unit must be opposite to the setting used in the PBX. For instance,
if the PBX is set to TE, then the Mediatrix unit must be set to NT.
Note: If you want to use a specific interface as the reference clock, you must set it to TE.
When the PRI interface Signalling Protocol drop-down is set to QSIG (see Step 9), the endpoint type
is only used in the second layer (LAPD) since it is a concept that does not exist in QSIG.
5.
Select the clock mode of the interface in the Clock Mode drop-down menu.
The interface can either generate the clocking for the line or accept the clock from the line.
Table 149: Clock Mode
Mode
auto
Description
The setting is derived from the endpoint type.
NT: clock master
TE: clock slave
Master The interface generates the clock.

Slave
The interface accepts the clock from the line.
The clock mode is used to give the user the possibility to set an endpoint in TE mode and still
generate the clock by specifying the clock mode to master. The clock source can then be selected
from the Clock Reference drop-down menu (see Setting PRI Hardware Parameters on page 181
for more details). The clock mode could be used, for instance, to synchronize several units in NT
mode via a T1 or E1 line.
6.
Select the port pinout in the Port Pinout drop-down menu.

Table 150: Port Pinout
Mode
7.
Description
Auto
The pinout is set according to the Endpoint Type parameter setting (Step 4).
Te
Forces the pinout to TE regardless of the Endpoint Type value.
Nt
Forces the pinout to NT regardless of the Endpoint Type value.
Set the Monitor Link State drop-down menu with the physical link state of the ISDN interface.
Table 151: Interface Link State
Parameter
Description
Enable
The ISDN endpoint's operational state is affected by its interface physical link state.
When the link state of the ISDN interface is down, the operational state of its
matching endpoint becomes disable.
Disable
The ISDN endpoint's operational state is not affected by its interface physical link
state.
Note that if the Monitor Link State parameter is enabled and the Ignore SIP OPTONS on no usable
endpoints parameter is also enabled in the SIP / Interop page, this will influence how the SIP options
are answered. See SIP Interop on page 316 for more details.
188
PRI Configuration
8.
Select the transmission encoding of bits in the Line Coding drop-down menu.
Table 152: Transmission Encoding
Coding
Description
B8ZS
Bipolar with 8-Zeros Substitution (T1 lines).
HDB3
High-Density Bipolar with 3-zeros (E1 lines).
AMI
Alternate Mark Inversion (E1 and T1 lines).
Make sure that the transmission encoding matches with the remote system. For further information,
see ITU-T Recommendation G.703.
9.
Select the frame format in the Line Framing drop-down menu.

Line Framing is used to synchronize the channels on the frame relay circuit (when a frame starts
and finishes). Without it, the sending and receiving equipment would not be able to synchronize
their frames.
Table 153: Line Framing
Format
Description
SF
Super frame. Sometimes known as D4 (T1 lines).
ESF
Extended super frame (T1 lines).
CRC4
Cyclic redundancy check 4 (E1 lines).
NO-CRC4 No Cyclic redundancy check 4 (E1 lines).

For further information, see ITU-T Recommendation G.704.
10.
Select the protocol to use for the signalling channel in the Signalling Protocol drop-down menu.
This signalling must match the connected ISDN equipment or network.
Table 154: Signalling Protocols
Protocol
Description
DSS1
Digital Subscriber Signaling System No.1
DMS100
Digital Multiplex System 100
NI2
National ISDN No.2
5ESS
5 Electronic Switching System
QSIG
ECMA's protocol for Private Integrated Services

Networks
The 5ESS and DMS100 protocols support basic call only and the model is derived directly from the
DSS1 protocol.
If you select the NI2 protocol, see InformationFollowing Operation on page 196 for the behaviour
if this parameter is present.
11.
Select the value of the network location in the progress indicator messages that the unit sends in
the Network Location drop-down menu.
This defines the location code to be inserted in ISDN causes code information elements, i.e., the
type of network to which the system belongs. The following values are available:
User
Private
Public
Transit
189
PRI Configuration
12.
International
Set the Preferred Encoding Scheme drop-down menu with the data encoding scheme in the bearer
capabilities (user information layer 1 protocol).
This encoding scheme is used when initiating a call on the ISDN side. The supported encoding
schemes are G.711 u-Law and G.711 a-Law.
G.711 u-Law may not be supported by DSS1 NT and TE endpoints. It is recommended to use G.711
a-Law as preferred encoding protocol.
13.
Set the Fallback Encoding Scheme drop-down menu with the fallback data encoding scheme in
case the preferred encoding scheme is not available.
The supported encoding schemes are G.711 u-Law and G.711 a-Law.
Note: The fallback encoding scheme is valid only when receiving a SETUP message. The user sending the
SETUP message does not indicate alternative bearer capability.
If the proposed encoding scheme in the bearer capability received in the SETUP message is
different than the preferred encoding scheme, then the fallback encoding scheme is used.
14.
Define the range of active bearer channels in the Channel Range field.
15.
Define the range to reserve channels for incoming calls in the Channels Reserved for Incoming
Calls field.
Bearer channels are by default usable for both incoming and outgoing calls. Use this range to
reserve channels for incoming calls.
Note:
Channels outside of the range defined by the Channel Range field are ignored.
Channels reserved in both the Channels Reserved for Incoming Calls and Channels Reserved for
Outgoing Calls fields are considered usable for both incoming and outgoing calls.
The string has the following syntax:
',': Separator between non-consecutive lists of channels or single channels.
'n': A single channel, where n is the channel number.
'm-n': List of channels where m is the start channel number and n is the end channel
number.
Note: The space character is ignored and duplication is not allowed. Channels must be specified in low to
high order.
Example: '1,12-15': The accepted channels are 1, 12, 13, 14 and 15.
16.
Define the range to reserve channels for outgoing calls in the Channels Reserved for Outgoing Calls
field.
Bearer channels are by default usable for both incoming and outgoing calls. Use this range to
reserve channels for outgoing calls.
Note:
Channels outside of the range defined by the Channel Range field are ignored.
Channels reserved in both the Channels Reserved for Incoming Calls and Channels Reserved for
Outgoing Calls fields are considered usable for both incoming and outgoing calls.
190
',': Separator between non-consecutive lists of channels or single channels.
'n': A single channel, where n is the channel number.
PRI Configuration
'm-n': List of channels where m is the start channel number and n is the end channel
number.
Note: The space character is ignored and duplication is not allowed. Channels must be specified in low to
high order.
Example: '1,12-15': The accepted channels are 1, 12, 13, 14 and 15.
17.
Select the strategy for selecting bearer channels in the Channel Allocation Strategy drop-down
menu.
Table 155: Channel Allocation Strategy
Allocation
Description
Ascending
Starting from the lowest-numbered non-busy bearer channel and

going toward the highest-numbered non-busy bearer channel, the
Mediatrix unit selects the first bearer channel available.
Descending
Starting from the highest-numbered non-busy bearer channel and

going toward the lowest-numbered non-busy bearer channel, the
RoundRobinAscending
The Mediatrix unit starts from the bearer channel that follows the
bearer channel used for the last call. For instance, if channel #1 was
used in the last call, the unit starts with channel #2. Going toward the
highest-numbered non-busy bearer channel, the unit selects the first
channel available. If the highest channel is unavailable, the search
continues from the lowest-numbered non-busy bearer channel.
RoundRobinDescending The Mediatrix unit starts from the bearer channel that precedes the
lowest-numbered non-busy bearer channel, the unit selects the first
channel available. If the lowest channel is unavailable, the search
continues from the highest-numbered non-busy bearer channel.
18.
Define the maximum number of active calls on the interface in the Maximum Active Calls field.
This limits the total number of concurrent calls on the interface. Entering 0 indicates no maximum
number of active calls.
19.
20.
Select whether or not the signal information element is enabled in the Signal Information Element
drop-down menu.
When activated at the Network UNI-side (Endpoint Type drop-down menu set to NT),
the signal information element is sent to the User UNI-side.
When activated at the User UNI-side (Endpoint Type drop-down menu set to TE), the
tone indicated in the signal information element is played in the IP direction.
Select whether or not inband tone generation is enabled in the Inband Tone Generation drop-down
menu.
When activated at the User UNI-side (Endpoint Type drop-down menu is set to TE), only the
ringback tone is generated.
When the Signalling Protocol drop-down menu is set to QSIG (Step 9) and this variable is activated,
the incoming side of the call plays the tones inband.
21.
Select whether or not inband DTMF dialing is enabled in the Inband DTMF Dialing drop-down menu.
If you select Enable, the Mediatrix unit accepts inband DTMF digits when Overlap Dialing occurs.
191
22.
PRI Configuration
Select whether or not overlap dialing is enabled in the Overlap Dialing drop-down menu.
Table 156: Overlap Dialing Parameters
Parameter
23.
Description
Enable
The Mediatrix unit transports the called-party number digit by digit, after the first
SETUP message, which contains no called party information at all.
Disable
The Mediatrix unit transports the full called party information in the first SETUP
message from the terminal. This means that the user must dial the number before
going off-hook.
Define the maximum length of the calling party name for calls from SIP to ISDN in the Calling Name
Max Length field.
Available values range from 0 to 82.
24.
Select whether or not exclusive B-Channel selection is enabled for calls from SIP to ISDN in the
Exclusive B-Channel Selection drop-down menu.
If you select Enable and initiate a call, only the requested B channel is accepted; if the requested
B channel is not available, the call is cleared.
25.
Select whether or not to enable the Sending Complete information element into SETUP messages
for calls from SIP to ISDN in the Sending Complete drop-down menu.
Some ISDN switches may require that the Sending Complete information element be included in
the outgoing SETUP message to indicate that the entire number is included and there are no further
destination digits to be sent.
26.
Select whether or not to enable sending the RESTART message upon a signalling channel UP
event in the Send Restart On Startup drop-down menu.
The RESTART message requests a restart for the interface specified.
27.
Set the link establishment strategy in the Link Establishment drop-down menu.
Table 157: Link Establishment Parameters
Parameter
Description
OnDemand When the data link is shut down, the unit establishes a new link only when
required.
Permanent
28.
When the data link is shut down, the unit immediately attempts to establish a new
link.
Set the STATUS causes that can be received without automatically clearing the call in the Accepted
Status Causes field.
The default action is to clear the call upon receiving a STATUS message. If a STATUS message is
received indicating a compatible call state and containing the supplied STATUS causes, the
clearing of the call is prevented.
',': Separator between non-consecutive lists of causes or single cause.
'n': A single cause, where n is the cause number.
'm-n': List of causes where m is the start cause number and n is the end cause number.
Note: The space character is ignored and cause duplication is not allowed.
Causes must be specified in low to high order.
Example: '1,124-127': The accepted causes are 1, 124, 125, 126 and 127.
192
PRI Configuration
29.
Set the range of PROGRESS causes accepted by the unit in the Accepted Progress Causes field.
Causes excluded from this range trigger call disconnections.
Note: You must consider the following:

The space character is not allowed.
Cause duplication is not allowed.
30.
Select the strategy for sending ISDN Progress messages in the Send Isdn Progress drop-down
menu.
Table 158: Send ISDN Progress Parameters
Parameter
Description
Send All
Send an ISDN Progress message in all situations where call progression is

signaled.
Send
Inband
Send an ISDN Progress message only when call progression contains an

indication of in-band information.
Send
Alerting
Send an ISDN Alerting message instead of ISDN Progress message when call
progression contains an indication of in-band information. If call progression does
not contain in-band information, no message is sent at this step.
The strategy for sending Progress messages should be adapted to the configuration of the peer
ISDN switch. Some switches may terminate calls when receiving one or many ISDN progress
messages.
31.
Select the strategy for sending the Progress Indicator Information Element in the Send Progress
Indicator IE drop-down menu.
Table 159: Send Progress Indicator IE Parameters
Parameter
Description
Send All
Send the Progress Indicator IE in all situations.
Send
Inband
Only
Send the Progress Indicator only when the Progress Description contains an
The strategy for the Progress Indicator IE should be adapted to the configuration of the peer ISDN
switch.
This parameter controls sending of a Progress Indicator IE in ISDN messages where Progress
Indicators are allowed. See the parameters in Interop Parameters Configuration on page 208 for
a control over which ISDN message allows Progress Indicators.
32.
Select the default value to insert in the "Type of Number" parameter of the "Calling Party Number"
IE when "Type of Number" is not already defined in the call properties in the Default TON for Calling
Party Number IE drop-down menu.
Note: A call property set by the "Properties Manipulation" feature of the Call Router has precedence over
this default value. See Chapter 42 - Call Router Configuration on page 467 for more details.
193
PRI Configuration
This parameter applies to the outgoing ISDN calls. Possible values are:
Table 160: Default TON for Calling Party Number IE Parameters
Parameter
Unknown
Description
Default value of Type of Number is set to Unknown.
International Default value of Type of Number is set to International.
33.
National
Default value of Type of Number is set to National.
Network
Specific
Default value of Type of Number is set to Network-Specific.
Subscriber
Default value of Type of Number is set to Subscriber.
Abbreviated
Default value of Type of Number is set to Abbreviated.
Select the default value to insert in the "Numbering Plan Identification" parameter of the "Calling
Party Number" IE when "Numbering Plan Identification" is not already defined in the call properties
in the Default NPI for Calling Party Number IE drop-down menu.
Table 161: Default NPI for Calling Party Number IE Parameters
Parameter
34.
Description
Unknown
Default value of Numbering Plan Identification is set to Unknown.
ISDN
Telephony
Default value of Numbering Plan Identification is set to ISDN Telephony.
Data
Default value of Numbering Plan Identification is set to Data.
Telex
Default value of Numbering Plan Identification is set to Telex.
National
Standard
Default value of Numbering Plan Identification is set to National Standard.
Private
Default value of Numbering Plan Identification is set to Private.
Select the default value to insert in the "Presentation Indicator" parameter of the "Calling Party
Number" IE when "Presentation Indicator" is not already defined in the call properties in the Default
PI for Calling Party Number IE drop-down menu.
If "Presentation Indicator" is not provided by the call properties, its value is determined by the
following two steps.
194
PRI Configuration
a.
First, it is set to the default value defined by "DefaultCallingPi".
b.
Second, the "Presentation Indicator" can be overridden by the CLIP and CLIR services: the
value can be set to "Restricted" by the CLIR service and the value can be set to "NotAvailable"
if there is no number to forward. This variable applies to the outgoing ISDN calls.
Possible values are:

Table 162: Default PI for Calling Party Number IE Parameters
Parameter
Description
Presentation
Allowed
Default value of Presentation Indicator is set to Presentation Allowed.
Presentation
Restricted
Default value of Presentation Indicator is set to Presentation Restricted.
Not Available Default value of Presentation Indicator is set to Not Available.

35.
Select the default value to insert in the "Screening Indicator" parameter of the "Calling Party
Number" IE when "Screening Indicator" is not already defined in the call properties in the Default SI
for Calling Party Number IE drop-down menu.
Table 163: Default SI for Calling Party Number IE Parameters
Parameter
Description
User Provided Not Screened Default value of Screening Indicator is set to User Provided Not
Screened.
36.
User Provided Verified And

Passed
Default value of Screening Indicator is set to User Provided

Verified And Passed.

Failed

Verified And Failed.
Network Provided
Default value of Screening Indicator is set to Network Provided.
Context Dependent
Screening Indicator is set to the value that makes the most sense
according to run-time context.
Select the default value to insert in the "Type of Number" parameter of the "Called Party Number"
IE when "Type of Number" is not already defined in the call properties in the Default TON for Called
Table 164: Default TON for Called Party Number IE Parameters
Parameter
Unknown
Description

National
195
PRI Configuration
Table 164: Default TON for Called Party Number IE Parameters (Continued)
Parameter
37.
Description
Network
Specific
Subscriber
Abbreviated
Select the default value to insert in the "Numbering Plan Identification" parameter of the "Called
in the Default NPI for Called Party Number IE drop-down menu.
Table 165: Default NPI for Called Party Number IE Parameters
Parameter
38.
Description
Unknown
ISDN
Telephony
Data
Telex
National
Standard
Private
Select the unit's behaviour when it receives a Notification Indicator IE with the description set to
User suspended in the Notification User Suspended drop-down menu.
Table 166: Notification User Suspended Parameters
Parameter
39.
Description
Ignore
The Mediatrix unit ignores the Notification Indicator IE with description set to
User suspended.
Disconnect
The Mediatrix unit disconnects the call on Notification Indicator IE with

description set to User suspended.
InformationFollowing Operation
The "informationFollowing" operation is supported for NI2 signaling only (see PRI Configuration on page 186
for more details).
When a SETUP message is received containing an "informationFollowing" operation, the unit immediately
sends a PROCEEDING message. The unit then waits normally for a FACILITY message containing the calling
party name, for a maximum time configured with the Maximum Facility Waiting Delay parameter (see Interop
Parameters Configuration on page 208 for more details).
The only difference between this behaviour and the usual behaviour (i.e. without the "informationFollowing"
operation), is the immediate sending of the PROCEEDING message before waiting for the calling party name.
196
BRI Configuration
Note that the "informationFollowing" operation is mutually exclusive with the Call Proceeding Delay parameter
(see Interop Parameters Configuration on page 208 for more details), which configures a delay before
sending the PROCEEDING message. If the PROCEEDING message is sent due to the "informationFollowing"
operation, Call Proceeding Delay parameter is ignored.
BRI Configuration
Standards Supported


ITU-T Recommendation I.430: Basic user-network interface Layer 1 specification

- Data link layer specification

This section applies to the following models:
Mediatrix 3404
Mediatrix 3408
Mediatrix 3734
Mediatrix 3741
Mediatrix 3742
A Basic Rate Interface (BRI) port supports 2 x 64 kbit/s B-channels for switched voice or data connections and
1 x 16 kbit/s D-channel for signalling.
Caution: The Mediatrix unit ISDN BRI ports are configurable to operate as network or terminal ports. The
pin-out of the sockets is switched according to this configuration. Wrong port configurations, wrong cabling
or wrong connections to neighbouring equipment can lead to short circuits in the BRI line powering. Refer
to the Hardware Installation Guide to avoid misconfigurations.
Caution: The Mediatrix unit BRI ports can be used as a S or T reference point, but not as U reference points
(2-wire). Never connect a U SCN line or a U TE into the Mediatrix unit BRI ports.
The Mediatrix 3404 / 3734 / 3741 / 3742 has 5 ISDN BRI ports. It supports up to 8 simultaneous ISDN voice/
data channels over any IP connection.
The Mediatrix 3408 has 10 ISDN BRI ports. It supports up to 16 simultaneous ISDN voice/data channels over
any IP connection.
The Mediatrix 4400 Series has up to 4 ISDN BRI ports depending on the model. It supports up to 8
simultaneous ISDN voice/data channels over any IP connection.
197
BRI Configuration
To configure the BRI parameters:

1.
In the web interface, click the ISDN link, then the Basic Rate Interface sub-link.
Figure 80: ISDN Basic Rate Interface Web Page
2
3
5
4
6
7
8
9
10
11
12
13
15
17
19
21
23
25
14
16
18
20
22
24
26
27
28
29
31
30
32
33
34
2.
In the Interface Configuration section of the Basic Rate Interface page, select to which interface you
want to apply the changes in the Select Interface drop-down menu at the top of the window.
The Mediatrix 3404 model has 5 interfaces in Slot 1, while the Mediatrix 3408 model has 10
interfaces in Slots 1 and 2 (5 in each).
3.
Select the endpoint type in the Endpoint Type drop-down menu.

Table 167: Endpoint Type
Type
198
Description
TE
Terminal Equipment.The endpoint emulates the subscriber (terminal) side of the digital
connection. You can connect the SCN to the endpoint.
NT
Network Termination. The endpoint emulates the central office (network) side of the
digital connection. You can connect a PBX or ISDN telephones to the endpoint.
BRI Configuration
The setting used for the Mediatrix unit must be opposite to the setting used in the PBX. For instance,
if the PBX is set to TE, then the Mediatrix unit must be set to NT.
Note: If you want to use a specific interface as the reference clock, you must set it to TE.
When the BRI interface Signalling Protocol drop-down is set to QSIG (see Step 7), the endpoint type
is only used in the second layer (LAPD) since it is a concept that does not exist in QSIG.
4.
Table 168: Clock Mode
Mode
auto
Description
The setting is derived from the endpoint type.
NT: clock master
TE: clock slave

Slave
The clock mode is used to give the user the possibility to set an endpoint in TE mode and still
generate the clock by specifying the clock mode to master. The clock source can then be selected
from the Clock Reference drop-down menu (see Chapter 5 - Hardware Parameters on page 61
for more details). The clock mode could be used, for instance, to synchronize several units in NT
mode via a BRI line.
Note: In a BRI configuration, setting the clock mode to master for a TE endpoint is invalid. Slave mode is
automatically applied in this case.
5.
Set the Monitor Link State drop-down menu with the physical link state of the ISDN interface.
Parameter
Description
Enable
The ISDN endpoint's operational state is affected by its interface physical link state.
When the link state of the ISDN interface is down, the operational state of its
matching endpoint becomes disable.
Disable
The ISDN endpoint's operational state is not affected by its interface physical link
state.
6.
Select the connection type of the endpoint in the Connection Type drop-down menu.
199
BRI Configuration
The type of connection depends on the equipment to which the Mediatrix unit port is connected and
it must be the same for all interconnected pieces of equipment.
Table 170: Connection Type
Type
Description
Point to Point
Can only attach one device (for instance a PBX or SCN) and acts as a T
reference point.
Point to
MultiPoint
Can attach more than one ISDN device and acts as a S reference point. Up
to 8 TEs and one NT can be connected to a S-bus.
Note: If you are using a Mediatrix unit connected to a S-Bus in point-to-multipoint TE mode, you cannot
currently connect any additional ISDN devices to the S-Bus.
The Point to MultiPoint configuration is not available in QSIG.
7.
Select the protocol to use for the signalling channel in the Signalling Protocol drop-down menu.
This signalling must match the connected ISDN equipment or network.
Table 171: Signalling Protocols
Protocol
Description
DSS1
DMS100
Digital Multiplex System 100
NI2
National ISDN No.2
5ESS
5 Electronic Switching System
QSIG
ECMA's protocol for Private Integrated

Services Networks
Note: The Dgw v2.0 Application currently supports only the DSS1 and QSIG signalling protocols.
8.
Select the value of the network location in the progress indicator messages that the unit sends in
the Network Location drop-down menu.
This defines the location code to be inserted in ISDN causes code information elements, i.e., the
type of network to which the system belongs. The following values are available:
9.
User
Private
Public
Transit
International
Set the Preferred Encoding Scheme drop-down menu with the data encoding scheme in the bearer
capabilities (user information layer 1 protocol).
This encoding scheme is used when initiating a call on the ISDN side. The supported encoding
G.711 u-Law may not be supported by DSS1 NT and TE endpoints. It is recommended to use G.711
a-Law as preferred encoding protocol.
10.
200
Set the Fallback Encoding Scheme drop-down menu with the fallback data encoding scheme in
case the preferred encoding scheme is not available.
BRI Configuration
The supported encoding schemes are G.711 u-Law and G.711 a-Law.
Note: The fallback encoding scheme is valid only when receiving a SETUP message. The user sending the
SETUP message does not indicate alternative bearer capability.
If the proposed encoding scheme in the bearer capability received in the SETUP message is
different than the preferred encoding scheme, then the fallback encoding scheme is used.
11.
menu.
Allocation
Description
Ascending

Descending

RoundRobinAscending
12.
For a Mediatrix 3404 / 3408, the maximum number of simultaneous calls is limited to 8, even if 5
BRI ports can physically support 10 calls.
13.
14.
Select whether or not the signal information element is enabled in the Signal Information Element
drop-down menu.
When activated at the Network UNI-side (Endpoint Type drop-down menu set to NT),
the signal information element is sent to the User UNI-side.
When activated at the User UNI-side (Endpoint Type drop-down menu set to TE), the
tone indicated in the signal information element is played in the IP direction.
Select whether or not inband tone generation is enabled in the Inband Tone Generation drop-down
menu.
When activated at the User UNI-side (Endpoint Type drop-down menu is set to TE), only the
ringback tone is generated.
When the BRI interface Signalling Protocol drop-down is set to QSIG (see Step 7) and this
parameter is activated, the incoming side of the call plays the tones inband.
15.
Select whether or not inband DTMF dialing is enabled in the Inband DTMF Dialing drop-down menu.
If you select Enable, the Mediatrix unit accepts inband DTMF digits when Overlap Dialing occurs.
201
16.
BRI Configuration
Select whether or not overlap dialing is enabled in the Overlap Dialing drop-down menu.
Table 173: Overlap Dialing Parameters
Parameter
17.
Description
Enable
The Mediatrix unit transports the called-party number digit by digit, after the first
SETUP message, which contains no called party information at all.
Disable
The Mediatrix unit transports the full called party information in the first SETUP
message from the terminal. This means that the user must dial the number before
going off-hook.
Define the maximum length of the calling party name for calls from SIP to ISDN in the Calling Name
Max Length field.
Available values range from 0 to 82.
18.
Select whether or not exclusive B-Channel selection is enabled for calls from SIP to ISDN in the
Exclusive B-Channel Selection drop-down menu.
If you select Enable and initiate a call, only the requested B channel is accepted; if the requested
B channel is not available, the call is cleared.
19.
Select whether or not to enable the Sending Complete information element into SETUP messages
for calls from SIP to ISDN in the Sending Complete drop-down menu.
Some ISDN switches may require that the Sending Complete information element be included in
the outgoing SETUP message to indicate that the entire number is included and there are no further
destination digits to be sent.
20.
Select whether or not to enable sending the RESTART message upon a signalling channel UP
event in the Send Restart On Startup drop-down menu.
The RESTART message requests a restart for the interface specified.
21.
Set the link establishment strategy in the Link Establishment drop-down menu.
Table 174: Link Establishment Parameters
Parameter
Description
OnDemand When the data link is shut down, the unit establishes a new link only when
required.
Permanent
22.
When the data link is shut down, the unit immediately attempts to establish a new
link.
Set the actual keypad string that is to be considered as a hook-flash in the Hook-Flash Keypad field.
An ISDN telephone may send INFORMATION messages that contain a Keypad Facility. You can
thus trigger a supplementary service (Hold, Conference, etc.) by sending a keypad facility.
Since the keypads can be received via several INFORMATION messages, they are accumulated
until they match or reset if the keypad reception timeout (second) has elapsed since the last keypad
has been received. The keypad reception timeout can only be modified via SNMP. If the keypad
reception timeout is set to 0, it disables the timeout, thus assuming that all keypads will be received
in a single INFORMATION message.
Setting this variable to an empty string disables the hook-flash detection.
The permitted keypad must be made up of IA5 characters. See ITU-T Recommendation T.50.
23.
Set the STATUS causes that can be received without automatically clearing the call in the Accepted
Status Causes field.
The default action is to clear the call upon receiving a STATUS message. If a STATUS message is
received indicating a compatible call state and containing the supplied STATUS causes, the
clearing of the call is prevented.
202
BRI Configuration
',': Separator between non-consecutive lists of causes or single cause.</li>
'n': A single cause, where n is the cause number.</li>
Note: The space character is ignored and cause duplication is not allowed.
24.
Set the range of PROGRESS causes accepted by the unit in the Accepted Progress Causes field.
Causes excluded from this range trigger call disconnections.
Note: You must consider the following:

The space character is not allowed.
Cause duplication is not allowed.
25.
Select the strategy for sending ISDN Progress messages in the Send Isdn Progress drop-down
menu.
Table 175: Send ISDN Progress Parameters
Parameter
Description
Send All
Send an ISDN Progress message in all situations where call progression is

signaled.
Send
Inband
Send an ISDN Progress message only when call progression contains an

Send
Alerting
Send an ISDN Alerting message instead of ISDN Progress message when call
progression contains an indication of in-band information. If call progression does
not contain in-band information, no message is sent at this step.
The strategy for sending Progress messages should be adapted to the configuration of the peer
ISDN switch. Some switches may terminate calls when receiving one or many ISDN progress
messages.
26.
Select the strategy for sending the Progress Indicator Information Element in the Send Progress
Indicator IE drop-down menu.
Table 176: Send Progress Indicator IE Parameters
Parameter
Description
Send All
Send the Progress Indicator IE in all situations.
Send
Inband
Only
Send the Progress Indicator only when the Progress Description contains an
The strategy for the Progress Indicator IE should be adapted to the configuration of the peer ISDN
switch.
203
BRI Configuration
This parameter controls sending of a Progress Indicator IE in ISDN messages where Progress
Indicators are allowed. See the parameters in Interop Parameters Configuration on page 208 for
a control over which ISDN message allows Progress Indicators.
27.
Set the TEI Negotiation drop-down menu with the proper Terminal Endpoint Identifier (TEI)
negotiation strategy.
Table 177: TEI Negotiation Parameters
Parameter
Description
Link Up
Each time the physical link comes up, the unit renegotiates the TEI value.
Power Up
When the physical link comes up, the unit does not renegotiate the TEI value. The
value obtained at power-up is reused.
Note: This parameter only applies on Point To Multipoint connections.

28.
Select the default value to insert in the "Type of Number" parameter of the "Calling Party Number"
IE when "Type of Number" is not already defined in the call properties in the Default TON for Calling
Table 178: Default TON for Calling Party Number IE Parameters
Parameter
Unknown
Description
29.
National
Network
Specific
Subscriber
Abbreviated
Select the default value to insert in the "Numbering Plan Identification" parameter of the "Calling
in the Default NPI for Calling Party Number IE drop-down menu.
Table 179: Default NPI for Calling Party Number IE Parameters
Parameter
204
Description
Unknown
ISDN
Telephony
Data
Telex
BRI Configuration
Table 179: Default NPI for Calling Party Number IE Parameters (Continued)
Parameter
30.
Description
National
Standard
Private
Select the default value to insert in the "Presentation Indicator" parameter of the "Calling Party
Number" IE when "Presentation Indicator" is not already defined in the call properties in the Default
PI for Calling Party Number IE drop-down menu.
If "Presentation Indicator" is not provided by the call properties, its value is determined by the
following two steps.
a.
First, it is set to the default value defined by "DefaultCallingPi".
b.
Second, the "Presentation Indicator" can be overridden by the CLIP and CLIR services: the
value can be set to "Restricted" by the CLIR service and the value can be set to "NotAvailable"
if there is no number to forward. This variable applies to the outgoing ISDN calls.

Table 180: Default PI for Calling Party Number IE Parameters
Parameter
Description
Presentation
Allowed
Default value of Presentation Indicator is set to Presentation Allowed.
Presentation
Restricted
Default value of Presentation Indicator is set to Presentation Restricted.
Not Available Default value of Presentation Indicator is set to Not Available.

31.
Select the default value to insert in the "Screening Indicator" parameter of the "Calling Party
Number" IE when "Screening Indicator" is not already defined in the call properties in the Default SI
for Calling Party Number IE drop-down menu.
Table 181: Default SI for Calling Party Number IE Parameters
Parameter
Description
User Provided Not Screened Default value of Screening Indicator is set to User Provided Not
Screened.

Passed

Verified And Passed.

Failed

Verified And Failed.
Network Provided
Default value of Screening Indicator is set to Network Provided.
Context Dependent
Screening Indicator is set to the value that makes the most sense
according to run-time context.
205
32.
BRI Configuration
Select the default value to insert in the "Type of Number" parameter of the "Called Party Number"
IE when "Type of Number" is not already defined in the call properties in the Default TON for Called
Table 182: Default TON for Called Party Number IE Parameters
Parameter
Unknown
Description
33.
National
Network
Specific
Subscriber
Abbreviated
Select the default value to insert in the "Numbering Plan Identification" parameter of the "Called
in the Default NPI for Called Party Number IE drop-down menu.
Table 183: Default NPI for Called Party Number IE Parameters
Parameter
34.
Description
Unknown
ISDN
Telephony
Data
Telex
National
Standard
Private
Select the unit's behaviour when it receives a Notification Indicator IE with the description set to
User suspended in the Notification User Suspended drop-down menu.
Table 184: Notification User Suspended Parameters
Parameter
206
Description
Ignore
The Mediatrix unit ignores the Notification Indicator IE with description set to
User suspended.
Disconnect
The Mediatrix unit disconnects the call on Notification Indicator IE with

description set to User suspended.
BRI Configuration
35.
Bypass Feature (Mediatrix 3404/3408/3734/3741/3742 Models)

In the event of a power or network failure, the bypass feature permits users to make and receive calls even
when the Mediatrix unit is not operating. The Mediatrix unit BRI 3 and BRI 4 ports may either act as a SCN
bypass. For instance, if you decide to connect a SCN line into the BRI 4 port, you can use a BRI telephone
connected into the BRI 3 port to make calls.
Furthermore:
The port on which the SCN line is connected must be configured as a TE.
The other port must be configured as a NT.
Refer to BRI Configuration on page 197 for more details on how to configure the line type.
During normal operation, the direct connection between the BRI 3 and BRI 4 ports is switched out through
commuting relays and both ports resume normal functions. When power is removed from the Mediatrix unit,
the relay setting is restored to a connected state and the SCN line can be used as an emergency line.
Consequently, a BRI telephone used on the other port is directly connected to this SCN line. When the power
is restored, this automatically removes the Bypass connection; this means that any ongoing call on the Bypass
connection is terminated.
Note: If you are using a crossover Ethernet cable to connect the SCN line to the Mediatrix unit and there is
a power failure, the bypass feature does not work properly.
Bypass Feature (Mediatrix 4402plus / 4404plus Models)

In the event of a power or network failure, the optional bypass feature permits users to make and receive calls
even when the Mediatrix unit is not operating. The Mediatrix unit BRI 1 and BRI 2 ports may either act as a
SCN bypass. For instance, if you decide to connect a SCN line into the BRI 2 port, you can use a BRI
telephone connected into the BRI 1 port to make calls.
Furthermore:
The port on which the SCN line is connected must be configured as a TE.
The other port must be configured as a NT.
Refer to BRI Configuration on page 197 for more details on how to configure the line type.
During normal operation, the direct connection between the BRI 1 and BRI 2 ports is switched out through
commuting relays and both ports resume normal functions. When power is removed from the Mediatrix unit,
the relay setting is restored to a connected state and the SCN line can be used as an emergency line.
Consequently, a BRI telephone used on the other port is directly connected to this SCN line. When the power
is restored, this automatically removes the Bypass connection; this means that any ongoing call on the Bypass
connection is terminated.
207
Interop Parameters Configuration

The interop parameters allow the Mediatrix unit to properly work, communicate, or connect with specific ISDN
devices.
To set the interop parameters:

1.
In the web interface, click the ISDN link, then the Interop sub-link.
Figure 81: ISDN Interop Web Page
2
3
4
5
7
9
11
6
8
10
12
13
2.
The Mediatrix 3734/3741/3742 models have 5 interfaces.
The number of interfaces available vary depending on the Mediatrix 4400 model you have.
3.
Define the behaviour of the Progress Indicator In Setup drop-down menu.

This menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the
SETUP message when acting as the originating side.
See the Send Progress Indicator IE parameter for other conditions for sending Progress Indicator
IE (PRI Configuration on page 186 and BRI Configuration on page 197).
4.
Define the behaviour of the Progress Indicator In Setup Ack drop-down menu.
SETUP ACK when acting as the terminating side.
5.
Define the behaviour of the Progress Indicator In Call Proceeding drop-down menu.
Thie menu defines whether or not the Progress Indicator Information Element (IE) is allowed in the
CALL PROCEEDING message in response to a SETUP message when acting as the terminating
side.
208
6.
Define the behaviour of the Progress Indicator In Progress drop-down menu.

PROGRESS message in response to a SETUP message when acting as the terminating side.
7.
Define the behaviour of the Progress Indicator In Alerting drop-down menu.

ALERTING message.
8.
Define the behaviour of the Progress Indicator In Connect drop-down menu.

CONNECT message.
9.
Define a value, in milliseconds (ms), in the Maximum Facility Waiting Delay (ms) field.
This value defines the maximum amount of time to wait for a FACILITY message, after receiving a
SETUP message, before going on with normal call processing.
After receiving a SETUP message, the system waits for this amount of time for a FACILITY
message. As soon as it receives a FACILITY message or the delay expires, it goes on with normal
call processing.
A FACILITY message can contain useful information for the call. For example, it can contain a
Calling Name.
You must enable the Supplementary Services to use the delay. See PRI Configuration on
page 186 or BRI Configuration on page 197 for more details.
Setting the value to 0 deactivates this waiting delay.
10.
Define whether or not a message with progress indicator No. 1 MUST be considered as offering
inband information available in the Use Implicit Inband Info drop-down menu.
If so, the network must activate the B-channel connection.
The progress indicator No. 1 means that the call is not end-to-end ISDN; further call progress
information may be available inband.
11.
Defines the maximum time, in milliseconds, to wait after receiving a SETUP message before
sending a CALL PROCEEDING message and going on with normal call processing in the Call
Proceeding Delay field.
After receiving a SETUP message, the system waits for a message from the called party. If the
message maps to a User Busy cause, a DISCONNECT message is sent instead of the CALL
PROCEEDING otherwise it goes on with normal call processing.
The value 0 deactivates this feature.
12.
Define how the Calling Name is delivered in the Calling Name Delivery field.
The Calling Party Name can be received and sent through three different methods: Facility
information element, Display information element or User-User information element.
Table 185: Calling Name Delivery Parameters
Parameter
Description
DisplayIe
Use a Display Information Element for delivering the Calling Name.
FacilityIe
Use a Facility Information Element for delivering the Calling Name.
209
Table 185: Calling Name Delivery Parameters (Continued)

Parameter
Description
UserUserI
e
Use a User-User Information Element for delivering the Calling Name.
SignalingP
rotocol
Use the delivery method defined by the signaling protocol.
When receiving an incoming call, the three possible sources of Calling Party Name are checked in
the following order: User-User, Display and Facility. The last found is used.
The Calling Party Name is accepted in a Display information element only when explicitly identified
as a Calling Party Name (i.e. only when "Display Type" = "Calling Party Name" in the information
element).
When initiating a call, the Calling Party Name is sent according to the method selected above. If the
method selected is not supported for the protocol in use, the default method for this protocol is used.
The following table shows which method is used vs. the configuration of CallingNameDelivery:
Table 186: Calling Name Delivery Method vs. Configuration
Calling Name Delivery
Protocol
eFacility
eDisplay
eUserUser
eSignalingProtocol
DSS1
IE User-User
IE User-User
IE User-User
IE User-User
Dms100
IE Facility
IE Display
IE Display
IE Display
NI-2
IE Facility
IE Facility
IE Facility
IE Facility
5ESS
IE Facility
IE Facility
IE User-User
IE Facility
QSIG
IE Facility
IE Facility
IE Facility
IE Facility
See PRI Configuration on page 186 and BRI Configuration on page 197 for more details on
signaling protocols.
13.
Define whether or not an ISDN message with a TEI broadcast needs to be interpreted as a TEI 0
when the connection type is 'PointToPoint' in the Allow TEI Broadcast in PTP drop-down menu.
See BRI Configuration on page 197 for more details on the connection type.
This parameter is available for the following BRI models:
14.
Mediatrix 3404
Mediatrix 3408
Mediatrix 3734
Mediatrix 3741
Mediatrix 3742
Play Local Ringback when no Media Stream


by using the CLI
You can force the local ringback generation when early-media is enabled but no media stream has been
received yet. This variable only affects incoming calls on the ISDN interface.
210
Note that this variable only applies to 180 SIP responses when early-media is enabled.
The following configurations are supported:
Table 187: Play Local Ringback Configuration
Configuration
Description
disable
Do not play local ringback when doing early-media.
enable
The local ringback is played after sending an ALERTING and no media stream has been
received yet from the outgoing interface.
To set how to play the local ringback when there is no media stream:
1.
In the isdnMIB, set the Play Local Ringback configuration in the

InteropPlayLocalRingbackWhenNoMediaStream variable.
isdn.InteropPlayLocalRingbackWhenNoMediaStream="Value"

Table 188: Play Local Ringback Values
Value Meaning
disable
enable
211
ISDN Timers Configuration
ISDN Timers Configuration

This section allows you to set timer parameters.
To set the ISDN timers:

1.
In the web interface, click the ISDN link, then the Timer sub-link.
Figure 82: ISDN Timer Web Page
2
3
4
2.
3.
Set the Auto Cancel Timeout field with the time, in seconds, the endpoint rings before the call is
Setting this variable to 0 disables the timeout. Calls will not be automatically cancelled and will ring
until the party answers.
4.
Set the value, in milliseconds (ms), of the Layer 1 Timer T3 in the L1 Timer T3 field.
Timer 3 (T3) is a supervisory timer that has to take into account the overall time to activate. This
time includes the time it takes to activate both the TE-NT and the NT-TE portion of the customer
access.
The expiry of Timer T3 is intended to provide an indication that the network side cannot complete
the activation procedure, probably due to a failure condition or the terminal cannot detect INFO 4.
5.
212
Services Configuration
This section allows you to set the ISDN optional services.
Standards Supported
ETS 300 207: Call Diversion and Call Rerouting
To set the ISDN optional services:

1.
In the web interface, click the ISDN link, then the Services sub-link.
Figure 83: ISDN Services Web Page
2
3
4
5
7
9
6
8
10
11
13
12
14
15
16
2
2.
3.
Select whether or not supplementary services FACILITY messages and FACILITY information
elements should be enabled on the ISDN interface in the Facility Services drop-down menu.
This controls how to use the FACILITY information element.
Table 189: Facility Services
Parameter
Disable
Description
Facility information elements are disabled in both directions (send and receive). No
Facility information element can be inserted in sent messages. When receiving a
FACILITY message containing supplementary services information, the Mediatrix
unit replies with a STATUS message saying the FACILITY is not supported.
213
Table 189: Facility Services (Continued)

Parameter
Enable
Description
Facility information elements are enabled in both directions (send and receive).
Facility information elements can be inserted in sent messages.
When receiving a FACILITY message containing supplementary services
information, the Mediatrix unit accepts and interprets the message, processing
supported supplementary service messages and silently discarding unsupported
supplementary service messages.
Generic procedures for the control of ISDN supplementary services are defined in the
recommendation ITU-T Q.932.
You can define a waiting delay as described in the section Interop Parameters Configuration on
page 208.
Note: To activate the ISDN hold feature, you must also set the Default Hook Flash Processing feature to
Use Signaling Protocol in General Configuration on page 421.
4.
Select whether or not to enable the CLIP service in the Calling Line Information Presentation dropdown menu.
The Calling Line Information Presentation (CLIP) is an optional service that can be offered by the
ISDN provider. CLIP is offered to the called party to see the calling party's ISDN number. CLIP is
complemented by privacy rules controlled by the Calling Line Information Restriction and Calling
Line Information Restriction Override parameters.
The Calling Line Information Presentation parameter has the following effect:
Parameter
5.
Description
UserOnly
Sends a Calling Number IE that contains the calling number digits when acting as
the User UNI-side (Endpoint Type drop-down menu set to TE) otherwise the
Calling Number IE is not sent.
Enable
Sends a Calling Number IE that contains the calling number digits.
Disable
The Calling Number IE is never sent.
Select whether or not to enable the CLIR service in the Calling Line Information Restriction dropdown menu.
The Calling Line Information Restriction (CLIR) service is offered to the calling party to restrict
presentation of the calling party's ISDN number to the called party.
Setting this parameter to Disable disables the CLIR service.
Setting this parameter to Enable enables the CLIR service. This has the following effects:
For all ISDN signaling protocols except QSIG:
On a TE interface (Endpoint Type drop-down menu set to TE) at the originating network
side, when sending a SETUP message with a Calling Party Number (CPN) IE, the
Presentation Indicator (PI) is set to "Restricted". The calling party number itself is
included in the CPN IE if available.
On a NT interface (Endpoint Type drop-down menu set to NT) at the originating

network side, when receiving a SETUP message with a CPN IE, the PI is set to
"Restricted". The calling party number itself is forwarded.
For the QSIG signaling protocol (PRI/BRI interface Signalling Protocol drop-down is set to QSIG):
214
Sending a SETUP message: The PI is set to "Restricted" in the CPN IE inserted in the
SETUP message sent to the ISDN, unless the CLIR override option is set. However,
even if PI is set to "Restricted", the calling number is included in the CPN IE.
Receiving a SETUP message: If PI is set in the received message, the calling party
number is removed, unless the CLIR override option is set.
See PRI Configuration on page 186 or BRI Configuration on page 197 for more details.
6.
Select whether or not to enable the CLIR override option in the Calling Line Information Restriction
Override drop-down menu.
This option allows the calling party number to be presented to the destination party even when the
Calling Party Number (CPN) IE's Presentation Indicator (PI) is set to "Restricted". This option is
typically used for police or emergency services.
Setting this variable to Disable disables the CLIR Override option.
Setting this variable to Enable enables the CLIR Override option. This has the following effects:
For all ISDN signaling protocols except QSIG:
The override option acts on the NT interface of the destination network side. It prevents
the number to be removed from the CPN IE inserted in the SETUP message sent to
the destination TE.
For the QSIG signaling protocol (PRI/BRI interface Signalling Protocol drop-down is set to QSIG):
The override option prevents the calling name to be removed from the CPN IE in a
received SETUP message.
7.
Select whether or not to send a Connected Number IE within the CONNECT message at the
originating ISDN side in the Connected Line Identification Presentation drop-down menu.
The Connected Line Identification Presentation (COLP) is an optional service offered at the
originating interface by the NT to the TE.
Table 190: COLP Parameters
Parameter
8.
Description
Enable
Sends a Connected Number IE within the CONNECT message, which contains the
connected number digits once the transformation of the routing table has been
applied.
Disable
The Connected Number IE is never sent.
Select whether or not to set the Connected Number Information Element restriction at the
destination ISDN side in the Connected Line Identification Restriction drop-down menu.
The Connected Line Identification Restriction (COLR) is a service offered to the TE at the
destination interface.
Table 191: COLR Parameters
Parameter
Description
Enable
When activated at the User UNI-side (Endpoint Type drop-down menu set to TE),
marks the Connected Number IE with a 'restricted' Presentation Indicator, which
keeps privacy over the connected number digits. This option has no effect when
activated at the Network UNI-side (Endpoint Type drop-down menu set to NT).
Disable
No restriction is applied.
215
9.
Select whether or not to set the Connected Number Information Element restriction override at the
originating ISDN side in the Connected Line Identification Restriction Override drop-down menu.
Table 192: COLR Override Parameters
Parameter
10.
Description
Enable
When activated at the Network UNI-side (Endpoint Type drop-down menu set to
NT), the connected number digits are delivered even if the Presentation Indicator is
set to 'restricted'. This option has no effect when activated at the User UNI-side
(Endpoint Type drop-down menu set to TE). This is a national option designed for
emergency services.
Disable
No restriction override is applied.
Define whether or not NOTIFY messages can be sent in the Outgoing Notify drop-down menu.
Table 193: Outgoing Notify Parameters
Parameter
Description
Enable
NOTIFY messages can be sent.
Disable
NOTIFY messages are never sent.
The following NOTIFY messages are supported:
11.
REMOTE HOLD: Sent when the remote peer holds the call.
REMOTE RETRIEVAL: Sent when the remote peer retrieves the call.
Set the Maintenance Service Call Termination drop-down menu with the call termination strategy
after reception of a service message requesting a maintenance on the associated bearer channel.
Table 194: Maintenance Service Parameters
Parameter
12.
Description
Graceful
The call proceeds normally until the user clears the call. The associated bearer is
then set to maintenance. This is the default value.
Abrupt
The call is terminated immediately and set to maintenance.
Define whether or not the optional Date/Time Information Element (IE) can be included in the
CONNECT and SETUP messages in the BRI and PRI.
Table 195: Date/Time IE Support Parameters
Parameter
Description
Disable
Date/Time is not sent
Local Time
Date/Time IE is sent, containing the local time according to the configured time zone in the Network
Host Time Configuration.
UTC
Date/Time IE is sent, containing the Coordinated Universal Time (UTC)
Note: Without a SNTP Synchronized connection, the Date/Time IE is not sent. See SNTP Configuration
on page 95 for more details on how to configure the SNTP.
216
13.
Define the AOC-E Support drop-down menu how to send the total charge at the (E)nd of the call in
AOC-E messages.
Table 196: AOC-E Support Parameters
Parameter
no
Description
The AOC-E support is disabled. No information is forwarded to the peer interface.
transparent On an NT interface, the information is sent as received from the network. No

information is sent if the network does not provide information.
On a TE interface, the information is forwarded to the peer interface if AOC
messages are received from the network.
automatic
On an NT interface, always send the information. If the network does not provide
information, 'noChargeAvailable' is sent.
explicit
On an NT interface, always send the information if the phone requests AOC on a

per-call basis. 'noChargeAvailable' is sent if the network does not provide
information. If the phone does not request AOC on a per-call basis, no information
is sent.
On a TE interface, send an AOC request to the network. If the network rejects the
request, no information is forwarded to the peer interface. Otherwise, the
information is forwarded to the peer interface if AOC messages are received from
the network.
14.
Define the AOC-D Support drop-down menu how to send the current charge (D)uring the call in
AOC-D messages.
Table 197: AOC-D Support Parameters
Parameter
no
Description
The AOC-D support is disabled. No information is forwarded to the peer interface.
transparent On an NT interface, the information is sent as received from the network. No

information is sent if the network does not provide information.
automatic
On an NT interface, always send the information. If the network does not provide
information, 'noChargeAvailable' is sent.
explicit
On an NT interface, always send the information if the phone requests AOC on a

per-call basis. 'noChargeAvailable' is sent if the network does not provide
information. If the phone does not request AOC on a per-call basis, no information
is sent.
On a TE interface, send an AOC request to the network. If the network rejects the
request, no information is forwarded to the peer interface. Otherwise, the
information is forwarded to the peer interface if AOC messages are received from
the network.
Note: The AOC features are not available in the NI2 and QSIG signalling protocols. See PRI Configuration
on page 186 for more details on how to configure the signalling protocol.
217
Note: To enable AOC support on the ISDN interface, you must enable the FACILITY services and at least
one of the following AOC support: AOC-E (End of Call) or AOC-D (During the Call).
Since the AOC from ISDN interface to SIP is currently not supported, enabling the AOC on an ISDN interface
configured as TE (user side) is only meaningful when using hairpinning.
15.
Set the Call Rerouting Behavior drop-down menu with how the call rerouting request received from
the private network side is supported.
The Call Rerouting supplementary service allows to reroute an incoming public ISDN call
(originated from the PSTN) within or beyond the private ISDN network (such a PBX) as specified in
the ETS 300 207 01, section 10.5. The Rerouting data are received and relayed through a Facility
message containing a Facility Information Element.
Rerouting requests are received in a Facility IE.
Table 198: Forward Call Rerouting Parameters
Parameter
Description
Unsupported Rerouting requests received are rejected. A reject answer is sent to the private
network.
Relay
Reroute
Rerouting requests are relayed as received to the public network side. If the peer
rejects or does not support the reroute request, the ISDN service may initiate a
new call to process the rerouting request locally.
Process
Locally
Received Rerouting requests are not relayed to the public network side. The
ISDN service attempts to connect to the rerouted address by initiating a new call.
Note: The Call Rerouting feature is not available in the NI2 and QSIG signalling protocols. See PRI
Configuration on page 186 for more details on how to configure the signalling protocol.
16.
Enter one or more numbesr in the MSN field.

You can enter a comma-separated list of numbers. The comma-separated list must use the
following syntax:
777, 888, 999, 555, 444.
This enables the Multiple Subscriber Numbers (MSN) supplementary service with these numbers.
A MSN is a telephone number associated with a line.
The MSN supplementary service enables each individual terminal on one access to have one or
more identities.
If the Called E.164 received from a call does not match any MSN numbers, the call is silently
discarded.
This supplementary service applies only on a BRI Interface configured in TE Point to Multi-Point.
See BRI Configuration on page 197 for more details.
17.
218
IP Sync Configuration
Note: This feature is available on the following BRI models:
Mediatrix 3404
Mediatrix 3408
Mediatrix 3734
Mediatrix 3741
Mediatrix 3742
Mediatrix 4401 / 4401plus
This section allows you to configure the Mediatrix unit in order to control the IP media synchronization using
clock reference signals sent over IP. It allows you to to synchronize multiple PRI or BRI units over a network
or to be synchronized by a remote Mediatrix unit.
Note: The Mediatrix units IP Sync settings do not support IPv6. See IPv4 vs. IPv6 Availability on page 87
for more details.
IP Sync uses two types of units:
Master unit. This unit sends Sync packets to all registered slaves.
Slave units. These units are registered to a master unit.
The Clock Syncnronization section displays the current status of the unit in regard to the Clock
Synchronization protocol.
Table 199: IP Sync Status
Parameter
Unit
Description
Inactive: Clock synchronization is not in use.
Master: The unit is a Master.
Slave: The unit is a Slave.
Precision
Estimate of the drift in PPB (parts per billion), of the local TDM clock with regards to the
reference (master) clock. A value of 10 000 means the local clock may be off by +/- 0,00001%
which is +/- 0.08 Hz since the frame sync frequency on the TDM bus is 8000Hz. This may
result in overruns or underruns on the TDM bus causing an average bit error ratio up to
0,00001. When the unit is not currently synchronized, the value is the hardware clock
precision. When the unit is a Master, the value in this variable should not be considered.
Buffer
On a Slave unit, displays the maximum jitter observed from incoming synchronization packets
over a five minutes period. When the unit is a Master, the value in this variable should not be
considered.
Enabling/Disabling IP Synchronization
This section describes how to enable or disable the IP synchronization feature.
Mediatrix 3xxx models: IP Synchronization is enabled when at least one hardware card listed in the BRI
Cards Status or PRI Cards Status section of the System > Hardware page has its Clock Reference value set
to None (see BRI Hardware Configuration on page 61 or PRI Hardware Configuration on page 62 for more
details). If two cards of a same unit have their Clock Reference values set to None, then the card located in
219
the lowest Slot ID is synchronized and you must configure the other card so that it synchronizes on the first
card by setting the corresponding Clock Reference parameter of the BRI Cards Configuration or PRI Cards
Configuration section to Other Card.
Mediatrix 44xx models: IP Synchronization is enabled when the Clock Reference value listed in the Unit
Status section of the System > Hardware is set to None (see BRI Hardware Configuration on page 61 for
more details).
If the IpSync service is stopped or if no Clock Reference value set to None, IP Synchronization is disabled.
Master Units Configuration

Master units send Sync packets to all registered slaves. These units are available only through a special
firmware profile.
To set the IP Sync parameters of a Master unit:

1.
In the web interface, click the ISDN link, then the IP Sync sub-link.
Figure 84: ISDN IP Sync Web Page, Master Unit
2.
In the Clock Synchronization Configuration section, set the Port field to the port number with which
the unit sends and receives clock synchronization packets.
This port is used on all network interfaces.
3.
Slave Units Configuration

Slave units are registered to a master unit. Any of the supported models may be a slave unit.
To set the IP Sync parameters of a Slave unit:

1.
In the web interface, click the ISDN link, then the IP Sync sub-link.
Figure 85: ISDN IP Sync Web Page, Slave Unit
2
4
2.
220
3.
Set the Static Master Host field with the FQDN or IP address of the Master unit on which the Slave
attempts to synchronize.
If the port is omitted or set to 0, the default port 5003 is used.
4.
Set the Static Redundant Master Host field with the FQDN or IP address of the redundant Master
unit on which a Slave attempts to synchronize when the Static Master Host is unreachable.
When the port is omitted, the default port 5003 is used.
5.
Resetting the Synchronization Data

You can erase all synchronization data that are currently saved. This desynchronizes the unit. You must
perform this operation in the Mediatrix units CLI. See Chapter 2 - Command Line Interface (CLI) on page 19
for more details.
To reset the synchronization data:

1.
In the CLI of the Mediatrix unit, type the following command:

IpSync.ResetSyncData
2.
You can also use the following two commands:

cd IpSync
ResetSyncData
221
222
R2 CAS Parameters
H A P T E R
22
R2 CAS Configuration
This chapter describes how to configure the R2 CAS parameters of the Mediatrix unit.
Standards Supported


ITU-T Recommendation Q.421: Digital line signalling code
ITU-T Recommendation Q.422: Digital Line signalling clauses for exchange line signalling equipment
ITU-T Recommendation Q.441: Signalling code
Note: This chapter applies only to the Mediatrix 3621, Mediatrix 3631, and Mediatrix 3632 models.
Introduction
CAS stands for Channel Associated Signaling. With this method of signalling, each traffic channel has a
dedicated signaling channel. In other words, the signalling for a particular traffic circuit is permanently
associated with that circuit. Channel-associated call-control is still widely used today, mostly in South America,
Africa, Australia, and in Europe.
The Mediatrix unit uses the MFC/R2 CAS protocol. This is a compelled sequence multi-frequency code
signaling. MFC/R2 can be used on international as well as national connections.
In MFC/R2 signaling, the equipment units at the exchanges that send and receive digits, and the signaling
between these units, are usually referred to as register and interregister signalling.
The terms forwards and backwards are heavily used in descriptions of MFC/R2. Forwards is the direction from
the calling party to the called party. Backwards is the direction from the called party to the calling party.
You can configure Mediatrix unit parameters for the E1 R2 CAS.
Line Signals for the Digital Version of MFC/R2

The MFC/R2 digital line signals (defined in ITU-T Q.421) are the ABCD bits of CAS in timeslot 16 of an E1.
They represent the states of the line, and are similar to the states of an analog line. In general, only bits A and
B are used. In most systems, bits C and D are set to fixed values and never change. There are some national
variants where bit C or D may be used for metering pulses.
Interregister Signals
The interregister, or interswitch, signals in MFC/R2 signaling (defined in ITU-T Q.441) are encoded as the
presence of 2, and only 2, out of 6 specific tones, spaced at 120 Hz intervals. Two sets of tones are defined
one for forward signals, and one for backward signals. There are 15 combinations of 2 out of 6 tones, so there
are 10 signals for the digits 0 to 9, and 5 additional signals available for supervisory purposes.
MFC/R2 uses a separate set of frequencies for the forward and backwards directions.
The interregister signals are sent in-band. They may pass transparently through several nodes in the network
between the two terminating switches. The signals are arranged in groups. When a call begins, the calling end
uses group I signals, and the called end uses group A. The called end may tell the calling end to switch to
using group II and group B signals, or to switch back to group A. In some countries, there are also groups III
and C, used for caller number transfer. Groups III and C do not exist in the ITU specifications.
223
Chapter 22 - R2 CAS Configuration
Selecting the R2 Signaling Protocol
MFC/R2 uses a system called compelled signaling. To ensure the sending end never sends signals too fast,
each signal from the sending end results in an acknowledgement from the receiving end. The sending end is
instructed signal by signal what it should send next a dialed digit, a digit of caller ID, etc.
Selecting the R2 Signaling Protocol

You must set the unit to use the R2 signaling protocol. You can do so in the System / Hardware page. The
Hardware page differs depending on the product and model you have.

1.
2.
If you want to configure the clock reference of a specific interface, you must set the Clock Mode
drop-down menu to Master. See R2 Channel Associated Signaling on page 226 for more details.
Reference
Description
None
Other
Card
The internal clock synchronizes with the other R2 interface of the Mediatrix unit.
This interface must be configured in Slave mode (Clock Mode drop-down menu of
the R2 Channel Associated Signaling section) to provide the clock reference to the
other interfaces.
Note: This choice is not available on the Mediatrix 3621 and 3631 models.
3.
Currently, R2 works only on the E1 line type.
Note: Before version 1.1r8.76, the Isdn service needed to be restarted when modifying the Line Type. Since
version 1.1r8.76, the Line Type variable has been moved to the Ex1Pri_1 service and now you rather need
to restart the unit instead of restarting the service when the Line Type is modified.
4.
Select the R2 signaling in the Signaling drop down menu.

instance, if you are in ISDN with a route isdn-Slot2/E1T1, then change to R2, you must change the
route to r2-Slot2/E1T1.
5.
224
R2 Auto-Configuration
R2 Auto-Configuration
The R2 Auto-configuration feature allows you to detect and to configure all R2 interfaces so that the R2 link
goes up and becomes usable with a minimal user interaction. When launching an auto-configuration process,
it stops automatically when all interfaces have been tested. For each interface, the auto-configuration process
is considered successful when the link becomes up or a failure when all combinations have been tried without
having a link up.
Caution: Launching the auto-configuration may terminate abruptly all ongoing R2 calls.
Note: Auto-configuration on all R2 interfaces may take some time to complete. Some of the current R2

1.
In the web interface, click the R2 link, then the Status sub-link.
Figure 87: R2 Status Configuration Section
2.

The process starts.
Preset
The R2 Preset Configuration section allows you to load a set of preset configuration for your R2 connections.
These preset files are located in the file system's persistent memory. They differ depending on the Mediatrix
unit you are using. Depending on your unit's profile, it may be possible that no preset files are available.
You can also export your current R2 configuration in a preset. Please note that these user-defined presets are
not kept in the event of a partial or factory reset.
225
R2 Channel Associated Signaling

1.
In the R2 Status tab, R2 Preset Configuration section, select one of the available preset files in the
Local Preset drop-down menu.
1
2.
Click Apply.
To export the current R2 configuration as a preset:

1.
In the R2 Preset Configuration section, type a name for the preset in the Preset Name field.
2.
Click Save.
The current R2 configuration is exported. Please note that these user-defined presets are not kept
in the event of a partial or factory reset.
Partial Reset

The R2 Channel Associated Signaling section allows you to define the general parameters related to R2.
226
To configure the R2 CAS parameters:

1.
In the web interface, click the R2 link, then the R2 Channel Associated Signaling Config sub-link.
Figure 90: R2 Channel Associated Signaling Web Page
2
3
4
5
6
7
9
10
11
12
13
14
15
2.
16
The number of interfaces available vary depending on the Mediatrix unit model you have.
3.
Table 200: R2 Interface Clock Mode
Mode
Description

Slave
The clock source can be selected from the Clock Reference drop-down menu (see Selecting the
R2 Signaling Protocol on page 224 for more details). The clock mode could be used, for instance,
to synchronize several units in master clock mode via an E1 line.
4.

Mode
5.
Description
Auto
The pinout is set according to the Clock Mode parameter setting (Step 3).
Te
Forces the pinout to TE regardless of the Clock Mode value.
Nt
Forces the pinout to NT regardless of the Clock Mode value.
Set the Monitor Link State drop-down menu with the physical link state of the R2 interface.
Parameter
Enable
Description
The R2 endpoint's operational state is affected by its interface physical link state.
When the link state of the R2 interface is down, the operational state of its
matching endpoint becomes "disable".
227
Table 202: Interface Link State (Continued)

Parameter
Disable
Description
The R2 endpoint's operational state is not affected by its interface physical link
state.
6.
Coding
Description
B8ZS
Bipolar with 8-Zeros Substitution (T1 lines). Currently not available.
HDB3
AMI
7.

their frames.
Format
Description
SF(D4)
Super frame. Sometimes known as D4 (T1 lines). Currently not available.
ESF
Extended super frame (T1 lines). Currently not available.
CRC4

8.
9.
menu.
Allocation
228
Description
Ascending

Descending

Table 205: Channel Allocation Strategy (Continued)

Allocation
RoundRobinAscending
Description
10.
11.
Set the Encoding Scheme drop-down menu with the voice encoding scheme in the bearer
capabilities.
This encoding scheme is used when initiating a call on the R2 side. The supported encoding
12.
Select the protocol to use for the line signaling in the Line Signaling Protocol drop-down menu.
This signaling must match the connected equipment or network. The Mediatrix unit currently
supports only the Q421-2BitsSignaling signaling, which is the R2 line signaling type ITU-U Q.421.
It is typically used for PCM systems.
13.
Select the R2 incoming digit signaling method in the Incoming Digit Signaling drop-down menu.
Digit signaling is also known as Address Signaling, selection signals and register signaling. The
digits are used primarily to indicate the called number, but can also have other meanings.
Table 206: Incoming Digit Signaling Parameters
Allocation
14.
Description
MfcR2
Multi Frequency Compelled - R2.
DtmfR2
Dual Tone Multi Frequency - R2.
Select the R2 outgoing digit signaling method in the Outgoing Digit Signaling drop-down menu.
Digit signaling is also known as Address Signaling, selection signals and register signaling. The
digits are used primarily to indicate the called number, but can also have other meanings.
Table 207: Outgoing Digit Signaling Parameters
Allocation
15.
Description
MfcR2
Multi Frequency Compelled - R2.
DtmfR2
Dual Tone Multi Frequency - R2.
Select the country in the Country Selection drop-down menu.

You have the following choices:
BrazilR2
MexicoR2
229
16.
R2 Signaling Variants
ArgentinaR2
SaudiArabiaR2
VenezuelaR2
PhilipinesR2
ITU-TR2
Set the Digit Attenuation field with the additional attenuation, in dB, for MFR2/DTMF digits
generation.
By default, MFR2/DTMF digits generation power is determined by country selection. This parameter
provides a mean to reduce this power.
This section allows you to decide whether or not you want to override the default R2 signaling parameters.
The Mediatrix unit uses the following default values:
Table 208: R2 Signaling Parameters Default Values
Default Value (ms)
Parameter
Bra.
230
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
Bits CD
ANI Length
0
(Variable
ANI
length)
0
(Variable
ANI
length)
0
(Variable
ANI
length)
0
(Variable
ANI
length)
0
(Variable
ANI
length)
0
(Variable
ANI
length)
0
(Variable
ANI
length)
DNIS Length
0
(Variable
DNIS
length)
0
(Variable
DNIS
length)
0
(Variable
DNIS
length)
0
(Variable
DNIS
length)
0
(Variable
DNIS
length)
0
(Variable
DNIS
length)
0
(Variable
DNIS
length)
ANI Request
Enable
Disable
Enable
Enable
Enable
Enable
Enable
Send ANI request

after nth DNIS Digits
0
(Variable
number
of DNIS
digits)
0
(Variable
number
of DNIS
digits)
0
(Variable
number
of DNIS
digits)
0
(Variable
number
of DNIS
digits)
0
(Variable
number
of DNIS
digits)
0
(Variable
number
of DNIS
digits)
Collect Call Blocked

Enabled
Enable
Disable
Disable
Disable
Disable
Disable
Disable
ANI Category
NatSubsc Nat
riberNoPr Subscrib
er NoPrio
io
Nat
Subscrib
er NoPrio
Nat
Subscrib
er NoPrio
Nat
Subscrib
er NoPrio
Nat
Subscrib
er NoPrio
Nat
Subscrib
er NoPrio
Line Free Category
LineFree
NoCharg
e
Line Free
NoCharg
e
Line Free
Charge
Line Free
Charge
Line Free
Charge
Line Free
Charge
Line Free
Charge
ANI Restricted
Enable
Disable
Disable
Disable
Disable
Disable
Disable
Incoming Decline
Method
Release
Release
Release
Release
Release
Release
Release
Override Default Country Settings

You can override the default R2 signaling parameters. In that case, you will have access to the R2 Signaling
Variants section to define the signaling you want.
To override the R2 signaling default settings:

1.
In the web interface, click the R2 link, then the Signaling sub-link.
Figure 91: R2 Signaling Variants Web Page
2.
3.
Select whether or not you want to override the default setting of R2 signaling parameters in the
Override Default Country Settings drop-down menu.
Table 209: R2 Signaling Override
Allocation
Description
Disable
The interface uses the default country configuration.
Enable
The interface uses the specific country configuration as defined in the

R2 Signaling Variants section. To retrieve the default configuration
associated with the current country, click the Reset to Default button.
Proceed to R2 Signaling Variants on page 231.
Overriding the default settings is considered as advanced configuration. Media5 recommends not
to modify the country variants unless you know exactly what you are doing.
This section allows you to define R2 signaling parameters. You can click the Reset to Default button at any
time to revert back to the default R2 signaling.
To set R2 signaling parameters:

1.
In the R2 Signaling Variants section, set the C and D bits when the device transmits line signals in
the Bits CD field.
The device ignores the C and D bits of received line signals.
231
Figure 92: R2 Signaling Variants Section
1
2
3
4
5
6
7
8
9
10
11
2.
Set the ANI Length field with the length of Automatic Number Identification (ANI) to be requested or
sent.
If a variable-length ANI is used, the End of ANI tone set in R2 Tones Forward Groups on page 244
is sent to indicate the end of the ANI digits. When fixed-length ANI is used and the available ANI
digits are longer than the requested length, the last n digits are sent. If available ANI digits are
shorter, then the End of ANI tone set in R2 Tones Forward Groups on page 244 is sent.
3.
0: Variable ANI length.
1..20: Specific ANI length.
Set the DNIS Length field with the length of the Dialed Number Identification Service (DNIS)
expected.
DNIS is the called party or the destination number. If a variable length is defined, then the I-15 digit
is used to indicate the end of DNIS.
4.
0: Variable DNIS length used.
1..20: Specific DNIS length expected.
Define whether or not ANI should be requested in the ANI Request field.
ANI is the calling party number. When ANI is requested, the calling party category followed by the
actual ANI is sent. If this parameter is enabled, the ANI request is sent after the nth DNIS digit
(defined in Step 5) is received.
5.
Set the Send ANI Request after nth DNIS Digits field with the number of DNIS digits to be received
before sending the ANI request (if the ANI Request field is set to Enable).
If a variable number is used, the ANI request is sent after all DNIS digits have been received.
6.
0: Variable number of DNIS digits.
1..10: Specific number of DNIS digits.
Define whether or not the Collect Call Blocked Option is used in the Collect Call Blocked Enabled
drop-down menu.
Two methods actually exist to do Collect Call Blockage. The first method refers to R2 signaling and
how, through the use of signals, collect calls can be blocked. The second method refers to how,
through the use of double answer, the same end can be achieved.
For an incoming collect call, a signal of Group II-8 is sent forward from the caller to the called party.
The called party implements the collect call blockage (when enabled) by sending backward to the
caller a signal of Group B-7 indicating that the collect calls are not being accepted by the called
party. Consequently, the originator of the call gets a busy tone and the local calling party circuit that
has been used for the call is dropped when the originator puts the phone on hook.
The double answer allows the destination side to reject or accept a collect call (toll). Since the owner
of the collect call is the person being called, the CO recognizes that the call is being dropped just
by the fact that the call was dropped. For regular, non-collect calls, the owner of the call is the
232
person calling and not the party being called. So, if the receiver of the call decides to refuse the call,
a double answer is generated within a specified time. If the receiver wants to answer the call, a
double answer is not generated and the receiver is then billed for the incoming call.
Table 210: Collect Call Blockage Parameters
Parameter
7.
Description
Enable
The signal of Group B-7 is sent if a Group II-8 (Collect Call) signal is
received from the caller or a double answer is generated within a
specified time depending on the value defined in the R2 Timer
Variants table (see R2 Timers Variants on page 236 for more
details).
Disable
No signal is sent in response to a Group II-8 (Collect Call) signal and/

or no double answer is generated by the called side upon incoming
calls.
Set the ANI Category drop-down menu with the group II forward signal to be sent upon receiving a
calling party category request.
This tone indicates the category of the calling party.
Table 211: ANI Category Parameters
Parameter
8.
Description
NatSubscriberNoPrio
The call is set up from a national subscribers line and is non-priority.
NatSubscriberPrio
The call is set up from a national subscribers line to which priority

treatment of calls has been granted.
NatMaintenance
The call comes from a national maintenance equipment.
NatSpare
Spare.
NatOperator
The call is set up from a national operators position.
NatData
The call will be used for national data transmission.
IntSubscriberNoPrio
The call is set up from an international subscribers line and is nonpriority.
IntData
The call will be used for international data transmission.
IntSubscriberPrio
The call is set up from an international subscribers line to which

priority treatment of calls has been granted.
IntOperator
The call is set up from an international operators position.
CollectCall
The call is set up for Call Collect.
Set the Line Free Category drop-down menu with the group B backward signal to be sent by the
incoming R2 register to indicate line free condition of the destination party.
Table 212: Line Free Category Parameters
Parameter
9.
Description
LineFreeNoCharge
The called partys line is free but is not to be charged on answer.
LineFreeCharge
The called partys line is free but is to be charged on answer.
Set the ANI Restricted drop-down menu with the behaviour of the unit following the reception of a
reject request after sending the Send next digit (ANI) request.
233
R2 Timers Variants
The request is generally rejected when the calling party is unable to send its identification.
Table 213: ANI Restricted Parameters
Parameter
10.
Description
Enable
A congestion tone is sent in response to the reject request and the

call MUST be dropped.
Disable
The unit uses the same behaviour as the End of ANI Tone and the
call WILL be completed.
Set the Incoming Decline Method drop down to indicate how to cancel a call attempt from R2 if the
called party rejects the call when in Seizure Acknowledged state (waiting for answer).
Table 214: Incoming Decline Method
Parameter
Description
Release:
B bit is set to 0 ans state is set to Released.
ClearBack
B bit is set to 0 until decline quard expires then B is set to 1

and state is set to Clear-back.
11.
R2 Timers Variants
This section allows you to decide whether or not you want to override the default R2 timers parameters. The
Mediatrix unit uses the following default values:
Table 215: R2 Timers Default Values
Default Value (ms)
Parameter
Bra.
234
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
Seizure Ack Timeout
2000
2000
2000
2000
2000
2000
2000
Fault Seizure Ack

Timeout
60000
60000
60000
60000
60000
60000
60000
Double Seizure
Timeout
100
100
100
100
100
100
100
Double Answer
Timeout
1000
1000
1000
1000
1000
1000
1000
Answer Timeout
ReAnswerTimeout
1000
1000
1000
1000
1000
1000
1000
Release Guard
Timeout
100
100
100
100
100
100
100
InterCall Guard
Timeout
100
100
100
100
100
100
100
Congestion Tone
Guard Timeout
1000
1000
1000
1000
1000
1000
1000
Unblocking Timeout
100
100
100
100
100
100
100
Address Complete
Timeout
8000
8000
8000
8000
8000
8000
8000
R2 Timers Variants
Table 215: R2 Timers Default Values (Continued)

Default Value (ms)
Parameter
Bra.
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
Wait Answer Timeout
60000
60000
60000
60000
60000
60000
60000
Digit Complete
Timeout
4000
4000
4000
4000
4000
4000
4000
Wait GroupB
Response Complete
Timeout
3000
3000
3000
3000
3000
3000
3000
Wait Immediate
Response Complete
Timeout
1000
1000
1000
1000
1000
1000
1000
Play Tone Guard

Timeout
70
70
70
70
70
70
70
Accept Call Timeout
2000
2000
2000
2000
2000
2000
2000
Clear Forward Guard

Timeout
1500
1500
1500
1500
1500
1500
1500
Clear Backward
Guard Timeout
1500
1500
1500
1500
1500
1500
1500
Fault On Answered
Guard Timeout
250
250
250
250
250
250
250
Fault On Clear
Backward Guard
Timeout
250
250
250
250
250
250
250
Fault On Seize Ack

Guard Timeout
250
250
250
250
250
250
250
Fault On Seize Guard

Timeout
250
250
250
250
250
250
250
Decline Guard
Timeout
1500
1500
1500
1500
1500
1500
1500

You can override the default R2 timers. In that case, you will have access to the R2 Timers Variants section
to define the timers you want.
To override the R2 timers default settings:

1.
In the web interface, click the R2 link, then the Timers sub-link.
Figure 93: R2 Timers Variants Web Page
235
2.
R2 Timers Variants
3.
Select whether or not you want to override the default setting of R2 timers in the Override Default
Country Settings drop-down menu.
Table 216: R2 Timers Override
Allocation
Description
Disable
Enable

R2 Timers Variants section. To retrieve the default configuration
Proceed to R2 Timers Variants on page 236.
R2 Timers Variants
This section allows you to define R2 timers. You can click the Reset to Default button at any time to revert back
to the default R2 timers.
To set R2 timers:
1.
In the R2 Timers Variants section, set the Seizure ACK Timeout field with the maximum time, in
milliseconds (ms), an outgoing R2 register waits for the seizure acknowledgement signal after
sending a seizure signal.
Figure 94: R2 Timers Variants Section
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
236
R2 Timers Variants
2.
Set the Fault Seizure Ack Timeout field with the maximum time, in milliseconds (ms), an incoming
R2 register waits for a seizure acknowledge failure condition to clear.
3.
Set the Double Seizure Timeout field with the minimum time, in milliseconds (ms), an outgoing R2
register waits after a double seizure is recognized before releasing the connection.
4.
Set the Double Answer Timeout field with the maximum time, in milliseconds (ms), an outgoing R2
register waits after receiving a clear-backward signal before releasing the connection.
5.
Set the Answer Timeout field with the maximum time, in milliseconds, the answer signal AB=01 is
applied before the clear backward signal AB=11 is sent.
This variable is generally used to reject collect call (toll) and is only available if the
CallCollectBlocked is enabled.
A value of 0 means that the signal is applied until the call is disconnected. However, if a special
event (flash hook) is detected in the answered state, then the clear backward signal AB=11 is
immediately applied for a period corresponding to the ReAnswer Timeout parameter.
6.
Set the ReAnswer Timeout field with the maximum time, in milliseconds, the clear backward signal
AB=11 is applied before the answer signal AB=01 is reapplied again.
This variable is generally used to reject collect call (toll) and is only available if the Collect Call
Blocked Enabled parameter is enabled (see R2 Signaling Variants on page 231 for more details).
The ReAnswerTimeout is only applied if the AnswerTimeout or an event generating the clear
backward signal is triggered.
7.
Set the Release Guard Timeout field with the maximum time, in milliseconds (ms), an incoming R2
register waits before sending an idle line signal when a clear forward line signal is received.
8.
Set the InterCall Guard Timeout field with the maximum time, in milliseconds (ms), an outgoing R2
register waits after receiving an idle line signal before attempting a new seizure of the line.
9.
Set the Congestion Tone Guard Timeout field with the maximum time, in milliseconds (ms), an
incoming R2 register waits after sending a congestion tone before sending a clear forward line
signal and transit to the idle state.
10.
Set the Unblocking Timeout field with the maximum time, in milliseconds (ms), a both-way trunk
waits before assuming an idle state when a blocking condition is removed.
This will prevent a too aggressive seizure of the trunk.
11.
Set the Address Complete Timeout field with the maximum time, in milliseconds (ms), that the caller
waits for the reception of an Address Complete Tone after sending all ANI or DNIS digits.
12.
Set the Wait Answer Timeout field with the maximal time, in milliseconds (ms), that the caller waits
for an Answer signal (ANSW) after receiving a Group B Line Free Signal Tone.
This timer is effective only when the line is free.
13.
Set the Digit Complete Timeout field with the maximal time, in milliseconds (ms), that the caller waits
for a Group I forward tone after sending either a Group A next DNIS digit, next ANI digit, or next
calling category Tone.
14.
Set the Wait GroupB Response Complete Timeout field with the maximal time, in milliseconds (ms),
that the caller waits for the confirmation of the end of the compelled sequence after receiving a
Group B Signal Tone.
The end of the compelled sequence is detected by a transition of the backward tone to off.
15.
Set the Wait Immediate Response Complete Timeout field with the maximal time, in milliseconds
(ms), that the caller waits for the confirmation of the end of the compelled sequence after receiving
a Group B Signal Tone.
The end of the compelled sequence is detected by a transition of the backward tone to off. This timer
is specific to the immediate accept Signal Tone.
237
R2 Digit Timers Variants
16.
Set the Play Tone Guard Timeout field with the maximum time, in milliseconds (ms), an incoming
R2 register waits after receiving the confirmation of the reception of the Group B Signal Tone before
playing one of the calling tones in the caller direction.
17.
Set the Accept Call Timeout field with the time, in milliseconds (ms), that the unit waits to accept a
R2 CAS call.
18.
Set the Clear Forward Guard Timeout field with the maximum time, in milliseconds (ms), an
outgoing R2 register waits after sending a clear forward line signal before transiting to the idle state.
19.
Set the Clear Backward Guard Timeout field with the maximum time, in milliseconds (ms), an
incoming R2 register waits after sending a clear backward line signal before sending the idle line
signal and transit in the idle state.
20.
Set the Fault On Answered Guard Timeout field with the maximum time, in milliseconds (ms), an
outgoing R2 register waits for the fault to clear before sending the clear forward line signal.
In the case bb = 0 is recognized while in the answered state, no immediate action is taken. However,
the clear forward signal is sent if bb = 1 is restored or the answered guard timeout is reached.
21.
Set the Fault On Clear Backward Guard Timeout field with the maximum time, in milliseconds (ms),
an outgoing R2 register waits for the fault to clear before sending the clear forward line signal.
In the case bb = 0 is recognized while in the clear backward state, no immediate action is taken.
However, the clear forward signal is sent if bb = 1 is restored or the clear backward guard timeout
is reached.
22.
Set the Fault On Seize Ack Guard Timeout field with the maximum time, in milliseconds (ms), an
In the case bb = 0 is recognized while in the seize acknowledge state prior to the answer signal, no
immediate action is taken. However, the clear forward signal is sent if bb = 1 is restored or the seize
acknowledge guard timeout is reached.
23.
Set the Fault On Seize Guard Timeout field with the maximum time, in milliseconds (ms), an
In the case bb = 0 is recognized while in the seize state, no immediate action is taken. However,
when the seize acknowledgement signal is recognized after the seize ack timeout period has
elapsed or the seize guard timeout is reached, the clear forward signal is sent.
24.
Set the Decline Guard Timeout field to determine the maximum time the AB=10 release signal is
applied before sending the AB=11clearback signal. This variable applies when
IncomingDeclineMethod is set to ClearBack while declining a call. The value is expressed in
milliseconds (ms).
25.

This section allows you to decide whether or not you want to override the default R2 digit timers. The Mediatrix
unit uses the following default values:
Table 217: R2 Digit Timers Default Values
Default Value (ms)
Parameter
Bra.
MFC Pulse Inter Digit
Timeout
238
100
Mex.
100
Arg.
100
Sau.
100
Ven.
100
Phi.
100
ITU-T
100
Table 217: R2 Digit Timers Default Values (Continued)

Default Value (ms)
Parameter
Bra.
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
MFC Pulse Min On

Timeout
150
150
150
150
150
150
150
MFC Max Sequence

Timeout
10000
20000
20000
20000
20000
20000
20000
MFC Max On Timeout 5000
10000
10000
10000
10000
10000
10000
MFC Max Off Timeout 5000
10000
10000
10000
10000
10000
10000

You can override the default R2 digit timers. In that case, you will have access to the R2 Digit Timers Variants
section to define the timers you want.
To override the R2 digit timers default settings:

1.
In the web interface, click the R2 link, then the Digit Timers sub-link.
Figure 95: R2 Digit Timers Variants Web Page
2.
3.
Select whether or not you want to override the default setting of R2 digit timers in the Override
Default Country Settings drop-down menu.
Table 218: R2 Digit Timers Override
Allocation
Description
Disable
Enable

R2 Digit Timers Variants section. To retrieve the default configuration
Proceed to R2 Digit Timers Variants on page 239.

This section allows you to define R2 digit timers. You can click the Reset to Default button at any time to revert
back to the default R2 digit timers.
239
R2 Link Timers Variants
To set R2 digit timers:

1.
In the R2 Digit Timers Variants section, set the MFC Pulse Inter Digit Timeout field with the minimum
delay, in milliseconds (ms), between the end of transmission of the last signal of the compelled cycle
and the start of the next one.
Figure 96: R2 Digit Timers Variants Section
1
3
5
2
4
2.
Set the MFC Pulse Min On Timeout field with the minimum time, in milliseconds (ms), a backward
tone can be on from the backward perspective.
3.
Set the MFC Max Sequence Timeout field with the maximum time, in milliseconds (ms), for a
complete compelled signaling cycle from the forward perspective.
4.
Set the MFC Max On Timeout field with the maximum time, in milliseconds (ms), a forward tone can
be on from the forward perspective.
5.
Set the MFC Max Off Timeout field with the maximum time, in milliseconds (ms), a forward tone can
be off from the forward perspective.
6.

This section allows you to decide whether or not you want to override the default R2 link timers. The Mediatrix
unit uses the following default values:
Table 219: R2 Link Timers Default Values
Default Value (ms)
Parameter
Bra.
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
Link Activation
Timeout
1000
1000
1000
1000
1000
1000
1000
Link Activation Retry

Timeout
3000
3000
3000
3000
3000
3000
3000

You can override the default R2 link timers. In that case, you will have access to the R2 Link Timers Variants
240
To override the R2 link timers default settings:

1.
In the web interface, click the R2 link, then the Link Timers sub-link.
Figure 97: R2 Link Timers Variants Web Page
2.
3.
Select whether or not you want to override the default setting of R2 link timers parameters in the
Override Default Country Settings drop-down menu.
Table 220: R2 Link Timers Override
Allocation
Description
Disable
Enable

R2 Link Timers Variants section. To retrieve the default configuration
Proceed to R2 Link Timers Variants on page 241.

This section allows you to define R2 link timers. You can click the Reset to Default button at any time to revert
back to the default R2 link timers.
To set R2 link timers:

1.
In the R2 Link Timers Variants section, set the Link Activation Timeout field with the maximum time,
in milliseconds (ms), the unit waits for an activation indication coming from the physical link.
The activation indication is used to indicate that the physical layer connection has been activated.
Figure 98: R2 Link Timers Variants Section
1
2
2.
Set the Link Activation Retry Timeout field with the maximum time, in milliseconds (ms), the unit
waits before attempting to re-establish the physical link.
The attempt is made when the physical layer connection has been deactivated.
3.
241
R2 Tones Variants
R2 Tones Variants
This section allows you to decide whether or not you want to override the default R2 tones parameters. The
Table 221: R2 Tones Default Values
Default Value (ms)
Parameter
Bra.
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
R2 Tones Forward Groups

End of DNIS Tone
I15
None
I15
I15
None
I15
I15
End of ANI Tone
I15
I15
I15
I15
I15
I15
I15
Restricted ANI Tone
I12
None
None
None
I12
None
None
R2 Tones Backward Groups
242
Send Next DNIS Digit

Tone
A1
A1
A1
A1
A1
A1
A1
Send Previous DNIS

Digit Tone
A9
None
A9
A2
A2
A9
A9
Switch to Group II
Tone
A3
A3
A3
A3
A3
A3
A3
Network Congestion
Tone
A4
A4
A4
A4
A4
A4
A4
Send Calling Party

Category Tone
A5
None
A5
A5
A5
A5
A5
Immediate Accept
Tone
None
None
A6
A6
A6
A6
A6
Send DNIS Digit N-2

Tone
A7
None
A7
A7
A7
A7
A7
Send DNIS Digit N-3

Tone
A8
None
A8
A8
A8
A8
A8
Repeat all DNIS Tone
None
None
A10
None
A10
A10
A10
Send Next ANI Digit

Tone
A5
A2
A5
A5
A9
A5
A5
Send Calling Party

Category Tone Switch
to Group C
None
A6
None
None
None
None
None
Send Special
Information Tone
None
None
B2
B2
B2
B2
B2
User Busy Tone
B3
B2
B3
B3
B3
B3
B3
Network Congestion
Tone
B4
B4
B4
B4
B4
B4
B4
Unassigned Number
Tone
B8
None
B7
B5
B5
B5
B5
Line Free with Charge

Tone
B1
B1
B6
B1
B6
B6
B6
R2 Tones Variants
Table 221: R2 Tones Default Values (Continued)

Default Value (ms)
Parameter
Bra.
Mex.
Arg.
Sau.
Ven.
Phi.
ITU-T
Line Free with Charge B6

Tone (Supplementary)
None
None
None
None
None
None
Line Free without

Charge Tone
B7
B5
B7
B6
B7
B7
B7
Line Out of Order

Tone
B8
None
B8
B8
B8
B8
B8
Changed Number
Tone
B3
None
None
None
None
None
None
Send Next ANI Digit

Tone
None
C1
None
None
None
None
None
Repeat All DNIS Tone

switch to Group A
None
C2
None
None
None
None
None

Tone switch to Group
A
None
C5
None
None
None
None
None
Network Congestion
Tone
None
C4
None
None
None
None
None
Send Previous DNIS

Digit Tone switch to
Group A
None
C6
None
None
None
None
None
Switch to Group II
Tone
None
C3
None
None
None
None
None

You can override the default R2 tones. In that case, you will have access to the R2 Tones Forward Groups
and R2 Tones Backward Groups sections to define the tone you want.
To override the R2 tones default settings:

1.
In the web interface, click the R2 link, then the Tones sub-link.
Figure 99: R2 Tones Variants Web Page
2.
243
3.
R2 Tones Variants
Select whether or not you want to override the default setting of R2 tones parameters in the Override
Default Country Settings drop-down menu.
Table 222: R2 Tones Override
Allocation
Description
Disable
Enable

R2 Tones Forward Groups and R2 Tones Backward Groups sections.
To retrieve the default configuration associated with the current
country, click the Reset to Default button. Proceed to R2 Tones
Forward Groups on page 244.
R2 Tones Forward Groups

These tones fall into Groups I and II. All Group I and II tones use the ITU-T R2 forward group tone frequencies.
Most of the tones in Group I are used for addressing and identification. Group II tones give information about
the origin of the call. Group II tones are also sent by an outgoing MFC/R2- or International MFC/R2-register in
response to one of the following backward signals:
Change over to the reception of Group B signals.

Send calling party's category.
Table 223: R2 Group I Forward Tones
MF
Tone
Description
I-1
Digit 1 (Language: French, if first signal sent in intl. link)
I-2
Digit 2 (Language: English, if first signal sent intl. link)
I-3
Digit 3 (Language: German, if first signal sent in intl. link)
I-4
Digit 4 (Language: Russian, if first signal sent in intl. link)
I-5
Digit 5 (Language: Spanish, if first signal sent in intl. link)
I-6
Digit 6 (Language: Spare, if first signal sent in intl. link)
I-7
I-8
I-9
Digit 9 (Discriminating digit, if first signal sent in intl. link)
10
I-10
Digit 0 (Discriminating digit, if first signal sent in intl. link)
11
I-11
Country code indicator, outgoing half-echo suppressor required
12
I-12
Country code indicator, no echo suppressor required
13
I-13
Test call indicator (call by automatic test equipment)
14
I-14
Country code indicator, outgoing half-echo suppressor inserted
15
I-15
Signal is not used
You can click the Reset to Default button at any time to revert back to the default R2 tones.
To set the R2 tones forward groups:

1.
In the R2 Tones Forward Groups section, select the forward Group 1 tone used to send after the
DNIS digits in the End of DNIS Tone drop-down menu.
You have the choice between none and I1 to I15 Group 1 forward signals MF Tone.
244
R2 Tones Variants
Figure 100: R2 Tones Forward Groups Section

1
3
2.
Select the forward Group 1 tone used to send after the ANI digits in the End of ANI Tone drop-down
menu.
3.
Select the forward Group 1 tone used to reject a query in the Restricted ANI Tone drop-down menu.
This tone is generally used in response to the identification request when the caller party is unable
to send his identification to the called party. If no tone is defined, the End of ANI tone is sent to the
caller.
4.
R2 Tones Backward Groups

Backward tones fall into Groups A and B. The Group A tones acknowledge Group I forward signals. Under
certain conditions, Group A tones also acknowledge Group II tones. Group B tones convey the following
information to an outgoing MFC/R2 register:
The condition of the switch equipment in the incoming exchange.

The condition of the called subscriber's line.
Table 224: R2 Group A Backward Tones
MF
Tone
Description
A-1
Send next digit (n + 1)
A-2
Send last but one digit (n -1)
A-3
Address-complete, changeover to reception of Group B signals
A-4
Congestion in the national network
A-5
Send calling partys category
A-6
Address-complete, charge, set-up speech conditions
A-7
Send last but two digit (n - 2)
A-8
Send last but three digit (n - 3)
A-9
Spare for national use
10
A-10
11
A-11
Send country code indicator
12
A-12
Send language or discrimination digit
13
A-13
Send nature of circuit
14
A-14
Request for information on use of an echo suppressor
15
A-15
Congestion in an international exchange or at its output

Table 225: R2 Group B Backward Tones
MGF Tone
Description
B-1
B-2
Send special information tone
245
R2 Tones Variants
Table 225: R2 Group B Backward Tones (Continued)

MGF Tone
Description
B-3
Subscribers line busy
B-4
Congestion
B-5
Unallocated number
B-6
Subscribers line free, charge
B-7
Subscribers line free, no charge
B-8
Subscribers line out of order
B-9
10
B-10
11
B-11
12
B-12
13
B-13
14
B-14
15
B-15

Table 226: R2 Group C Backward Tones
MGF Tone
Description
C-1
Send next number of Subscriber A
C-2
Send first digit BNUM
C-3
Send Group II and Group B signals
C-4
Congestion
C-5
Send next digit BNUM
C-6
Repeat last digit BNUM
C-7
C-8
C-9
10
C-10
11
C-11
12
C-12
13
C-13
14
C-14
15
C-15
To set the R2 tones backward groups:

1.
In the R2 Tones Backward Groups section, select the backward Group A tone used to request the
next DNIS digit in the Send Next DNIS Digit Tone drop-down menu.
You have the choice between none and A1 to A15 Group A backward signals MF Tone.
246
R2 Tones Variants
Figure 101: R2 Tones Backward Groups Section

1
3
5
2
4
6
7
9
11
13
15
8
10
12
14
16
17
18
19
20
21
22
23
24
25
26
2.
Select the backward Group A tone used to request the previous DNIS digit in the Send Previous
DNIS Digit Tone drop-down menu.
3.
Select the backward Group A tone used to request to the caller a switch of Group II signals in the
Switch to Group II Tone drop-down menu.
4.
Select the backward Group A tone to be sent when a congestion network is detected in the Network
Congestion Tone drop-down menu.
5.
Select the backward Group A tone sent when the backward group requests the calling party
category in the Send Calling Party Category Tone drop-down menu.
6.
Select the backward Group A tone sent when the backward group accepts the call immediately in
the Immediate Accept Tone drop-down menu.
7.
Select the backward Group A tone used to request the previous - 1 DNIS digit in the Send DNIS
Digit N-2 Tone drop-down menu.
8.
Select the backward Group A tone used to request the previous - 2 DNIS digit in the Send DNIS
Digit N-3 Tone drop-down menu.
9.
Select the backward Group A tone used to request all DNIS digits in the Repeat All DNIS Tone dropdown menu.
10.
Select the backward Group A tone used to request the next ANI digit in the Send Next ANI Digit
Tone drop-down menu.
247
11.
R2 Tones Variants
Select the backward Group A tone used to request the calling party category and then switch to
Group C signals in the Send Calling Party Category Tone Switch to Group C drop-down menu.
12.
Select the backward Group B tone used to send a special information tone in the Send Special
Information Tone drop-down menu.
You have the choice between none and B1 to B15 Group B backward signals MF Tone.
13.
Select the backward Group B tone used to signal a user busy in the User Busy Tone drop-down
menu.
14.
Select the backward Group B tone used to signal a network congestion in the Network Congestion
15.
Select the backward Group B tone used to signal a unassigned number in the Unassigned Number
16.
Select the backward Group B tone used to signal that the line is free and charge must be applied in
the Line Free with Charge Tone drop-down menu.
17.
Select a supplementary backward Group B tone used to detect that the line is free and charge must
be applied in the Line Free with Charge Tone (Supplementary) drop-down menu.
This is a supplementary (optional) tone used on the reception only. You have the choice between
none and B1 to B15 Group B backward signals MF Tone.
18.
Select the backward Group B tone used to signal that the line is free and no charges must be
applied in the Line Free without Charge Tone drop-down menu.
19.
Select the backward Group B tone used to signal that the line is out of order in the Line Out of Order
20.
Select the backward Group B tone used to signal that the subscriber has changed number in the
Changed Number Tone field.
21.
Select the backward Group C tone used to request the next ANI digit in the Send Next ANI Digit
You have the choice between none and C1 to C15 Group C backward signals MF Tone.
22.
Select the backward Group C tone used to request all DNIS digits and then switch to Group A
signals in the Repeat All DNIS Tone switch to Group A drop-down menu.
23.
Select the backward Group C tone used to request the next DNIS digit and then switch to Group A
signals in the Send Next DNIS Digit Tone switch to Group A drop-down menu.
24.
Select the backward Group C tone used to signal a network congestion in the Network Congestion
25.
248
Select the backward Group C tone used to request the previous DNIS digit and then switch to Group
A signals in the Send Previous DNIS Digit Tone switch to Group A drop-down menu.
26.
Select the backward Group A tone used to request to the caller a switch of Group II signals in the
Switch to Group II Tone drop-down menu.
27.
clock reference signals sent over IP. It allows you to to synchronize multiple PRI or BRI units over a network
for more details.
Parameter
Unit
Description
Precision
Buffer
considered.
IP Synchronization is enabled when at least one hardware card listed in the BRI Cards Status or PRI Cards
Status section of the System > Hardware page has its Clock Reference value set to None (see BRI Hardware
Configuration on page 61 or PRI Hardware Configuration on page 62 for more details). If two cards of a
same unit have their Clock Reference values set to None, then the card located in the lowest Slot ID is
synchronized and you must configure the other card so that it synchronizes on the first card by setting the
corresponding Clock Reference parameter of the BRI Cards Configuration or PRI Cards Configuration section
to Other Card.

firmware profile.
249

1.
In the web interface, click the R2 link, then the IP Sync sub-link.
Figure 102: R2 IP Sync Web Page, Master Unit
2.
3.


1.
In the web interface, click the R2 link, then the IP Sync sub-link.
Figure 103: R2 IP Sync Web Page, Slave Unit
2
4
2.
3.
4.
5.

for more details.
250
PRI R2 CAS Statistics

1.

2.

cd IpSync
ResetSyncData


by using the CLI
The Mediatrix unit collects meaningful statistics for each PRI digital card that can be read via SNMP and CLI.
location of the cards in the unit, Slot 2 being on the left when looking at the rear of the unit.
To view the PRI statistics:

1.
In the ex1Pri_1MIB, locate the statisticsGroup and expand it.

Table 228: R2 CAS Statistics Displayed
Statistic
statsInfoFramesTra
nsmitted
Description
statsInfoFramesRec Number of HDLC frames received.

eived
statsInfoOctetsTran
smitted
Number of octets transmitted. This value is obtained by cumulating the

octets transmitted in the HDLC frames.
statsInfoOctetsRece Number of octets received. This value is obtained by cumulating the

ived
octets received in the HDLC frames.
statsInfoFCSErrors
Frame check sequence (FCS) errors indicate that frames of data are
being corrupted during transmission. FCS error count is the number of
frames that were received with a bad checksum (CRC value) in the HDLC
frame. These frames are dropped and not propagated in the upper layers.
statsInfoFramesDro
pped
Number of frames dropped. This value is obtained by cumulating the

number of frames dropped when transferring the data from the framer
chip to the device internal buffer.
251
Table 228: R2 CAS Statistics Displayed (Continued)

Statistic
Description
statsInfoOctetsDrop
ped
Number of octets dropped. This value is obtained by cumulating the

number of octets dropped when transferring the data from the framer chip
to the device internal buffer.
statsInfoNegativeFr
ameSlipsTransmitte
d
A frame is skipped when the frequency of the transmit clock is greater

than the frequency of the transmit system interface working clock based
on 2.048 MHz (on E1) or 1.544 MHz (on T1).
statsInfoNegativeFr
ameSlipsReceived
A frame is skipped when the frequency of the received route clock is

greater than the frequency of the receive system interface working clock
based on 2.048 MHz (on E1) or 1.544 MHz (on T1).
statsInfoPositiveFra
meSlipsTransmitted
A frame is repeated when the frequency of the transmit clock is less than
the frequency of the transmit system interface working clock based on
2.048 MHz (on E1) or 1.544 MHz (on T1).
meSlipsReceived
A frame is repeated when the frequency of the receive route clock is less
than the frequency of the receive system interface working clock based
statsInfoFramingErr
or
The framing error count indicates that a FAS (Frame Alignment Signal)
word has been received with an error.
statsInfoCodeViolati
ons
The code violations count indicates that an encoding error on the PCM
line has been detected.
statsInfoCRCErrors
The CRC errors count is incremented when a multiframe has been

received with a CRC error.
statsInfoE-BitError
The E-Bit error count gives information about the outgoing transmit PCM
line if the E-bits are used by the remote end for submultiframe error
indication. Incrementing is only possible in the multiframe synchronous
state.
Due to signaling requirements, the E-bits of frame 13 and frame 15 of the
CRC multiframe can be used to indicate an error in a received
submultiframes:
CRC error : E = 0
252
Table 228: R2 CAS Statistics Displayed (Continued)

Statistic
Description
statsInfoBlockError
The Block Error count is incremented once per multiframe if a multirame

has been received with a CRC error or a bad frame alignment has been
detected.
To reset the statistics:

1.
In the ex1Pri_1MIB, set the statsInfoResetStats variable to 10: resetStats.

ex1Pri_1.statsInfoResetStats=10
253
254
E&M CAS Parameters
H A P T E R
23
E&M CAS Configuration

This chapter describes how to configure the E&M CAS parameters of the Mediatrix unit.
Standards Supported
ITU-T Recommendation Q.310-Q.332: Specifications of

Signalling System R1a
a. Line signalling, Register signalling
Note: This chapter applies to the following models:

Mediatrix 3531
Mediatrix 3532
Mediatrix 3621
Mediatrix 3631
Mediatrix 3632
Introduction
CAS stands for Channel Associated Signaling. With this method of signaling, each traffic channel has a
dedicated signaling channel. In other words, the signaling for a particular traffic circuit is permanently
associated with that circuit. Channel-associated call-control is still widely used today, mostly in South America,
Africa, Australia, and in Europe.
E&M (earth & magneto, or ear & mouth) is a type of CAS signalling that defines line signaling and register
signaling. It is also called Signalling System R1 and is mainly used in North America.
E&M was originally developed to allow PABXs in different geographic locations to communicate over an
analog private circuit. Some digital interfaces such as CAS also use versions of E&M signaling.
The terms forwards and backwards are heavily used in descriptions of E&M. Forwards is the direction from
the calling party to the called party. Backwards is the direction from the called party to the calling party.
You can configure Mediatrix unit parameters for the E1/T1 E&M CAS.
Line Signals for the Digital Version of E&M

Line signalling uses the ABCD bits of CAS. Several types of E&M signalling exist.
Line Signals for the Analogue Version of E&M

E&M has its roots in older analogue signalling. The digital versions of E&M replicate the operation of older
analogue signalling versions.
Inter-Register Signals (Defined in ITU-T Q.310-Q.332)

Inter-register signalling uses either R1 tones (defined in Q.310-Q.332) or DTMF depending on the signalling
type and detailed user-defined variant configuration.
The inter-register signals are sent in-band. They may pass transparently through several nodes in the network
between the two terminating switches.
255
Chapter 23 - E&M CAS Configuration
Selecting the E&M Signalling Protocol
Selecting the E&M Signalling Protocol

You must set the unit to use the E&M signalling protocol. You can do so in the System / Hardware page. The
Hardware page differs depending on the product and model you have.

1.
2.
If you want to configure the clock reference of a specific interface, you must set the Clock Mode
drop-down menu to Master. See E&M Channel Associated Signaling on page 258 for more
details.
Reference
Description
None
Other
Card
The internal clock synchronizes with the other E&M interface of the Mediatrix unit.
This interface must be configured in Slave mode (Clock Mode drop-down menu of
the E&M Channel Associated Signaling section) to provide the clock reference to
the other interfaces.
3.
4.
Select the E&M signaling in the Signaling drop down menu.

When changing from E&M to ISDN/R2 or ISDN/R2 to E&M, you must change your routes
accordingly. For instance, if you are in ISDN with a route isdn-Slot2/E1T1, then change to E&M, you
must change the route to e&m-Slot2/E1T1.
5.
256
E&M Auto-Configuration
E&M Auto-Configuration
The E&M Auto-configuration feature allows you to detect and to configure all E&M interfaces so that the E&M
link goes up and becomes usable with a minimal user interaction. When launching an auto-configuration
process, it stops automatically when all interfaces have been tested. For each interface, the autoconfiguration process is considered successful when the link becomes up or a failure when all combinations
have been tried without having a link up.
Caution: Launching the auto-configuration may terminate abruptly all ongoing E&M calls.
Note: Auto-configuration on all E&M interfaces may take some time to complete. Some of the current E&M

1.
In the web interface, click the E&M link, then the Status sub-link.
Figure 105: E&M Status Configuration Section
2.

The process starts.
Preset
The E&M Preset Configuration section allows you to load a set of preset configuration for your E&M
connections. These preset files are located in the file system's persistent memory. They differ depending on
the Mediatrix unit you are using. Depending on your unit's profile, it may be possible that no preset files are
available.
You can also export your current E&M configuration in a preset. Please note that these user-defined presets
are not kept in the event of a partial or factory reset.
257
E&M Channel Associated Signaling

1.
In the E&M Status tab, E&M Preset Configuration section, select one of the available preset files in
the Local Preset drop-down menu.
1
2.
Click Apply.
To export the current E&M configuration as a preset:

1.
In the E&M Preset Configuration section, type a name for the preset in the Preset Name field.
2.
Click Save.
The current E&M configuration is exported. Please note that these user-defined presets are not kept
in the event of a partial or factory reset.
Partial Reset

The E&M Channel Associated Signaling section allows you to define the general parameters related to E&M.
258
To configure the E&M CAS parameters:

1.
In the web interface, click the E&M link, then the E&M Channel Associated Signaling Config sublink.
Figure 108: E&M Channel Associated Signaling Web Page
2
3
4
5
6
7
9
8
10
11
13
2.
12
3.
Table 229: E&M Interface Clock Mode
Mode
Description

Slave
The clock source can be selected from the Clock Reference drop-down menu (see Selecting the
E&M Signalling Protocol on page 256 for more details). The clock mode could be used, for
instance, to synchronize several units in master clock mode via an E1 line.
4.

Mode
Auto
Description
The pinout is set according to the Clock Mode parameter setting (Step 3).
Master Forces the pinout to Master regardless of the clock mode.

Slave
5.
Forces the pinout to Slave regardless of the clock mode.
Set the Monitor Link State drop-down menu with the physical link state of the E&M interface.
Parameter
Enable
Description
The E&M endpoint's operational state is affected by its interface physical link state.
When the link state of an E&M interface is down, the operational state of its
matching endpoint becomes "disable".
259
Table 231: Interface Link State (Continued)

Parameter
Description
Disable
The E&M endpoint's operational state is not affected by its interface physical link
state
6.
Coding
Description
B8ZS
Bipolar with 8-Zeros Substitution (T1 lines).
HDB3
AMI
7.

their frames.
Format
Description
SF
Super frame. Sometimes known as D4 (T1 lines).
ESF
Extended super frame (T1 lines).
CRC4

8.
9.
menu.
Allocation
260
Description
Ascending

Descending

Table 234: Channel Allocation Strategy (Continued)

Allocation
RoundRobinAscending
Description
10.
11.
Set the Encoding Scheme drop-down menu with the voice encoding scheme in the bearer
capabilities.
This encoding scheme is used when initiating a call on the E&M side. The supported encoding
12.
Select the E&M Signaling Type Selection drop-down menu with the proper E&M signalling type.
Table 235: Signalling Type Selection
Signalling Type
Wink start
Description
Sets configuration parameters for basic Wink Start signalling. After
setting the SignallingType, individual configuration parameters can be
overriden for detail adjustment.
Note that the outgoing and incoming register signaling uses DTMF
signaling type by default.
Immediate Start
Sets configuration parameters for basic Immediate Start signalling.

After setting the Signaling Type, individual configuration parameters
can be overriden for detail adjustment.
Note that the outgoing and incoming register signaling uses DTMF
Feature Group B
Sets configuration parameters for the Feature Group B signalling

defined by National Exchange Carrier Association. After setting the
Signaling Type, individual configuration parameters can be overriden
for detail adjustment.
Note that the outgoing and incoming register signaling uses MF R1
Feature Group D
Sets configuration parameters for the Feature Group D signalling

defined by National Exchange Carrier Association. After setting the
Signaling Type, individual configuration parameters can be overriden
for detail adjustment.
Note that the outgoing and incoming register signaling uses MF R1
13.
Set the Digit Attenuation field with the additional attenuation, in dB, for MFR1/DTMF digits
generation.
261
E&M Signalling Variants
By default, MFR1/DTMF digits generation power is determined by variant selection. This parameter
provides a mean to reduce this power.
14.

This section allows you to decide whether or not you want to override the default E&M signalling parameters.
The Mediatrix unit uses the following default values:
Table 236: E&M Signaling Parameters Default Values
Default Value (ms)
Parameter
Wink Start
FGBa
Immediate Start
FGDb
Bits BCD
ANI Length
DNIS Length
Incoming Register Signaling DTMF
DTMF
MfR 1
MfR 1
Outgoing Register Signaling DTMF
DTMF
MfR 1
MfR 1
Incoming Dial Map
%dnis%t
%dnis%t
%kp%dnis%st
%kp%ani%st%k
p%dnis%st
Outgoing Dial Map
%dnis%t
%dnis%t
%kp%dnis%st
%kp%dnis%st
Wait Wink
Enable
Disable
Enable
Enable
Wait Wink Ack
Disable
Disable
Disable
Enable
Send Wink
Enable
Disable
Enable
Enable
Send Wink Ack
Disable
Disable
Disable
Enable
a. Feature Group B
b. Feature Group D
Override Default Signaling Settings

You can override the default E&M signaling parameters. In that case, you will have access to the E&M
Signaling Variants section to define the signaling you want.
To override the E&M signaling default settings:

1.
In the web interface, click the E&M link, then the Signaling sub-link.
Figure 109: E&M Signaling Variants Web Page
2.
262
3.
Select whether or not you want to override the default setting of E&M signalling parameters in the
Override Default Signaling Settings drop-down menu.
Table 237: E&M Signaling Override
Allocation
Description
Disable
The interface uses the default configuration associated with the

selected Signaling Type. The configuration set in the current row has
no effect on the default configuration of the selected Signaling Type.
Enable
The interface uses the specific signalling configuration as defined in

the E&M Signaling Variants section. To retrieve the default
configuration associated with the current Signaling Type, click the
Reset to Default button. Proceed to E&M Signalling Variants on
page 263.
Overriding the default settings is considered an advanced configuration. Media5 recommends not
to modify the signalling variants unless you know exactly what you are doing.

This section allows you to define E&M signalling parameters. You can click the Reset to Default button at any
time to revert back to the default E&M signalling.
To set E&M signalling parameters:

1.
In the E&M Signaling Variants section, set the B, C and D bits when the device transmits line signals
in the Bits BCD field.
The device ignores the B, C and D bits of received line signals.
Figure 110: E&M Signaling Variants Section
1
3
5
7
9
11
2
4
6
8
10
BCD bits definition
0..7 : BCD bits are set to the specified value.
8 : BCD bits follow the A bit.
Note: On E1-CAS, the ABCD bit value cannot be set to 0000. If the BCD variable is set to 000 or to follow
the A bit, then the bitmask BCD=001 will be used instead.
2.
Set the ANI Length field with the expected length of Automatic Number Identification (ANI) to be
requested or sent.
ANI is the number of the calling party.
3.
0: Variable ANI length.
1..20: Specific ANI length.
Set the DNIS Length field with the expected length of the Dialed Number Identification Service
(DNIS).
263
DNIS is the called party or the destination number.
4.
0: Variable DNIS length used.
1..20: Specific DNIS length expected.
Set the Incoming Register Signaling drop-down menu with the proper incoming register signaling
method.
Table 238: Incoming Register Signalling Parameters
Parameter
5.
Description
MfR1
Multi Frequency - R1.
DTMF
Dual Tone Multi Frequency.
Set the Outgoing Register Signaling drop-down menu with the proper outgoing register signaling
method.
Table 239: Outgoing Register Signalling Parameters
Parameter
6.
Description
MfR1
Multi Frequency - R1.
DTMF
Dual Tone Multi Frequency.
Set the Incoming Dial Map field with the dial map expression to match against on the incoming side.
The dial map expression uses a special format using macros that enables you to construct a custom
dial map using the ANI, DNIS and other special tones as separator. The dial map macros supported
are:
%dnis : DNIS (Dialed Number Identification Service).
%ani : ANI (Automatic Number Identification).
%kp : KP (start-of-pulsing) tone (MF R1 only).
%st : ST (end-of-pulsing) tone (MF R1 only).
%t : Interdigit timeout.
A,B,C,D,*,# : Control tones (DTMF only).
A,B,C,D,E : Control tones (MF R1 only).
Examples:
7.
%dnis*%ani : (dnis)*(ani)
%kp%dnis%st%kp%ani%st : KP(dnis)STKP(ani)ST
A%dnisBA%aniB : A(dnis)BA(ani)B
Set the Outgoing Dial Map field with the dial map format for the outgoing dial string.
The dial string generation uses a special format using macros that enables you to construct a
custom dial string using the ANI, DNIS and other specials tones used as separator. The dial string
uses regular expression to replace the macros with the proper call parameter value. The dial map
macros supported are:
%dnis : DNIS (Dialed Number Identification Service).
%ani : ANI (Automatic Number Identification).
%kp : KP (start-of-pulsing) tone (MF R1 only).
%st : ST (end-of-pulsing) tone (MF R1 only).
%t : Interdigit timeout.
A,B,C,D,*,# : Control tones (DTMF only).
A,B,C,D,E : Control tones (MF R1 only).
Examples : ANI=1234 DNIS=6789 KP=A ST=D
264
%dnis*%ani : 6789*1234
E&M Timers Variants
%kp%dnis%st%kp%ani%st : A6789DA1234D
A%dnisBA%aniB : A6789BA1234B
8.
Set the Wait Wink drop-down menu with whether or not the outgoing register should wait for a wink
before proceeding with digit transmission.
9.
Set the Wait Wink Ack drop-down menu with whether or not the outgoing register should wait for a
wink acknowledge after all digit reception.
10.
Set the Send Wink drop-down menu with whether or not the incoming register should send a wink
to notify the remote side that digit information can be sent.
11.
Set the Send Wink Ack drop-down menu with whether or not the incoming register should send a
wink to acknowledge the receipt of all digits.
12.
E&M Timers Variants

This section allows you to decide whether or not you want to override the default E&M timers parameters. The
Table 240: E&M Timers Default Values
Default Value (ms)
Parameter
Wink Start
FGBa
Immediate Start
FGDb
Backward Wait Pre Wink Timeout
50
50
50
50
Backward Send Wink Timeout
200
200
200
200
Backward Wait First Digit Timeout
10000
10000
10000
10000
Backward Clear Backward Timeout
2000
2000
2000
2000
Backward Digit Complete Timeout
4000
4000
4000
4000
Forward Wait Wink Timeout
5000
5000
5000
5000
Forward Wait Max Wink on Timeout 5000
5000
5000
5000
Forward Wait Pre Dial Timeout
140
140
140
140
Forward Wait Answer Timeout
180000
180000
180000
180000
Forward Clear Forward Timeout
2000
2000
2000
2000
Release Guard Timeout
200
200
200
200
a. Feature Group B
b. Feature Group D

You can override the default E&M timers. In that case, you will have access to the E&M Timers Variants
265
E&M Timers Variants
To override the E&M timers default settings:

1.
In the web interface, click the E&M link, then the Timers sub-link.
Figure 111: E&M Timers Variants Web Page
2.
3.
Select whether or not you want to override the default setting of E&M timers in the Override Default
Signaling Settings drop-down menu.
Table 241: E&M Timers Override
Allocation
Description
Disable

Enable
The interface uses the specific signaling configuration as defined in

the E&M Timers Variants section. To retrieve the default configuration
associated with the current signalling type, click the Reset to Default
button. Proceed to E&M Timers Variants on page 266.
E&M Timers Variants

This section allows you to define E&M timers. You can click the Reset to Default button at any time to revert
back to the default E&M timers.
To set E&M timers:

1.
In the E&M Timers Variants section, set the Backward Wait Pre Wink Timeout field with the amount
of time, in milliseconds (ms), an incoming register waits before sending the wink that acknowledges
the seizure.
Figure 112: E&M Timers Variants Section
1
3
5
2
4
6
7
9
11
2.
266
8
10
Set the Backward Send Wink Timeout field with the duration, in milliseconds (ms), of the wink signal
is applied by the incoming register to signal seizure acknowledgment.
E&M Digit Timers Variants
3.
Set the Backward Wait First Digit Timeout field with the maximum time, in milliseconds (ms), an
incoming register waits for the first incoming digit after the line seizure.
4.
Set the Backward Clear Backward Timeout field with the maximum time, in milliseconds (ms), an
incoming register waits after sending a clear backward line signal before transiting to the idle state.
5.
Set the Backward Digit Complete Timeout field with the maximum time, in milliseconds (ms), the
incoming register waits for the next digit before considering the digit sequence as completed.
6.
Set the Forward Wait Wink Timeout field with the maximum time, in milliseconds (ms), an outgoing
register waits for seizure acknowledgement after seizing the line.
7.
Set the Forward Wait Max Wink On Timeout field with the maximum time, in milliseconds (ms), an
outgoing register waits for the seizure acknowledgment wink to complete.
8.
Set the Forward Wait Pre Dial Timeout field with the amount of time, in milliseconds (ms), an
outgoing register waits after the wink to start dialing.
9.
Set the Forward Wait Answer Timeout field with the maximum time, in milliseconds (ms), an
outgoing register waits for the call to be answered.
10.
Set the Forward Clear Forward Timeout field with the maximum time, in milliseconds (ms), an
outgoing register waits after sending a clear forward line signal before transiting to the idle state.
11.
Set the Release Guard Timeout field with the maximum time, in milliseconds (ms), a register waits
after sending an idle line signal to prevent a new seizure of the line.
12.

This section allows you to decide whether or not you want to override the default E&M digit timers. The
Table 242: E&M Digit Timers Default Values
Default Value (ms)
Parameter
Wink Start
FGBa
Immediate Start
FGDb
KP On Timeout
100
100
100
100
KP Off Timeout
68
68
68
68
a. Feature Group B
b. Feature Group D

You can override the default E&M digit timers. In that case, you will have access to the E&M Digit Timers
Variants section to define the timers you want.
267
To override the E&M digit timers default settings:

1.
In the web interface, click the E&M link, then the Digit Timers sub-link.
Figure 113: E&M Digit Timers Variants Web Page
2.
3.
Select whether or not you want to override the default setting of E&M digit timers in the Override
Default Signaling Settings drop-down menu.
Table 243: E&M Digit Timers Override
Allocation
Description
Disable

Enable

the E&M Digit Timers Variants section. To retrieve the default
configuration associated with the current signalling, click the Reset to
Default button. Proceed to E&M Digit Timers Variants on page 268.

This section allows you to define E&M digit timers. You can click the Reset to Default button at any time to
revert back to the default E&M digit timers.
To set E&M digit timers:

1.
In the E&M Digit Timers Variants section, set the KP On Timeout field with the time, in milliseconds
(ms), during which the MF R1 KP tone is on.
Figure 114: E&M Digit Timers Variants Section
1
2
268
2.
Set the KP Off Timeout field with the duration, in milliseconds (ms), of the pause after the MF R1
KP tone.
3.
E&M Link Timers Variants

This section allows you to decide whether or not you want to override the default E&M link timers. The
Table 244: E&M Link Timers Default Values
Default Value (ms)
Parameter
Wink Start
FGBa
Immediate Start
FGDb
Link Activation Timeout
1000
1000
1000
1000

Timeout
3000
3000
3000
3000
a. Feature Group B
b. Feature Group D
Override Default Signalling Settings

You can override the default E&M link timers. In that case, you will have access to the E&M Link Timers
Variants section to define the timers you want.
To override the E&M link timers default settings:

1.
In the web interface, click the E&M link, then the Link Timers sub-link.
Figure 115: E&M Link Timers Variants Web Page
2.
3.
Select whether or not you want to override the default setting of E&M link timers parameters in the
Table 245: E&M Link Timers Override
Allocation
Description
Disable

Enable

the E&M Link Timers Variants section. To retrieve the default
configuration associated with the current signalling, click the Reset to
Default button. Proceed to E&M Link Timers Variants on page 270.
269
E&M Tones Variants

This section allows you to define E&M link timers. You can click the Reset to Default button at any time to
revert back to the default E&M link timers.
To set E&M link timers:

1.
In the E&M Link Timers Variants section, set the Link Activation Timeout field with the maximum
time, in milliseconds (ms), the unit waits for an activation indication coming from the physical link.
The activation indication is used to indicate that the physical layer connection has been activated.
Figure 116: R2 Link Timers Variants Section
1
2
2.
Set the Link Activation Retry Timeout field with the maximum time, in milliseconds (ms), the unit
waits before attempting to re-establish the physical link.
The attempt is made when the physical layer connection has been deactivated.
3.
E&M Tones Variants

This section allows you to decide whether or not you want to override the default E&M tones parameters. The
Table 246: E&M Tones Default Values
Default Value (ms)
Parameter
Wink Start
Immediate Start
FGBa
FGDb
KP Tone
MFC 10
MFC 10
MFC 10
MFC 10
ST Tone
MFC 13
MFC 13
MFC 13
MFC 13
a. Feature Group B
b. Feature Group D
Override Default Signalling Settings

You can override the default E&M tones. In that case, you will have access to the E&M Tones Variants section
to define the tone you want.
270
E&M Tones Variants
To override the E&M tones default settings:

1.
In the web interface, click the E&M link, then the Tones sub-link.
Figure 117: E&M Tones Variants Web Page
2.
3.
Select whether or not you want to override the default setting of E&M tones parameters in the
Table 247: E&M Tones Override
Allocation
Description
Disable

Enable
The interface uses the specific signalling configuration as defined in

the E&M Tones Variants section. To retrieve the default configuration
associated with the current signalling, click the Reset to Default
button. Proceed to E&M Tones Variants on page 271.
E&M Tones Variants

This section allows you to define E&M tones. You can click the Reset to Default button at any time to revert
back to the default E&M tones.
To set the E&M tones:

1.
In the E&M Tones Variants section, set the KP Tone drop-down menu with the proper KP (start-ofpulsing) signal.
You have the choice between none and MFR0 to MFR14 Tones.
Figure 118: E&M Tones Variants Section
1
2
2.
Set the ST Tone drop-down menu with the proper ST (end-of-pulsing) signal.
You have the choice between none and MFR0 to MFR14 Tones.
3.
271
clock reference signals sent over IP. It allows you to to synchronize multiple BRI or PRI units over a network
for more details.
Parameter
Unit
Description
Precision
Buffer
considered.
IP Synchronization is enabled when at least one hardware card listed in the BRI Cards Status or PRI Cards
Status section of the System > Hardware page has its Clock Reference value set to None (see BRI Hardware
Configuration on page 61 or PRI Hardware Configuration on page 62 for more details). If two cards of a
same unit have their Clock Reference values set to None, then the card located in the lowest Slot ID is
synchronized and you must configure the other card so that it synchronizes on the first card by setting the
corresponding Clock Reference parameter of the BRI Cards Configuration or PRI Cards Configuration section
to Other Card.

firmware profile.
272

1.
In the web interface, click the E&M link, then the IP Sync sub-link.
Figure 119: E&M IP Sync Web Page, Master Unit
2.
3.


1.
In the web interface, click the E&M link, then the IP Sync sub-link.
Figure 120: E&M IP Sync Web Page, Slave Unit
2
4
2.
3.
4.
5.

for more details.
273
PRI E&M Statistics

1.

2.

cd IpSync
ResetSyncData
PRI E&M Statistics


by using the CLI
The Mediatrix unit collects meaningful statistics for each PRI digital card that can be read via SNMP and CLI.
location of the cards in the unit, Slot 2 being on the left when looking at the rear of the unit.
To view the PRI statistics:

1.
In the ex1Pri_1MIB, locate the statisticsGroup and expand it.

Table 249: E&M Statistics Displayed
Statistic
statsInfoFramesTra
nsmitted
Description
statsInfoFramesRec Number of HDLC frames received.

eived
statsInfoOctetsTran
smitted
Number of octets transmitted. This value is obtained by cumulating the

octets transmitted in the HDLC frames.
statsInfoOctetsRece Number of octets received. This value is obtained by cumulating the

ived
octets received in the HDLC frames.
statsInfoFCSErrors
Frame check sequence (FCS) errors indicate that frames of data are
being corrupted during transmission. FCS error count is the number of
frames that were received with a bad checksum (CRC value) in the HDLC
frame. These frames are dropped and not propagated in the upper layers.
statsInfoFramesDro
pped
Number of frames dropped. This value is obtained by cumulating the

number of frames dropped when transferring the data from the framer
chip to the device internal buffer.
274
PRI E&M Statistics
Table 249: E&M Statistics Displayed (Continued)

Statistic
Description
statsInfoOctetsDrop
ped
Number of octets dropped. This value is obtained by cumulating the

number of octets dropped when transferring the data from the framer chip
to the device internal buffer.
statsInfoNegativeFr
ameSlipsTransmitte
d
A frame is skipped when the frequency of the transmit clock is greater

than the frequency of the transmit system interface working clock based
statsInfoNegativeFr
ameSlipsReceived
A frame is skipped when the frequency of the received route clock is

greater than the frequency of the receive system interface working clock
based on 2.048 MHz (on E1) or 1.544 MHz (on T1).
meSlipsTransmitted
A frame is repeated when the frequency of the transmit clock is less than
the frequency of the transmit system interface working clock based on
2.048 MHz (on E1) or 1.544 MHz (on T1).
meSlipsReceived
A frame is repeated when the frequency of the receive route clock is less
than the frequency of the receive system interface working clock based
statsInfoFramingErr
or
The framing error count indicates that a FAS (Frame Alignment Signal)
word has been received with an error.
statsInfoCodeViolati
ons
The code violations count indicates that an encoding error on the PCM
line has been detected.
statsInfoCRCErrors
The CRC errors count is incremented when a multiframe has been

received with a CRC error.
statsInfoE-BitError
The E-Bit error count gives information about the outgoing transmit PCM
line if the E-bits are used by the remote end for submultiframe error
indication. Incrementing is only possible in the multiframe synchronous
state.
Due to signaling requirements, the E-bits of frame 13 and frame 15 of the
CRC multiframe can be used to indicate an error in a received
submultiframes:
CRC error : E = 0
275
PRI E&M Statistics
Table 249: E&M Statistics Displayed (Continued)

Statistic
Description
statsInfoBlockError
The Block Error count is incremented once per multiframe if a multirame

has been received with a CRC error or a bad frame alignment has been
detected.
To reset the statistics:

1.
In the ex1Pri_1MIB, set the statsInfoResetStats variable to 10: resetStats.

ex1Pri_1.statsInfoResetStats=10
276
SIP Parameters
H A P T E R
24
SIP Gateways
This chapter describes how to add and remove SIP gateways in the Mediatrix unit.
SIP Gateways Configuration

Multiple SIP gateways may be used for a number of reasons, such as:
Redirecting ISDN calls to different SIP servers depending on the call.

Hunt calls across several gateways.
Adding a SIP gateway triggers a warning message if the total number of registrations configured reached the
defined limit. See Number of Registrations on page 295 for more details.
There are two types of SIP Gateways:
Trunk Gateway
A trunk gateway is generally used when the device is connected to a PBX or phone network; it can also be
used when connected to terminal equipment while using a SIP trunk to a SIP server.
Figure 121: Trunk Gateway
The characteristics of a trunk gateway are as follows:
It works with endpoint and user (unit) registrations.
A listening port allows for dialogs to be established by any peer.
Connections can be persistent or not, depending on the type of transport: UDP and TCP
transports are limited to non-persistent connections; TLS establishes persistent connections to
the outbound proxy, home domain proxy and registrar and non-persistent connections to other
targets.
The call router shows a single SIP source/destination for the gateway.
SIP dialogs are established independently of each other, depending on the SIP keep alive
mechanism defined. See Keep Alive on page 287 for more details on the SIP keep alive
mechanism.
When the destination is a FQDN, each SIP transaction is possibly sent to a different IP address,
depending on the DNS query result. A trunk gateway assumes all SIP servers identified by a
single FQDN have a synchronized state.
Endpoint Gateway
An endpoint gateway is generally used when the device is connected to terminal equipment where each
endpoint has a separate SIP connection to the SIP server.
279
Chapter 24 - SIP Gateways
Figure 122: Endpoint Gateway
An endpoint gateway is a type of gateway introduced to satisfy use cases with failover/failback based on
registrations for a single user.
The characteristics of an endpoint gateway are as follows:
It works with endpoint registrations only (no unit registrations can be associated to an endpoint
gateway).
SIP dialogs for a given SIP user can only be established once the user is registered to the
server.
It creates a persistent connection for each SIP user. This connection allows for dialogs to be
established only by the server to which the user is registered. No listening port allows a
connection to be established by a peer.
Failover/failback to another server requires the SIP user to register on that server prior to
establishing a dialog.
The call router and gateway status tables show an instance of the gateway for each user of the
gateway.
To configure multiple SIP gateways:

1.
In the web interface, click the SIP link, then the Gateways sub-link.
Figure 123: SIP Gateways Web Page
You can add a new gateway by clicking the

5 gateways.
button. The Mediatrix unit supports a maximum of
You can delete an existing gateway by clicking the

2.
button.
If you are adding a new gateway, enter its name in the Name field.
The Dgw v2.0 Application supports only alphanumeric characters, -, and _.
3.
Select the type of SIP gateway to be configured in the Type drop-down menu.
The default value is Trunk; select Endpoint for an endpoint gateway.
4.
Select the network interface on which the gateway listens for incoming SIP traffic in the Signaling
Network drop-down menu.
This value applies to all transports (e.g., UDP, TCP, etc.).
The LAN interface may be used as a SIP gateway to be bound on the LAN. However, there is no
routing between the LAN and the uplink interface.
280
5.
Define the list of networks (separated by ",") to use for the media (voice, fax, etc.) stream in the
Media Networks field.
You can use the Media Networks Suggestion columns drop-down menu to select between
suggested values, if any.
The value must match one of the "InterarfaceName" values in the "NetworkInterfacesStatus" table
of the BNI service. The order in the list defines the priority.
When the media stream is negotiated, the following rules apply:
If the list of media networks is empty, the Mediatrix unit uses the IP address of the
network defined in the Signaling Network drop-down menu.
Only active networks are used.
Only the first active network of an IP address family (IPv4, IPv6) is used. All
subsequent networks of the same IP family are ignored.
Note: When generating an offer and multiple networks are available for the media, ANAT grouping (RFC
4091) is automatically enabled. When generating an answer, the ANAT grouping state is detected form the
offer.
6.
If the gateway type is set to "Trunk", set the SIP port on which the gateway listens for incoming
unsecure SIP traffic in the Port field.
This is used only when the UDP and/or TCP transports are enabled.
If two or more SIP gateways use the same port, only the first SIP gateway starts correctly. The
others are in error and not started. The SIP gateway is also in error and not started if the port is
already used.
The default value is 0. If you set the port to 0, the default SIP port 5060 is used.
Note: The port 0 is the equivalent to the well known port, which is 5060 in SIP. Using 0 and 5060 is not
the same. At the SIP packets level, if you set the port to 0, it will not be present in the SIP packet. If you set
the port to 5060, it will be present in the SIP packet. For example: 23@test.com if the port is 0 and
23@test.com:5060 if the port is 5060.
Note: When the gateway type is set to "Endpoint" the SIP Port and Secure Port have no effect and must
be set to 0 since each user has a unique UDP/TCP port.
For "Endpoint" gateways, only the base port can be set with the variable Persistent Base Port in the SIP
Transport configuration web interface.
7.
If the gateway type is set to "Trunk", set the SIP port on which the gateway listens for incoming
secure SIP traffic in the Secure Port field.
This is used only when the TLS transport is enabled.
The default value is 0. If you set the port to 0, the default secure SIP port 5061 is used.
Note: The port 0 is the equivalent to the well known port, which is 5061 in SIP for TLS. Using 0 and 5061
is not the same. At the SIP packets level, if you set the port to 0, it will not be present in the SIP packet. If
you set the port to 5061, it will be present in the SIP packet. For example: 23@test.com if the port is 0 and
23@test.com:5061 if the port is 5061.
8.

The state of the SIP gateways is displayed in the SIP Gateway Status section.
Table 250: SIP Gateway States
State
Ready
Description
The gateway is ready to make and receive calls.
281
Chapter 24 - SIP Gateways
Table 250: SIP Gateway States

State
282
Description
Cannot start, port already in

use
The gateway cannot open its IP port because the port is

already used by another service. This generally occurs when
the administrator adds a new gateway but forgets to configure
a different IP port.
Network down
The SIP gateway is not started or the network interface on

which the SIP gateway is associated does not have an IP
address.
Restarting
The SIP gateway cannot make or receive calls while it is

restarting.
Waiting for time

synchronization
The gateway is started but it cannot open its SIP TLS port
because the real-time clock is not synchronized. This generally
occurs when the SNTP server is not set or is unreachable.
Server unreachable
The gateway is started but it cannot make and receive calls

because the SIP server is unreachable. This state is only
reported when a KeepAlive mechanism is used.
Unregistered
Indicates some registrations that are mandatory for this

gateway failed. See Unregistered Unit Behaviour on page 299
for more details.
Invalid Config
The gateway cannot start due to an inconsistent configuration.
H A P T E R
25
SIP Servers
This chapter describes how to configure the SIP server parameters of the Mediatrix unit.
Standards Supported
RFC 2543: SIP: Session Initiation Protocol
RFC 3261: The Session Initiation Protocol (SIP)
RFC 3903: Session Initiation Protocol (SIP) Extension for

Event State Publication
3GPP TS 24.229: IP multimedia call control protocol based on

SIP and SDP.a
a. Supported except for sections 5.1.1.3, 5.1.1.4.3, 5.1.6, 7 and 8.
It describes the following:
How to define the SIP servers IP information.

How to define the SIP gateways IP information.
Introduction
The Mediatrix unit uses the following types of servers:
Table 251: SIP Servers
Server
Description
Registrar Server
Accepts REGISTER requests and places the information it receives in those

requests into the location service for the domain it handles.
Proxy Server
An intermediary program that acts as both a server and a client for the purpose of
making requests on behalf of other clients. A proxy server primarily plays the role
of routing, which means its job is to ensure that a request is passed on to another
entity that can further process the request. Proxies are also useful for enforcing
policy and for firewall traversal. A proxy interprets, and, if necessary, rewrites
parts of a request message before forwarding it.
Outbound Proxy Server An intermediary entity that acts as both a server and a client for the purpose of
making requests on behalf of other clients. The outbound proxy receives all
outbound traffic and forwards it. Incoming traffic may or may not go through the
outbound proxy. The outbound proxys address is never used in the SIP packets,
it is only used as a physical network destination for the packets.
When the outbound proxy is enabled, the proxy is still used to create the To and
From headers, but the packets are physically sent to the outbound proxy.
SIP Outbound Proxy (From RFC 3261)

A proxy that receives requests from a client, even though it may not be the server resolved by the
Request-URI. Typically, a user agent is manually configured with an outbound proxy.
When enabled, the initial route for all SIP requests contains the outbound proxy address, suffixed with
the loose routing parameter lr. The Request-URI still contains the home domain proxy address.
Requests are directed to the first route (the outbound proxy).
283
Chapter 25 - SIP Servers
TLS Persistent Connections Status
Table 251: SIP Servers (Continued)

Server
Description
Messaging Server Host A Messaging system host is a server that accepts MWI SUBSCRIBE requests
and places the information it receives in those requests into the location service
for the domain it handles.
TLS Persistent Connections Status

The TLS Persistent Connections Status table allows you to browse the status of the TLS persistent
connections of the Mediatrix unit. These connections are associated with the SIP servers (outbound proxy,
registrar and home domain proxy). Note that this section is not displayed if there is no information to show.
TLS connection is currently only supported with Trunk gateway.
Figure 124: SIP TLS Persistent Connections Status Section
The following information is available:

Table 252: TLS Persistent Connection Parameters
Parameter
Description
Gateway
The SIP gateway used to register.
Local Port
Local port used by the TLS persistent connection.
Remote Host
The remote host used to establish the TLS persistent connection. The remote host
can be a host name or an IP address of the proxy, outbound proxy or registrar.
Remote IP Address The resolved IP address of the remote host used to establish the TLS persistent
connection.
Status
The current state of the TLS persistent connection.
Up: The TLS connection is established.
Down: The TLS connection is not established.
SIP Servers Configuration

This section describes how to configure the IP address and port number of the SIP servers.
If any of the SIP servers parameters corresponds to a domain name that is bound to a SRV record, the
corresponding port must be set to 0 for the unit to perform DNS requests of type SRV (as per RFC 3263).
Otherwise, the unit will not use DNS SRV requests, but will rather use only requests of type A because it does
not need to be specified which port to use.
284
Multiple SIP Gateways
To set the SIP servers configuration:

1.
In the web interface, click the SIP link, then the Configuration sub-link.
Figure 125: SIP Servers Web Page
2
4
3
5
2.
Enter the SIP registrar server static IP address or domain name and port number in the Registrar
Host field.
You must enter the information as IP address:Port number. For instance:
192.168.0.5:5060
3.
Enter the SIP Proxy server static IP address or domain name and port number in the Proxy Host
field.
192.168.0.5:5060
4.
Enter the Messaging system host static IP address or domain name and port number in the
Messaging Server Host field.
If the host corresponds to a domain name that is bound to a SRV record, the port must be set to 0
for the unit to perform DNS SRV queries; otherwise only type A record lookups will be used.
You can define whether or not an endpoint needs to subscribe to a messaging system in Endpoints
Registration on page 291.
5.
Enter the SIP outbound proxy server static IP address or domain name and port number in the
Outbound Proxy Host field.
The outbound proxy is enabled if the IP address is valid (i.e., not 0.0.0.0:0). Setting the address to
0.0.0.0:0 or leaving the field empty disables the outbound proxy.
Note: The Endpoint gateway can only have a single next hop. If no outbound proxy is set then, if used, the
proxy host, the registrar host and the messaging server host must be set to the same FQDN or IP Address.
If the hosts are not set to the same URL or IP address, the SIP Gateway State will be set to Invalid Config.
See Table 250 on page 281 for the list of SIP Gateway states.
6.

The Mediatrix unit allows you to have multiple SIP gateways (interfaces). You can configure each SIP gateway
to register to a specific registrar. You can also configure each SIP gateway to send all requests to an outbound
proxy. See Chapter 24 - SIP Gateways on page 279 for more details.
SIP Gateway Specific Registrar Servers

This section allows you to define whether the available SIP gateways use the default registrar server or rather
use a specific registrar server.
285
To set specific registrars servers information:

1.
In the Registrar Servers section of the Servers page, select whether or not a SIP gateway uses a
specific registrar server in the Gateway Specific drop-down menu.
If you select No, the SIP gateway uses the server information as set in the SIP Default Servers
section.
Figure 126: SIP Servers Specific Registrar Section
1
2.
Enter the IP address or domain name and port number of the registrar server currently used by the
registration in the Registrar Host field.
192.168.0.5:5060
3.
If you do not need to set other parameters, do one of the following:
To save your settings without refreshing the registration, click Submit.
To save your settings and refresh the registration now, click Submit & Refresh
Registration.
SIP Gateway Specific Messaging Servers

This section allows you to define whether the available SIP gateways use the default proxy and outbound
proxy server or rather use specific servers.
To set specific proxy servers information:

1.
In the Messaging Servers section of the Servers page, select whether or not a SIP gateway uses a
specific proxy and outbound proxy server in the Gateway Specific drop-down menu.
If you select No, the SIP gateway uses the server information as set in the SIP Default Servers and
Messaging Subscription (Messaging Subscription on page 353) sections.
Figure 127: SIP Servers Messaging Section
1
2.
Enter the IP address or domain name and port number of the messaging server currently used by
the registration in the Proxy Host field.
192.168.0.5:5060
3.
Enter the IP address or domain name and port number of the outbound proxy server currently used
by the registration in the Outbound Proxy Host field.
192.168.0.5:5060
4.
286
Registration.
SIP Gateway Specific Proxy Servers

This section allows you to define whether the available SIP gateways use the default proxy and outbound
proxy server or rather use specific servers.
To set specific proxy servers information:

1.
In the Proxy Servers section of the Servers page, select whether or not a SIP gateway uses a
specific proxy and outbound proxy server in the Gateway Specific drop-down menu.
If you select No, the SIP gateway uses the server information as set in the SIP Default Servers
section.
Figure 128: SIP Servers Specific Proxy Section
1
2.
Enter the IP address or domain name and port number of the proxy server currently used by the
registration in the Proxy Host field.
192.168.0.5:5060
Note: The Endpoint gateway can only have a single next hop. If no outbound proxy is set then, if used, the
proxy host, the registrar host and the messaging server host must be set to the same FQDN or IP Address.
If the hosts are not set to the same URL or IP address, the SIP Gateway State will be set to Invalid Config.
See Table 250 on page 281 for the list of SIP Gateway states.
3.
Enter the IP address or domain name and port number of the outbound proxy server currently used
by the registration in the Outbound Proxy Host field.
192.168.0.5:5060
4.
Registration.
Keep Alive
You can select the method used to perform the SIP keep alive mechanism. With this mechanism, the Mediatrix
unit sends messages periodically to the server to ensure that it can still be reached. The SIP keep alive
mechanism is only supported with Trunk gateways.
287
To use the SIP keep alive mechanism:

1.
In the Keep Alive section of the Servers page, select the keep alive method to use in the Keep Alive
Method drop-down menu.
Figure 129: Keep Alive Section
1
2
3
Table 253: Keep Alive Parameters

Parameter
Description
None
No keep alive is performed.
SipOptions
SIP OPTIONS are sent periodically for each gateway to the

corresponding server. Any response received from the
server means that it can be reached. No additional
processing is performed on the response. If no response is
received after the retransmission timer expires
(configurable via the Transmission Timeout field in SIP
Interop on page 316), the gateway considers the server as
unreachable. In this case, any call attempt through the
gateway is refused. SIP OPTIONS are still sent when the
server cannot be reached and as soon as it can be reached
again, new calls are allowed.
Ping
A Ping is sent periodically for each gateway to the

corresponding server. The response received from the
server means that it is reachable. If no response is received
after the retransmission timer expires (configurable via the
Transmission Timeout field in SIP Interop on page 316),
the gateway considers the server as unreachable. In this
case, any call attempt through the gateway is refused. The
Pings are still sent when the server is unreachable and as
soon as it becomes reachable again, new calls are allowed.
On Endpoint gateways, the keep alive mechanism is always considered to be None.

2.
Set the interval, in seconds, at which SIP Keep Alive requests using SIP OPTIONS or Ping are sent
to verify the server status in the Keep Alive Interval field.
3.
Select the behaviour of the device when performing the keep alive action in the Keep Alive
Destination drop-down menu.
Table 254: SIP Keep Alive Destination Parameters
Parameter
4.
288
Description
First SIP Destination
Performs the keep alive action through the first SIP

destination. This corresponds to the outbound proxy host
when specified, otherwise it is the proxy host.
Alternate Destination
Performs the keep alive action through the alternate

destination target (see SIP Gateway Specific Keep Alive
Destinations on page 289 for more details).
Outbound Proxy Loose Router Configuration
SIP Gateway Specific Keep Alive Destinations

This section allows you to override the default Keep Alive destination alternate target when the Keep Alive
Destination drop-down menu is set to Alternate Destination (see Keep Alive on page 287 for more details).
To set specific keep alive destinations:

1.
In the Keep Alive Destinations section of the Servers page, set the Alternate destination target
server FQDN and port for a specific SIP gateway in the default field.
192.168.0.5:5060
Figure 130: SIP Servers Specific Keep alive Targets

1
2.

Standards Supported
RFC 3261: SIP: Session Initiation Protocol, section 6

by using the CLI
You can specify the type of routing of the outbound proxy configured in SIP Servers Configuration on
page 284.
Default configurations that apply to all the endpoints of the Mediatrix unit.
You can define specific configurations for each endpoint in your Mediatrix unit. For instance, you
could enable a codec for all the endpoints of the Mediatrix unit and use the specific configuration
parameters to disable this same codec on one specific endpoint.
Using one or more specific parameter usually requires that you enable an override variable and set the specific
configuration you want to apply.
Loose Router
A proxy is said to be loose routing if it follows the procedures defined in the RFC 3261 specification
(section 6) for processing of the Route header field. These procedures separate the destination of the
request (present in the Request-URI) from the set of proxies that need to be visited along the way
(present in the Route header field). A proxy compliant to these mechanisms is also known as a loose
router.
289
The following types are available:

Table 255: Outbound Proxy Router Status
Type
Description
LooseRouter This is the most current method for SIP routing, as per RFC 3261, and will become the
standard behaviour once RFC 3261 compliance is achieved. See Introduction on
page 283 for details.
StrictRouter
Pre-RFC 3261, RFC 2543 compatible SIP routing.

The initial route for all SIP requests contains the home domain proxy address (the RequestURI). Requests are directed to the outbound proxy.
In other words, the Request-URI is constructed as usual, using the home domain proxy and
the user name, but is used in the route set. The Request-URI is filled with the outbound
proxy address.
NoRouteHea Removes the route header from all SIP packets sent to an outbound proxy. This does not
der
modify persistent TLS connection headers.
Note: The Router header will not be removed from the SIP packets if the unit is configured
to use the TLS Fallback feature. This feature requires the information of the SIP Outbound
Proxy in the SIP packet to work correctly.
To set the outbound proxy router status:

1.
In the sipEpMIB, set the defaultProxyOutboundType variable to the proper value.

sipEp.defaultProxyOutboundType="Value"

Table 256: Outbound Proxy Router Values
Value
2.
Meaning
100
LooseRouter
200
StrictRouter
300
NoRouteHeader
If you want to set a different routing type for one or more SIP gateways, set the following variables:
gwSpecificproxyEnableConfig variable for the specific SIP gateway you want to
configure to enable.
gwSpecificProxyOutboundType variable for the specific SIP gateway you want to
configure to the proper value.

You can also use the following lines in the CLI or a configuration script:
sipEp.gwSpecificProxy.EnableConfig[GatewayName="default"]="1"
sipEp.gwSpecificProxy.OutboundType[GatewayName="Specific_Gateway"]="Value"
where:
290
Specific_Gateway is the name of the SIP gateway you want to configure.
Value is the refresh router status as defined in Step 1.
H A P T E R
26
SIP Registration
This chapter describes how to configure the registration parameters of the Mediatrix unit.
Standards Supported
RFC 3261: The Session Initiation Protocol (SIP)
RFC 3863: Presence Information Data Format (PIDF)
RFC 3903: Session Initiation Protocol (SIP) Extension for

Event State Publication
Endpoints Registration
Each endpoint of the Mediatrix unit has its own registration information. You can set information for each
endpoint such as its telephone number and friendly name.
Adding an endpoint registration triggers a warning message if the total number of registrations configured
reached the defined limit. See Number of Registrations on page 295 for more details.
To set endpoints registration information:

1.
In the web interface, click the SIP link, then the Registrations sub-link.
Figure 131: SIP Registrations Web Page
2.
In the Endpoints Registration and Subscription section of the Registrations page, enter a user name
for each endpoint in the User Name column.
The user name (such as a telephone number) uniquely identifies this endpoint in the domain. It is
used to create the Contact and From headers. The From header carries the permanent location (IP
address, home domain) where the endpoint is located. The Contact header carries the current
location (IP address) where the endpoint can be reached.
291
Endpoints Registration
Contacts are registered to the registrar. This enables callers to be redirected to the endpoints
current location.
Note: If two or more endpoints have the same user name, a single registration request and/or subscription
request will be performed under that user name.
3.
Enter another name for each endpoint in the Friendly Name column.
This is a friendly name for the endpoint. It contains a descriptive version of the URI and is intended
to be displayed to a user interface.
4.
Define whether or not the endpoint registration needs to register to the registrar in the Register
column.
An endpoint configured to register (set to Enable) will become unavailable for calls from or to SIP
when not registered.
You can define the behaviour of an endpoint when it becomes unavailable in the
defaultRegistrationUnregisteredBehavior MIB variable.
5.
Define whether or not the endpoint needs to subscribe to a messaging system in the Messaging
drop-down menu.
The current state of the subscription is displayed in the Endpoints Messaging Subscription Status
table.
Table 257: MWI Subscription State
State
Description
Unsubscribed
The unit/endpoint is not subscribed and never tries to subscribe. This case
occurs if the network interface used by the SIP gateway is not up or the unit/
endpoint is locked.
Subscribing
The subscription is currently trying to subscribe.
Subscribed
The subscription is successfully subscribed.
Refreshing
The subscription is trying to refresh.
Unreachable
The last subscription attempt failed because the messaging server is

unreachable.
AuthFailed
The last subscription attempt failed because authentication was not

successful.
Rejected
The last subscription attempt failed because the messaging server rejects
the subscription.
ConfigError
The last subscription attempt failed because it was badly configured. Check
if the username and the messaging host are not empty.
InvalidResponse The received 200 OK response contact does not match the contact of the
messaging server, or the 200 OK response for an unsubscribe contains a
contact.
You can enter the address of the Messaging server in SIP Servers Configuration on page 284.
6.
Select on which SIP gateway the user configuration is applied in the Gateway Name drop-down
menu.
You must have SIP gateways already defined. See Chapter 24 - SIP Gateways on page 279 for
more details. If you select all, the configuration applies to all gateways available.
7.
292
To save your settings and refresh the registration now, click Submit & Refresh.
Unit Registration
Contact Domain

by using the CLI
You can set the host part of the SIP contact field. If an empty string is specified, the listening IP address is
used.
To set the contact domain:

1.
In the sipEp MIB, set the UserAgentContactDomain variable in the UserAgent table or,
2.
In the CLI or a configuration script, use:

SipEp.UserAgent[EpId=index).ContactDomain=value
Accept Language
The AcceptLanguage parameter allows a user to indicate the preferred language that will be used for
displayed phrases, session descriptions, status responses carried as message bodies in the response. It is
used to fill the Accept-Language SIP header field.
You can configure the parameter by:
using a MIB browser

using the CLI
creating a configuration script containing the configuration variables
To set AcceptLanguage:
1.
In the SipEp Mib. set the AcceptLanguage variable in UserAgent table or,
2.
In the CLI or a configuration script use

SipEp.UserAgent[index-value].AcceptLanguage=<value>
Index is the endpoint name
Unit Registration
Unit registration is used to register a contact not directly related to endpoints. This is generally used to indicate
to a registrar the IP location of the Mediatrix unit when it is used as a gateway.
Adding a unit registration triggers a warning message if the total number of registrations configured reached
the defined limit. See Number of Registrations on page 295 for more details.
To set unit registration information:

1.
In the Unit Registration section of the Registrations page, enter a user name in the User Name
column.
Figure 132: SIP Registrations Unit Registration Section
1
293
Registration Configuration
The user name (such as a telephone number) uniquely identifies this user in the domain.
You can add a new user by clicking the
button.
You can delete an existing user by clicking the

2.
button.
Select on which SIP gateway the user configuration is applied in the Gateway Name drop-down
menu.
You must have SIP gateways already defined. See Chapter 24 - SIP Gateways on page 279 for
more details. If you select all, the configuration applies to all gateways available.
3.
Registration Configuration
This section allows you to define registration refresh parameters.
See Additional Registration Refresh Parameters on page 296 for more registration parameters.
To set the registration configuration:

1.
In the Registration Configuration section of the Registrations page, set the Default Registration
Refresh Time field with the time, in seconds, at which a registered unit begins updating its
registration before the registration expiration.
Figure 133: SIP Registrations Registration Configuration Section
1
3
In SIP, a registration is valid for a period of time defined by the registrar. Once a unit is registered,
the SIP protocol requires the User Agent to refresh this registration before the registration expires.
Typically, this re-registration must be completed before the ongoing registration expires, so that the
User Agent's registration state does not change (i.e., remains 'registered').
For instance, if the parameter is set to 43 and the registration lasts one hour, the unit will send new
REGISTER requests 59 minutes and 17 seconds after receiving the registration acknowledgement
(43 seconds before the unit becomes unregistered).
Note: Normally, the Mediatrix unit cannot make or receive calls until the REGISTER has completed
successfully. Because the timeout for a SIP transaction in UDP is 32 seconds, it is possible to have an
ongoing re-REGISTER transaction at the same moment that the registration itself expires. This could
happen if the Default Registration Refresh Time field is set to a value lower than 32.
In that case, the user agent becomes unregistered, and will become registered again only when the reREGISTER request is answered with a positive response from the server. See Gateway Specific
Registration Retry Time on page 298 for a workaround if the unit cannot make calls during that period.
Setting this parameter to 0 means that the User Agent will fall into the 'unregistered' state BEFORE
sending the re-REGISTER requests.
This value MUST be lower than the value of the "expires" of the contact in the 200 OK response to
the REGISTER, otherwise the unit rapidly sends REGISTER requests continuously.
You can also set a different registration refresh time for one or more SIP gateways by using the MIB
parameters of the Mediatrix unit. See Registration Refresh on page 297 for more details.
2.
294
Set the Proposed Expiration Value In Registration field with the suggested expiration delay, in
seconds, of a contact in the REGISTER request.
Number of Registrations
The SIP protocol allows an entity to specify the expires parameter of a contact in a REGISTER
request. The server can return this expires parameter in the 200 OK response or select another
expires. In the REGISTER request, the expires is a suggestion the entity makes.
The expires parameter indicates how long, in seconds, the user agent would like the binding to be
valid.
Available values are from 1 s to 86,400 s (one day).
This value does not modify the delay before a re-REGISTER.
The delay is the expires of the contact in the 200 OK response to the REGISTER
request minus the value set in the Default Registration Refresh Time field.
If the expires of the contact in the 200 OK response to the REGISTER is not present
or not properly formatted, then the delay is the default registration proposed expiration
value minus the value set in the Default Registration Refresh Time field.
Setting the parameter to 0 disables the expiration suggestion.

You can also set a different expiration delay for one or more SIP gateways by using the MIB
parameters of the Mediatrix unit. See Registration Expiration on page 297 for more details.
3.
Set the Default Expiration Value in Registration field with the default registration expiration, in
seconds.
This value is used when the contact in a registration response contains no expires or the expires
is badly formatted. In this case, the delay before a re-REGISTER is the value set in this field minus
the value set in the Default Registration Refresh Time field (Step 1).
You can also set a different expiration value in registration for one or more SIP gateways by using
the MIB parameters of the Mediatrix unit. See Expiration Value in Registration on page 298 for
more details.
4.
Number of Registrations
The Mediatrix unit limits the total number of registrations to 100. The total number of registrations is the sum
of all the endpoints and gateways (SIP Gateways Configuration on page 279) pairs. The Mediatrix unit
supports a maximum of 5 gateways. An endpoint configured with "All" gateways generates as many pairs as
the number of gateways. In a setup with 3 gateways, one endpoint configured with "All" as the gateway name
counts for 3 in the total number of registrations.
The registrations are enabled gateway by gateway until the limit is reached. Endpoints Registrations are used
first, then Unit Registrations. The remaining registrations are not registered and do not appear in the status
table. If you click the Submit And Refresh button and the configured number of registrations exceeds the
defined limit, a warning is displayed on the web interface (as well as in the CLI and SNMP interfaces) and a
syslog notify (Level Error) is sent.
Adding a gateway or an endpoint triggers a warning message if the total number of registrations configured
reached the defined limit.
Lets suppose for instance that we have the current SIP Gateways configuration and the following SIP
Registration configuration:
Figure 134: Example, Gateway Configuration
295
Additional Registration Refresh Parameters
Figure 135: Example, Registrations Configuration
The following table describes how to compute the total number of registrations for this example:
Table 258: Number of Registrations Example
Parameter
Setting
Nb of Registrations
Endpoint Registration 1 in
Figure 135
Gateway Name set to alla
Endpoint Registration 2 in
Figure 135
Gateway Name set to gw2
Unit Registration 1 in Figure 135
Gateway Name set to all
Gateway Name set to all
Gateway Name set to gw1
Gateway Name set to default 1
Total Number of registrations
12
a. When the Gateway Name is set to all, this must be multiplied by the number of gateways set in Figure 134.
In this example, there are 3 gateways set.


by using the CLI
Default Registration Retry Time

You can configure the interval in seconds (s) on which a failed registration is retried.
This variable defines the time, relative to the failure of the registration, at which the device retries the
registration.
To specify the default registration retry time value:

1.
In the sipEpMIB, locate the registrationGroup folder.
2.
Set the DefaultRegistrationRetryTime variable with the desired interval value.

sipEp.DefaultRegistrationRetryTime="Value"
where Value may be between 1 and 86400 seconds.
296

Registration Refresh
You can set the default registration refresh time in the web page (Registration Configuration on page 294),
but you can also set a different registration refresh time for one or more SIP gateways.
To set registration refresh parameters:

1.
2.
If you want to set a different registration refresh time for one or more SIP gateways, set the following
variables:
gwSpecificRegistrationEnableConfig variable for the specific SIP gateway you
want to configure to enable.
gwSpecificRegistrationRefreshTime variable for the specific SIP gateway you want
to configure to the proper value.

sipEp.gwSpecificRegistration.EnableConfig[GatewayName="Specific_Gateway"]="1"
sipEp.gwSpecificRegistration.RefreshTime[GatewayName="Specific_Gateway"]="Value"
where:
Value is the refresh time value.
Registration Expiration
You can set the default registration proposed expiration value in the web page (Registration Configuration
on page 294), but you can also set a different registration refresh time for one or more SIP gateways.
To configure the registration expiration:

1.
2.
If you want to set a different registration refresh time for one or more SIP gateways, set the following
variables:
gwSpecificRegistrationProposedExpirationValue variable for the specific SIP
gateway you want to configure to the proper value.

sipEp.gwSpecificRegistration.ProposedExpirationValue[GatewayName="Specific_Gatew
ay"]="Value"
where:
Value is the expiration delay value.
This value does not modify the time before a re-REGISTER.
297
The delay is the expires of the contact in the 200 OK response to the REGISTER
request minus the value set in the gwSpecificRegistrationRefreshTime parameter.
If the expires of the contact in the 200 OK response to the REGISTER is not present
or not properly formatted, then the delay is the default registration proposed expiration
value minus the value set in the gwSpecificRegistrationRefreshTime parameter.
Expiration Value in Registration

You can set the default expiration value in registration in the web page (Registration Configuration on
page 294), but you can also set a different expiration value in registration for one or more SIP gateways.
This value is used when the contact in a registration response contains no expires or the expires is badly
formatted. In this case, the delay before a re-REGISTER is the value set in this field minus the value set in the
in the 'RefreshTime' variable (Registration Refresh on page 297).
To configure the expiration value in registration for a specific gateway:

1.
2.
To set a different expiration value in registration for one or more SIP gateways, set the following
variables:
gwSpecificRegistrationExpirationValue variable for the specific SIP gateway you
want to configure to the proper value.

3.
To set a different expiration value in registration for one or more SIP gateways, put the following
lines in the configuration script:
sipEp.gwSpecificRegistration.ExpirationValue[GatewayName="Specific_Gateway"]="Va
lue"
where:
Value is the expiration value in registration value.
Gateway Specific Registration Retry Time

You can set a different Registration Retry Time for one or more SIP gateways.
This variable defines the time, relative to the failure of the registration, at which the SIP gateway retries the
registration.
To specify the registration retry time value for a specific gateway:

1.
2.
To set a different registration retry time for one or more SIP gateways, set the following variables:
gwSpecificRegistrationRetryTime variable for the specific SIP gateway you want to

3.
To set a different expiration value in registration for one or more SIP gateways, put the following
lines in the configuration script:
sipEp.gwSpecificRegistrationRetryTime[GatewayName="Specific_Gateway"]="Value"
where:
298
Value is the expiration value in registration retry time.
Unregistered Endpoint Behaviour

You can specify whether an endpoint should remain enabled or not when not registered. This is useful if you
want your users to be able to make calls even if the endpoint is not registered with a SIP server.
The following values are supported:
Table 259: Unregistered Endpoint Behaviour Parameters
Value
Description
disablePort
When the endpoint is not registered, it is disabled. The user cannot make or receive calls.
Picking up the handset yields a fast busy tone, and incoming INVITEs receive a 403
Forbidden response.
enablePort
When the endpoint is not registered, it is still enabled. The user can receive and initiate
outgoing calls. Note that because the endpoint is not registered with a registrar, its public
address is not available to the outside world; the endpoint will most likely be unreachable
except through direct IP calling.
To specify unregistered endpoint behaviour:

1.
2.
Set the defaultRegistrationUnregisteredBehavior variable.

sipEp.defaultRegistrationUnregisteredBehavior="Value"
where Value may be as follows:.

Table 260: Unregistered Endpoint Behaviour Values
Value
3.
Meaning
disablePort
enablePort
If you want to set a different behaviour for one or more SIP gateways, set the following variables:
gwSpecificRegistrationUnregisteredBehavior variable for the specific SIP
gateway you want to configure to the proper value.

sipEp.gwSpecificRegistration.UnregisteredBehavior[GatewayName="Specific_Gateway"
]="Value"
where:
Value is one of the values described in Step 2.
Unregistered Unit Behaviour

You can specify whether the SIP gateway state should be affected or not by the unit registrations state.
299

Table 261: Unregistered Unit Behaviour Parameters
Value
Description
NoEffect
The unit registrations state has no effect on the SIP gateway state.
DisableGate
way
The SIP gateway goes in the 'unregistered' state when all unit registrations are not in the
'registered' state. The 'unregistered' state indicates some registrations that are mandatory
for this gateway failed.
To specify unregistered unit behaviour:

1.
2.
Set the defaultUnitRegistrationUnregisteredBehavior variable.

sipEp.defaultUnitRegistrationUnregisteredBehavior="Value"

Table 262: Unregistered Unit Behaviour Values
Value
Meaning
100
NoEffect
200
DisableGateway
Behaviour on Initial-Registration Reception

You can configure the behaviour of the Mediatrix unit upon reception of a 380 or 504 carrying an XML body
with a specified 'initial-registration' action.
Table 263: Behaviour on Initial-Registration Reception Parameters
Value
Description
NoRegistration
No registration refresh are sent upon reception of the message.
EndpointRegistration
Registration refresh of the endpoint associated with the call is sent upon
reception of the message.
UnitRegistration
Registration refresh of all the usernames configured as 'unit registration'

are sent upon reception of the message.
UnitAndEndpointRegistration Registration refresh of the endpoint associated with the call and of all the
usernames configured as 'unit registration' are sent upon reception of the
message.
To specify the behaviour on Initial-Registration reception:

1.
2.
Set the behaviorOnInitialRegistrationReception variable with the proper behaviour.

sipEp.behaviorOnInitialRegistrationReception="Value"
300
SIP User Agent Header

Table 264: Behaviour on Initial-Registration Reception Values
Value
Meaning
100
NoRegistration
200
EndpointRegistration
300
UnitRegistration
400
UnitAndEndpointRegistration
If the registration(s) succeed, then the call is re-attempted.

If the registration(s) fail, then the call is terminated.
3.
Set the registrationDelayOnInitialRegistrationReception variable with the registration

delay, in milliseconds, on Initial-Registration Reception.
This variable configures the time interval between the unregistration confirmation (or final response)
and the registration attempt that follows.
This variable is only used when behaviorOnInitialRegistrationReception is configured to a
value other than 'NoRegistration'.
Note: This variable only applies on registration refresh triggered by the

behaviorOnInitialRegistrationReception feature.
sipEp.registrationDelayOnInitialRegistrationReception="Value"
Registration Delay Value

The quality of calls may be altered if a large quantity of registrations, more than 100, is requested at the same
time. To avoid this situation, you can configure the maximum number of seconds that the system uses to apply
a random algorithm, which is used to determine a delay before requesting a user registration or an endpoint
registration.
When the value is 0, the request registration is done immediately.
Note: The random algorithm applies individually to all registrations, meaning registrations order may not
follow their corresponding index.
To specify the registration delay value:

1.
In the sipEpMIB, set the interopRegistrationDelayValue variable with the proper delay value.
sipEp.interopRegistrationDelayValue="Value"
where Value may be between 0 and 600 seconds.


by using the CLI
301
The User-Agent header field contains information about the user agent client originating the request. For
instance, the information of the User-Agent header could be something like the following:
User-Agent: Softphone Beta1.5
You can specify whether or not the Mediatrix unit sends this information when establishing a communication.
To enable sending the SIP User Agent header:

1.
In the sipEpMIB, set the interopSendUAHeaderEnable variable to enable.

sipEp.interopSendUaHeaderEnable="1"
302
H A P T E R
27
SIP Authentication
This chapter describes how to configure authentication parameters of the Mediatrix unit.
Standards Supported
Basic and Digest authentication as per RFC 3261
Caution: The SIP > Authentication page is not accessible if you have the User or Observer access right.
See Users on page 607 for more details.
Authentication Configuration
Authentication information allows you to add some level of security to the Mediatrix unit endpoints by setting
user names and passwords.
You can add four types of authentication information:
Table 265: Authentication Information
Authentication
Description
endpoint-specific Applies only to challenges received for SIP requests related to a specific endpoint. For
instance, the registration associated with the endpoint in the user agent table or the
INVITE sent to initiate a call from the endpoint. You can define several user names and
passwords for each endpoint of the Mediatrix unit. An endpoint can thus register with
several different realms.
gateway-specific
Applies only to challenges received for SIP requests on a specific SIP gateway. You can
define several user names and passwords for each endpoint of the Mediatrix unit. An
endpoint can thus register with several different realms.
unit
Applies to all challenges received for SIP dialog. You can define several user names
and passwords for the Mediatrix unit. These user names and passwords apply to all
endpoints of the unit.
user namespecific
Applies only to challenges for a context that uses a specific user name.
The Authentication table may have between 20 and 100 rows. Each of these rows can either be associated
with the unit, a specific gateway, a specific endpoint, or a specific user name. If you have less than 20 rows,
the Mediatrix unit automatically adds new rows up to the minimum of 20.
When a challenge occurs (either 401 or 407), the first entry in the Authentication table that matches the user
name/password request is used to reply to the challenge. You can configure the use name and password in
the web interface. The order of the tried entries in the Authentication table is from the first row to the last row.
The challenge matches an authentication entry if the realm of the challenge matches the realm specified in
the Realm field or if the Validate Realm field is set to disable. For each entry matching certain criteria
(described below), the challenge is replied with the entry's user name and password. If no entry matches the
criteria, the authentication fails. To match the authentication request, the entry must also meet one of the
following criteria:
The challenge needs to be for a SIP request related to the endpoint specified in the Endpoint
column if the corresponding Apply To column is set to Endpoint.
The challenge needs to be for a SIP request performed on the SIP gateway specified in the
Gateway column if the corresponding Apply To column is set to Gateway.
303
Chapter 27 - SIP Authentication
The challenge needs to be for a context that uses the user name specified in the User Name
field if the corresponding Apply To column is set to Usename. The user name associated with
a context is:
the user name of the FROM if the context sent the original SIP request, or
the user name of the request URI if the context received the original SIP request
The challenge applies to a unit if the corresponding Apply To column is set to Unit.
Creating/Editing an Authentication Entry

The web interface allows you to create authentication entries or modify the parameters of an existing one.
To create or edit SIP authentication parameters:

1.
In the web interface, click the SIP link, then the Authentication sub-link.
Figure 136: SIP Configuration Authentication Web Page
B
D
2.
Do one of the following:

a.
If you want to add an authentication entry before an existing entry, locate the proper row in the
table and click the
button of this row.
If you want to add an authentication entry at the end of the existing rows, click the
button
at the bottom right of the Authentication section.
b.
If you want to add several authentication entries at the same time, enter the number of entries
you want to add in the Number of rows to add at the bottom of the page.
c.
If you want to edit a single authentication entry, locate the proper row in the table and click the
button.
d.
If you want to edit several authentication entries of the current page at the same time, click the
Edit All Entries button at the bottom of the page.
This brings you to the proper Authentication panel.
304
Table 266: Authentication Panel Single Entry
10
Table 267: Authentication Panel Page
3.
10
Select which criterion to use for matching an authentication request with an authentication entry in
the Criteria column.
Table 268: Authentication Entity
Parameter
4.
Description
Unit
The authentication entry is used on all challenges.
Endpoint
The authentication entry used for all challenges related to

a specific endpoint.
Gateway
The authentication entry is used for all challenges related

to a specific SIP gateway.
Username
The authentication entry is used for all challenges related

to a specific user name..
Enter a string that identifies an endpoint in the UserAgent.

This field is available only if you have selected Endpoint in the Criteria column for the specific row.
5.
Enter a string that identifies a SIP gateway in the GatewayStatus table.

This field is available only if you selected Gateway in the Criteria column for the specific row.
6.
Enter a string that identifies a username in the SIP request to authenticate. this fiel is available only
if you selected Username in the Criteria column for the specific row.
305
Chapter 27 - SIP Authentication
7.
Select whether or not the current credentials are valid for any realm in the corresponding Validate
Realm drop-down menu.
Table 269: Realm Authentication Parameters
Parameter
8.
Description
Disable
The current credentials are valid for any realm. The corresponding Realm field is
read-only and cannot be modified.
Enable
The credentials are used only for a specific realm set in the corresponding Realm
field.
Enter a realm for each authentication row in the Realm column.

When authentication information is required from users, the realm identifies who requested it.
9.
Enter a string that uniquely identifies this endpoint in the realm in the User Name column.
10.
Enter a user password in the Password column.
11.
Registration.
Moving an Authentication Entry

The order of the tried entries in the Authentication table is from the first row to the last row. The rows sequence
is thus very important. If you want the unit to try to match one row before another one, you must put that row
first.
To move an authentication entry up or down:

1.
Either click the
or
arrow of the row you want to move until the entry is properly located.
Deleting an Authentication Entry

You can delete an authentication row from the table in the web interface.
To delete an authentication entry:

1.
306
Click the
H A P T E R
28
SIP Transport Parameters

This chapter describes the SIP transport parameters you can set.
SIP Transport Type

Standards Supported
RFC 2246: The TLS Protocol Version 1.0
RFC 3261: SIP, Session Initiation Protocol
You can globally set the transport type for all the endpoints of the Mediatrix unit to either UDP (User Datagram
Protocol), TCP (Transmission Control Protocol), or TLS (Transport Layer Security).
The Mediatrix unit will include its supported transports in its registrations.
Please note that RFC 3261 states the implementations must be able to handle messages up to the maximum
datagram packet size. For UDP, this size is 65,535 bytes, including IP and UDP headers. However, the
maximum datagram packet size the Mediatrix unit supports for a SIP request or response is 5120 bytes
excluding the IP and UDP headers. This should be enough, as a packet is rarely bigger than 2500 bytes.
To set the SIP transport type parameters:

1.
In the web interface, click the SIP link, then the Transport sub-link.
Figure 137: SIP Configuration Transport Web Page
2
4
6
7
2.
3
5
In the General Configuration section, enable or disable the transport registration in the Add SIP
Transport in Registration drop-down menu.
When enabled, the Mediatrix unit includes its supported transports in its registrations. It registers
with one contact for each transport that is currently enabled. Each of these contacts contains a
transport parameter.
307
Chapter 28 - SIP Transport Parameters
SIP Transport Type
This is especially useful for a system where there are no SRV records configured to use a
predefined transport order for receiving requests. When sending a request, the unit either follows
the SRV configuration, or, if not available, any transport parameter received from a redirection or
from a configured SIP URL.
Note: If the Mediatrix unit has the following configuration:

the Add SIP Transport in Registration drop-down menu is set to Disable
the UDP transport type is disabled
the TCP transport type is enabled
The unit will not work properly unless the SIP server uses the TCP transport type by default.
This is also true if the Mediatrix unit has the TCP transport disabled and the UDP transport enabled. In this
case, the unit will not work properly unless the SIP server uses the UDP transport protocol by default.
3.
Indicate whether or not the unit must include its supported transport in the Contact header in the
Add SIP Transport in Contact Header drop-down menu.
The supported transports are included in all SIP messages that have the Contact header, except
for the REGISTER message.
Available values are Enable and Disable. If you set the menu to Enable, the Mediatrix unit will send
SIP messages with the transport parameter in the Contact header set to:
transport=tcp when TCP is enabled and UDP is disabled
transport=udp when UDP is enabled and TCP disabled
no transport parameter when both TCP and UDP are enabled
transport=tls when secure transport (TLS) is selected
4.
Define the base port used to establish persistent connections with SIP servers when the TLS
transport is enabled in the Persistent Base Port field.
5.
Set the time interval, in seconds, before retrying the establishment of a persistent connection in the
Failback Interval field.
This is the interval that the Mediatrix unit waits before retrying periodically to establish a persistent
connection using a single IP address or a FQDN. This timer is started when a persistent connection
goes down or fails to connect to the destination.
A gateway automatically performs failover and failback procedure when it is configured with more
than one transport protocol or when responses to DNS queries return more than one IP address.
The failover and failback procedures are triggered when a SIP transaction fails. The type of
transaction that enables the failover depends on the type of the gateway. Additional SIP conditions
can be configured through SipFailoverConditions variables (see SIP Failover Conditions on
page 313).
During failover, the failed transaction is reattempted on another IP address with a lower priority
(according to the DNS response). When the Failback Interval expires, the IP address with a higher
priority is reattempted.
6.
In the TLS Trusted Certificate Level field, define how a peer certificate is considered trusted for a
TLS connection.
Table 270: Certificate Trust Level for TLS Connections Parameters
Parameter
308
Description
Locally
Trusted
A certificate is considered trusted when the certificate authority (CA) that signed the
peer certificate is present in the Others Certificates table (see Chapter 47 Certificates Management on page 569 for more details). The certificate revocation
status is not verified.
OCSP
Optional
A certificate is considered trusted when it is locally trusted and is not revoked by its
certificate authority (CA). The certificate revocation status is queried using the
Online Certificate Status Protocol (OCSP). If the OCSP server is not available or
the verification status is unknown, the certificate is considered trusted.
Additional Transport Parameters
Table 270: Certificate Trust Level for TLS Connections Parameters (Continued)
7.
Parameter
Description
OCSP
Mandatory
A certificate is considered trusted when it is locally trusted and is not revoked by its
certificate authority (CA). The certificate revocation status is queried using the
Online Certificate Status Protocol (OCSP). If the OCSP server is not available or
the verification status is unknown, the certificate is considered not trusted.
Set the TCP Connect Timeout field with the maximum time, in seconds, the unit should try to
establish a TCP connection to SIP hosts.
This timeout value is useful to have a faster detection of unreachable remote hosts. This timer can
also affect the TLS connection establishment time.
8.
In the Protocol Configuration section, enable or disable the UDP, TCP, and TLS transport type to
use in their corresponding drop-down menu.
Endpoint gateway supports UDP transport only. On a Trunk gateway, UDP and TCP are mutually
exclusive with TLS. Activating TLS automatically disables these unsecure protocols.
The successful configuration of a secure transport requires a little more than the activation of the
TLS protocol itself. You need to:
synchronize the time in the unit (see Time Configuration on page 96 & SNTP
Configuration on page 95 for more details).
install the security certificates used to authenticate the server to which you will connect
(see Chapter 47 - Certificates Management on page 569 for more details).
Use secure media (see Security on page 201 for more details).
configure the unit so that a transport=tls parameter is added to the Contact header of
your SIP requests (see Step 3).
Caution: If you have enabled Secure RTP (SRTP) on at least one line, it is acceptable to have the secure
SIP transport (TLS) disabled for testing purposes. However, you must never use this configuration in a
production environment, since an attacker could easily break it. Enabling TLS for SIP Transport is strongly
recommended and is usually mandatory for security interoperability with third-party equipment.
9.
Set the priority order of each transport type in the corresponding QValue field.
A qvalue parameter is added to each contact. The qvalue gives each transport a weight, indicating
the degree of preference for that transport. A higher value means higher preference.
The format of the qvalue string must follow the RFC 3261 ABNF (a floating point value between
0.000 and 1.000). If you specify an empty string, no qvalue is set in the contacts.
10.


by using the CLI
309
Transport TLS Cipher Suite Settings

You can define the allowed cipher suites for the network security settings when using TLS connection.
Table 271: Cipher Suites Configuration Parameters
Parameter
CS1
CS2
CS3
Description
This is the default value and represents the cipher suites configuration prior to this variable
being introduced. This should be changed if additional network security is required. This
value includes the following cipher suites:
This represents a secure configuration using SHA-1. This value includes the following cipher
suites:
This represents the most secure configuration using SHA-2. Only the most secure cipher
suites are allowed when using this value.
TLS_CDH_RSA_WITH_AES_128_CBC_SHA256
To set the TLS transport cipher suite configuration parameter:

1.
In the sipMIB, set the TLS transport cipher suite configuration in the TransportTlsCipherSuite
variable.
sip.TransportTlsCipherSuite="Value"
310

Value
Meaning
100
CS1
200
CS2
300
CS3
UDP Source Port Behaviour

On Trunk gateway type, you can configure whether or not the Mediatrix unit always uses the same local port
(the port on which it is listening for incoming packets) when sending SIP traffic over UDP. This is called
symmetric UDP source port. Symmetric UDP ports are sometimes needed to traverse NAT/Firewall devices.
When changing this setting, all destinations are automatically sent out of the penalty box, when applicable.
This variable has no effect on Endpoint gateways. Endpoint gateways always use the same UDP port for
sending and receiving messages.
Table 273: UDP Source Port Parameters
Parameter
Description
disable
The SIP signalling over UDP uses a randomly-generated originating port. ICMP errors are
processed correctly.
enable
The SIP signalling sent over UDP originates from the same port as the port on which the user
agent is listening. ICMP messages are not processed, which means that unreachable targets
will take longer to detect.
To set the UDP source port behaviour:

1.
In the sipEpMIB, set whether or not the unit uses the symmetric source port feature in the
interopSymmetricUdpSourcePortEnable variable.
sipEp.interopSymmetricUdpSourcePortEnable="Value"

Table 274: UDP Source Port Values
Value Meaning
2.
disable
enable
Restart the SipEp service by accessing the scmMIB and setting the serviceCommandsRestart
variable for the SipEp service to restart.
scm.serviceCommands.Restart[Name=SipEp]="10"
TLS Client Authentication

When acting as a TLS server, it is customary not to request from the clients that they authenticate themselves
via the TLS protocol. However, if mutual authentication is required between client and server, you can set the
Mediatrix unit so that it requests client authentication when acting as a TLS server.
311

Table 275: TLS Client Authentication Parameters
Parameter
Description
disable
The Mediatrix unit does not require TLS clients to provide their host certificate for the
connection to be allowed. This is the default value.
enable
The TLS clients must provide their host certificate for the connection to be allowed. In this case,
the level of security used to validate the host certificate is TrustedCertificate, whatever the
value set in the Certificate Validation drop-down menu of the TLS Interop section ( SIP >
Interop web page). See TLS Interop on page 322 for more details.
To set TLS client authentication:

1.
In the sipEpMIB, set whether or not the Mediatrix unit requests client authentication when acting as
a TLS server in the interopTlsClientAuthenticationEnable variable.
sipEp.interopTlsClientAuthenticationEnable="Value"

Table 276: TLS Client Authentication Values
Value Meaning
0
disable
enable
Force DNS NAPTR In TLS

The Mediatrix unit allows you to force a DNS NAPTR request when the SIP transport is TLS.
This variable only applies to calls over TLS when the Supported DNS Queries drop-down menu of the SIP >
Misc page is set to NAPTR (see DNS Configuration on page 351 for more details).
Table 277: Force DNS NAPTR in TLS Parameters
Parameter
Description
disable
The DNS SRV request is sent directly with the SIP transport in SIP URI as recommended in
RFC 3263, section 4.1.
enable
A DNS NAPTR request is sent to obtain the DNS record associated with SIP over TLS. An SRV
request is performed afterward. If no SIP over TLS entry is returned, the call fails.
To force DNS NAPTR in TLS:

1.
In the sipEpMIB, set whether or not to force a DNS NAPTR request in the
InteropForceDnsNaptrInTls variable.
sipEp.interopForceDnsNaptrInTls="Value"

Table 278: Force DNS NAPTR in TLS Values
Value Meaning
312
disable
enable
SIP Failover Conditions

You can configure additional SIP-level conditions for failover. These conditions can also be configured
specifically per gateway.
To set the SIP failover conditions:
1.
In the sipEpMIB, set the DefaultSipFailoverConditions variable to the proper SIP failover
condition value.
sipEp.defaultSipFailoverConditions="Value"
where Value is a sequence of keywords separated by commas; spaces and tabs are ignored. If
Value is empty, only the connection-level failover conditions apply.
Supported keywords list is:
5xxOnRegistration: 5xx (Server Failure) response received on a registration attempt.
Note: The syntax is designed to support multiple keywords even though only a single keyword is defined
for now.
2.
If you want to set failover conditions for a specific SIP gateway, set the following variables:
gwSpecificFailoverEnableConfig variable for the specific SIP gateway you want to
gwSpecificFailoverSipFailoverConditions variable for the specific SIP gateway
you want to configure to the proper value.
sipEp.gwSpecificFailover.EnableConfig[GatewayName="default"]="5xxOnRegistration"
sipEp.gwSpecificFailover.SipFailoverConditions[GatewayName="Specific_Gateway"]="
Value"
where:
Value is a SIP failover condition as defined in Step 1.
313
314
H A P T E R
29
Interop Parameters
This chapter describes the interoperability parameters that allow the Mediatrix unit to properly work,
communicate, or connect with specific IP devices.
Standards Supported
draft-ietf-sipping-realtimefax-00
ITU-T Recommendation T.38, section D.2.3
RFC 3264: An Offer/Answer Model with the Session

Description Protocol (SDP)
RFC 3515: The Session Initiation Protocol (SIP) Refer

Method
Behavior on T.38 INVITE Not Accepted

This section describes the units behaviour after receiving an error to a SIP INVITE for T.38 fax.
To set the T.38 interop parameters:

1.
In the web interface, click the SIP link, then the Interop sub-link.
Figure 138: SIP Interop Web Page
2.
In the Behavior on T.38 INVITE Not Accepted section, for each of 406, 415, 488, and 606 SIP code,
set the behaviour after receiving the code in the error response to an INVITE for T.38 fax in the
corresponding Behavior drop-down menu.
Table 279: Behavior on T.38 INVITE Not Accepted Parameters
Behavior
Description
Drop Call
The call is dropped by sending a BYE.
ReInviteForClearChannelOnly
A re-INVITE is sent with audio codecs that support clear channel

faxes.
Re-Establish Audio
A re-INVITE is sent to re-establish the audio path. Also, fax

detection is disabled for the remainder of the call.
UsePreviousMediaNegotiation No re-INVITE is sent and the audio codec from the last
successful negotiation is used. For the remainder of the call,
T.38 is disabled and fax detection may trigger a switch to a clear
channel codec that was available in the last successful
negotiation.
315
Chapter 29 - Interop Parameters
3.
SIP Interop
SIP Interop
Standards Supported
This section describes the SIP interoperability parameters of the Mediatrix unit .
To set the SIP interop parameters:

1.
In the SIP Interop section of the Interop page, set whether or not the x-Siemens-Call-Type header
is added to the SIP packets sent by the unit in the Secure Header drop-down header.
You can set the Mediatrix unit so that it triggers the addition of the x-Siemens-Call-Type header to
the SIP packets sent by the unit when secure transport is in use.
Table 280: Secure Transport Header Parameters
Parameter
Description
disable
The x-Siemens-Call-Type header is not added to the SIP packets sent by the
unit.
enable
The x-Siemens-Call-Type header is added to the SIP packets sent by the unit, and
assigned the value ST-secure, as soon as secure transport and secure payload
are being used. If secure transport or secure payload are not used, the header is
not added.
Figure 139: SIP Interop Section
1
2
3
5
4
6
7
8
2.
Select the username to use when the username is empty or undefined in the Default Username
Value drop-down menu.
Table 281: Default Username Value
Parameter
3.
Description
Anonymous
Sets the username to anonymous.
Host
Sets the username to the same value as the host.
Define the behaviour of the Mediatrix unit when answering a SIP OPTIONS request in the
OPTIONS Method Support drop-down menu.
Table 282: OPTIONS Method Support Parameters
Parameter
316
Description
None
The Mediatrix unit responds with an error 405 Method not

allowed.
AlwaysOK
The Mediatrix unit responds with a 200 OK regardless of

the content of the OPTIONS request.
SIP Interop
4.
Define whether or not the SIP OPTIONS requests should be ignored when all endpoints are
unusable in the Ignore OPTONS on no usable endpoints drop-down menu.
Table 283: Ignore SIP Options Parameters
Parameter
Description
Enable
The unit ignores SIP OPTIONS requests when all

endpoints are unusable. When at least one endpoint is
usable, then the SIP OPTIONS requests are answered
as configured in the OPTIONS Method Support dropdown menu (see Step 10).
Disable
The SIP OPTIONS requests are answered as configured

in the OPTIONS Method Support drop-down menu (see
Step 10) regardless of the state of the endpoints.
Note that this feature may be influenced by whether or not you have enabled the Monitor Link State
parameter. For more information:
5.
ISDN PRI interface: PRI Configuration on page 186
ISDN BRI interface: BRI Configuration on page 197
R2 PRI interface: R2 Channel Associated Signaling on page 226
Set the value of the user parameter in SIP URIs sent by the unit in the SIP URI User Parameter
Value field.
If you leave the field empty, the parameter is not added.
E.g : sip:1234@domain.com;user=InteropSipUriUserParameterValue
Note that when the Map Plus To TON International drop-down menu is set to Enable, the
parameter's value might be overwritten (Misc Interop on page 323).
6.
Set the Behavior On Machine Detection drop-down menu with the SIP devices behavior when a
machine (fax or modem) is detected during a call.
Table 284: Behavior on Machine Detection Parameters
Parameter
Description
Re-INVITE On Fax T38 Only
A SIP re-INVITE is sent only on a fax detection and T.38

is enabled.
Re-INVITE On No Negotiated Data

Codec
A SIP re-INVITE is sent on a fax or modem detection if no

data codec was previously negotiated in the original SDP
negotiation. In the case where at least one data codec
was previously negotiated in the SDP negotiation, the
device switches silently to a data codec without sending a
SIP re-INVITE. Note sthat if there is no data codec
enabled on the device, no SIP re-INVITE is sent and the
call is dropped by sending a BYE.
Re-INVITE Unconditional
A SIP re-INVITE is sent with data codecs upon detection

of a fax or modem even if a data codec was negotiated in
the initial offer-answer. The T.38 codec is offered if it is
enabled and a fax is detected.
See Data Codec Selection Procedure on page 221 for more details on the procedure the Mediatrix
unit follows when selecting data codec.
317
7.
SDP Interop
Set the Registration Contact Matching field with the matching behaviour for the contact header
received in positive responses to REGISTER requests sent by the unit.
Table 285: Registration Contact Matching Parameters
Parameter
8.
Description
Strict
Matches the complete contact's SIP URI including any URI parameters, if any, as
per RFC 3261 sections '10.2.4 Refreshing Bindings' and '19.1.4 URI Comparison'.
The contact's SIP URI of a 2XX positive response MUST match the contact's SIP
URI of the REGISTER request.
IgnoreUriP
arams
Matches the username and the host port part of the contact's SIP URI. All URI
parameters are ignored.
Set the Transmission Timeout field with the time to wait for a response or an ACK before
considering a transaction timed out.
This corresponds to timers B, F and H for all transport protocols and timer J for UDP. These timers
are defined in section A of RFC 3261.
This timeout affects the number of retransmissions. Retransmissions continue to follow the timing
guidelines described in RFC 3261.
If a DNS SRV answer contains more than one entry, the Mediatrix unit will try these entries if the
entry initially selected does not work. You can configure the maximum time, in seconds, to spend
waiting for answers to messages, from a single source. Retransmissions still follow the algorithm
proposed in RFC 3261, but the total wait time can be overridden by using this feature.
For example, if you are using DNS SRV and more than one entry are present, this timeout is the
time it takes before trying the second entry.
Available values are from 1 to 32 seconds.
9.
SDP Interop
Standards Supported
RFC 3264: An Offer/Answer Model with Session Description

Protocol (SDP)
This section describes the SDP interoperability parameters of the Mediatrix unit.
318
SDP Interop
To set the SDP interop parameters:

1.
In the SDP Interop section of the Interop page, Offer Answer Model part, select the codec
negotiation rule when generating a SDP answer in the Answer Codec Negotiation drop-down menu.
Table 286: Answer Codec Negotiation Parameters
Parameter
Description
All Common - Local

Priority
When generating an answer to an offered session, all common codecs

are listed in the local order of priority. The local priority is defined for each
codec in the Telephony > CODECS page by clicking the
button of
each codec and looking in the Voice Priority and Data Priority fields. See
Chapter 14 - Voice & Fax Codecs Configuration on page 181 for more
details.
First Common Local Priority
When generating an answer to an offered session, only the first common

codec with the higher local priority is listed. The local priority is defined
for each codec in the Telephony > CODECS page by clicking the
button of each codec and looking in the Voice Priority and Data Priority
fields. See Chapter 14 - Voice & Fax Codecs Configuration on
All Common - Peer

Priority
When generating an answer to an offered session, all common codecs

are listed. The codecs order is the same as in the peer offer.
First Common Peer Priority
When generating an answer to an offered session, only the first common

codec is listed. The codecs order is the same as in the peer offer.
Figure 140: SDP Interop Section
1
2
3
4
5
6
7
8
2.
Select whether or not the Mediatrix unit requires strict adherence to RFC 3264 when receiving an
answer from the peer when negotiating capabilities for the establishment of a media session in the
Enforce Offer Answer Model drop-down menu.
Table 287: Offer/Answer Model Parameters
Parameter
Disable
Description
The peer can freely:
Send back a brand new list of codecs or add new ones to the offered
list.
Add new media lines.
As long as at least one codec sent back was present in the initial offer, the call
is allowed to go on. Any media line added by the peer is simply ignored.
319
SDP Interop
Table 287: Offer/Answer Model Parameters

Parameter
Enable
Description
The following guidelines from the Offer-Answer Model must be strictly
followed. An answer must:
Include at least one codec from the list that the Mediatrix unit sent
in the offer.
Contain the same number of media lines that the unit put in its offer.
Otherwise, the answer is rejected and the unit ends the call. This is the default
value.
3.
Define the behaviour of the Mediatrix unit when receiving less media announcements in the
response than in the offer in the Allow Less Media In Response drop-down menu.
Table 288: Less Media Announcements Parameters
Parameter
4.
Description
Disable
The Mediatrix unit rejects the response with less media announcements than
in the offer.
Enable
The Mediatrix unit tries to find matching media when the response contains
less media announcement than in the offer. This is a deviation from the Offer/
Answer model.
Define the behaviour of the Mediatrix unit when receiving a SDP answer activating a media that had
been previously deactivated in the offer in the Allow Media Reactivation in Answer drop-down
menu.
Table 289: Media Reactivation Parameters
Parameter
5.
Description
Enable
A media reactivated in an incoming answer is ignored. This behaviour goes

against the SDP Offer/Answer model described by IETF RFC 3264.
Disable
A media reactivated in an incoming answer ends the current media negotiation

and the call. This behaviour follows the SDP Offer/Answer model described by
IETF RFC 3264.
In the Multiple Active Media part, define the behaviour of the Mediatrix unit when offering media or
answering to a media offer with audio and image negotiation in the Allow Audio and Image
Negotiation drop-down menu.
Table 290: Audio and Image Negotiation Parameters
Parameter
320
Description
Enable
The unit offers audio and image media simultaneously in

outgoing SDP offers and transits to T.38 mode upon
reception of a T.38 packet. Also, when the unit answers
positively to a SDP offer with audio and image, it transits
to T.38 mode upon reception of a T.38 packet.
Disable
Outgoing offers never include image and audio

simultaneously. Incoming offers with audio and image
media with a non-zero port are considered as offering
only audio.
SDP Interop
6.
Define the behaviour of the Mediatrix unit when answering a request offering more than one active
media in the Allow Multiple Active Media in Answer drop-down menu.
Figure 141: Allow Multiple Active Media in Answer
Parameter
7.
Description
disable
The answer contains only one active media. The media specified as active
in the answer is the top-most matching one in the offer. Other media are
set to inactive.
enable
Each matching active media in the offer is specified as active in the

answer. Other media are set to inactive
In the Other part, define how to set the direction attribute and the connection address in the SDP
when answering a hold offer with the direction attribute sendonly in the On Hold SDP Stream
Direction in Answer drop-down menu.
The following parameters are supported:
Table 291: sendonly Direction Attribute
Parameter
Description
inactive
The stream is marked as inactive and if the stream uses IPv4, the
connection address is set to '0.0.0.0'.
revconly
If the stream is currently active or receive only, it is marked as recvonly

and the connection address is set to the IP address of the unit.
If the stream is currently send only or inactive, it is marked as inactive
and if the stream uses IPv4, the connection address is set to '0.0.0.0'.
This method is in conformance with RFC 3264.
In both cases, no direction attribute is present in the SDP if the

interopSdpDirectionAttributeEnable variable is set to disable (see Direction Attribute on
8.
Set the Codec vs Bearer Capabilities Mapping Preferred Codec Choice drop-down menu with the
behaviour of the Codec vs. Bearer Capabilities Mapping table.
This modifies the selection of the preferred codec in the incoming SDP. This parameter is available
only on ISDN interfaces.
The Codec vs. Bearer Capabilities Mapping table parameters are located in the Telephony >
CODECS > CODEC vs. Bearer Capabilities Mapping section. See Codec vs. Bearer Capabilities
Mapping on page 188 for more details.
Table 292: Codec vs Beareer Capabilities Mapping Preferred Codec Choice Parameters
Parameter
Description
First Codec
The first valid codec in the incoming SDP is considered

the preferred one and is used when looking up the
Codec vs. Bearer Capabilities Mapping table.
Prioritize Clear Channel
When a clear channel codec is in the incoming SDP, it

is always considered as the preferred one, no matter
where it stands in the codec list, and is used when
looking up the Codec vs. Bearer Capabilities Mapping
table.
321
9.
TLS Interop
Note: If you are experiencing media negotiation problems (because the Mediatrix unit sends a BYE after
receiving a 200 OK), try to set the Enforce Offer Answer Model value to Disable and the Allow Less Media
In Response value to Enable.
TLS Interop
This section describes the TLS interoperability parameters of the Mediatrix unit .
To set the TLS interop parameters:

1.
In the TLS Interop section of the Interop page, select the level of security used to validate the TLS
server certificate when the unit is acting as a TLS client in the Certificate Validation drop-down
menu.
Figure 142: TLS Interop Section
1
Note: This parameter has no effect on the TLS client authentication when the unit is acting as a TLS server
(see the interopTlsClientAuthenticationEnable variable in TLS Client Authentication on page 311).
Table 293: TLS Certificate Validation Parameters
Parameter
2.
322
Description
No Validation
No validation of the peer certificate is performed. All TLS connections are

accepted without any verification. Note that at least one certificate must be
returned by the peer even if no validation is made. This option provides no
security and should be restricted to a lab use only.
Trusted
Certificate
Allows a TLS connection only if the peer certificate is trusted. A certificate is

considered trusted when the certificate authority (CA) that signed the peer
certificate is present in the Management > Certificates page (Chapter 47 Certificates Management on page 569). This option provides a minimum level
of security and should be restricted to a lab use only.
Dns Srv
Response
Allows a TLS connection if the peer certificate is trusted and contains a known
host name. A known host name can be the FQDN or IP address configured as
the SIP server, or can also be returned by a DNS SRV request. In this case,
the match is performed against the DNS response name. If it matches either
one of the Subject Alternate Name (SAN) or Common Name (CN) in the peer
certificate, the connection is allowed. This option provides an acceptable level
of security, but not as good as Host Name.
HostName
Allows a TLS connection if the peer certificate is trusted and contains a known
host name. A known host name can only be the FQDN or IP address
configured as the SIP server. If it matches either one of the Subject Alternate
Name (SAN) or Common Name (CN) in the peer certificate, the connection is
allowed. This option provides the highest level of security.
Misc Interop
Misc Interop
This section describes miscellaneous interoperability parameters of the Mediatrix unit .
To set the Misc interop parameters:

1.
In the Misc Interop section of the Interop page, select whether or not the Mediatrix unit enables the
mapping between the + prefix of the user name and the type of number property in the Map Plus
To TON International drop-down menu.
When enabled, the service has the following behaviour:
For a call to SIP, the Mediatrix unit prefixes the user name with '+' if the call has the call
property type of number set to international. The unit also adds the user parameter
with the value phone to the SIP URI. For instance:
sip:1234@domain.com;user=phone.
For a call from SIP, the Mediatrix unit sets the call property type of number to
international if the user name has the prefix '+'.
Figure 143: Misc Interop Section
1
2
3
4
2.
Define the Ignore Plus in Username drop-down menu as to whether or not the plus (+) character is
ignored when attempting to match a challenge username with usernames in the Authentication
table.
Table 294: Ignore Plus (+) Character in Username Parameters
Parameter
3.
Description
Enable
The plus (+) character is ignored when attempting to match a username in the
authentication table.
Disable
The plus (+) character is not ignored when attempting to match a username in the
authentication table.
Select whether or not the pound character (#) must be escaped in the username part of a SIP URI
in the Escape Pound (#) in SIP URI Username drop-down menu.
Table 295: Escape Pound Parameters
Parameter
Description
Enable
The Pound character (#) is escaped in the username

part of a SIP URI.
Disable
The Pound character (#) is not escaped in the

username part of a SIP URI.
Note that RFC 3261 specifies that the pound character
(#) needs to be escaped in the username part of a SIP
URI.
323
4.
Additional Interop Parameters
Select the format of the escaped characters to be used in all SIP headers in the Escape Format
drop-down menu.
Table 296: Escape Format Parameters
Parameter
5.
Description
Lower Hexadecimal
Escaped characters are displayed in a lowercase

hexadecimals format.
Upper Hexadecimal
Escaped characters are displayed in a uppercase

hexadecimals format.


by using the CLI
The interop parameters allow the Mediatrix unit to properly work, communicate, or connect with specific IP
devices.
Call Waiting Private Number Criteria for SIP INFO

You can specify the call waiting criteria, in the form of a regular expression, that defines a private number
received in a SIP INFO.
To set the Call Waiting Private Number Criteria:

1.
In the sipEpMIB, set the Call Waiting Private Number Criteria in the
InteropCallWaitingSipInfoPrivateNumberCriteria variable.
sipEp.interopCallWaitingSipInfoPrivateNumberCriteria="Value"
For example, the value "(Anonymous|anonymous)" would define a calling number that is either
"Anonymous" or "anonymous" as private. The regular expression symbols to match the beginning
and end of the number are implicit and do not need to be specified. See Regular Expressions on
The variable is effective only if the Default Hook-Flash Processing parameter of the SIP > Misc page
is set to TransmitUsingSignalingProtocol (see General Configuration on page 421 for more
details).
Max-Forwards Header
Standards Supported
Max-Forwards serves to limit the number of hops a request can make on the way to its destination. It consists
of an integer that is decremented by one at each hop. If the Max-Forwards value reaches 0 before the request
reaches its destination, it is rejected with a 483 (Too Many Hops) error response. The Max-Forwards SIP
header is always present and the default value is 70.
324
Direction Attributes in a Media Stream

The Mediatrix unit allows you to define various direction attributes pertaining to the media stream.
When Putting a Call on Hold

Standards Supported

Protocol (SDP)
The Mediatrix unit can provide the direction attribute and the meaning of the connection address 0.0.0.0 sent
in the SDP when an endpoint is put on hold.
Table 297: Direction Attributes
Parameter
Description
inactive
The stream is put on hold by marking it as inactive. This is the default value. This setting
should be used for backward compatibility issues.
sendonly
The stream is put on hold by marking it as sendonly. This method allows the Mediatrix unit to
be in conformance with RFC 3264.
To define the direction attribute when putting a call on hold:

1.
In the sipEpMIB, set the interopOnHoldSdpStreamDirection variable to the proper value.

sipEp.interopOnHoldSdpStreamDirection="Value"

Table 298: Direction Attributes Values
Value
Meaning
100
inactive
200
sendonly
This configuration has no effect if the interopSdpDirectionAttributeEnable variable is set to

disable (see Direction Attribute on page 325 for more details).
Direction Attribute
Standards Supported

Protocol (SDP)
You can define if the SDP direction attribute is supported by the unit.
This variable applies only when the negotiated media uses an IPv4 address. The application always behaves
as if this variable is set to Enable for media using an IPv6 address.
Table 299: SDP Direction Attribute
Parameter
disable
Description
No direction attribute is present in the SDP sent by the Mediatrix unit.
The Mediatrix unit ignores any direction attribute found in the SDP received from the
peer.
The method to put a session on hold is in conformance with RFC 2543.
325
Table 299: SDP Direction Attribute (Continued)

Parameter
enable
Description
The Mediatrix unit always sends the direction attribute in the SDP of an initiated call. For
all other SDP messages sent by the unit, refer to Enable/Disable SDP Detect Peer
Direction Attribute Support on page 326.
If present in the SDP, the direction attribute is preferred over the connection address to
transmit session modification information.
To define if the direction attribute is present:

1.
In the sipEpMIB, set the interopSdpDirectionAttributeEnable variable to the proper value.

sipEp.interopSdpDirectionAttributeEnable="Value"

Table 300: SDP Direction Attribute
Value Meaning
0
disable
enable
Enable/Disable SDP Detect Peer Direction Attribute Support

You can define if the SDP direction attribute support should be autodetected in the SDP received from the
peer.
This variable is used only when the negotiated media uses an IPv4 address and when the
interopSdpDirectionAttributeEnable is enabled (see Direction Attribute on page 325 for more details).
The application always behaves as if this variable is set to 'Disable' for media using an IPv6 address.
Table 301: SDP Detect Peer Direction Attribute Parameters
Parameter
Description
disable
The Mediatrix unit always sends the direction attribute in the SDP without autodetection
of peer support.
enable
The initial handshake determines if the peer supports the direction attribute. The
direction attribute will be present when the peer supports it.
To define if the SDP detect peer direction attribute is enabled or disabled:

1.
In the sipEpMIB, set the interopSdpDetectPeerDirectionAttributeSupportEnable variable to

the proper value.
sipEp.interopSdpDetectPeerDirectionAttributeSupportEnable="Value"

Table 302: SDP Detect Peer Direction Attribute Values
Value Meaning
326
disable
enable
On Hold SDP Connection Address

You can define the value of the connection address sent in the SDP when an endpoint is on hold and no longer
listening to media packets.
This variable is used only when the negotiated media uses an IPv4 address. The application always behaves
as if this variable is set to 'MediaAddress' for media using an IPv6 address.
Table 303: On Hold SDP Connection Address Parameters
Parameter
Description
HoldAddress
The connection address sent in the SDP is '0.0.0.0' if the media uses an IPv4 address.
This method is described by RFC 2543.
MediaAddress
The connection address sent in the SDP is the listening address.
To define the on hold SDP connection address:

1.
In the sipEpMIB, set the interopOnHoldSdpConnectionAddress variable to the proper value.

sipEp.interopOnHoldSdpConnectionAddress="Value"

Table 304: On Hold SDP Connection Address Values
Value
Meaning
100
HoldAddress
200
MediaAddress
Answering a Hold Offer with the Direction Attribute sendonly

Standards Supported

Protocol (SDP)
You can define how to set the direction attribute in the SDP when answering a hold offer with the direction
attribute 'sendonly'.
Parameter
Description
inactive
The stream is marked as inactive and if the stream uses an IPv4

address, the connection address is set according to the
InteropOnHoldSdpConnectionAddress variable (On Hold SDP
Connection Address on page 327).
revconly
If the stream is currently active or receive only, it is marked as recvonly

and the connection address is set to the IP address of the unit.
If the stream is currently send only or inactive, it is marked as inactive
and the connection address is set according to the
InteropOnHoldSdpConnectionAddress variable (On Hold SDP
Connection Address on page 327).
327
To define the behaviour with the sendonly direction attribute:

1.
In the sipEpMIB, set the InteropOnHoldAnswerSdpStreamDirection variable to the proper value.

sipEp.interopOnHoldAnswerSdpStreamDirection="Value"

Value Meaning
100
inactive
200
Recvonly
In both cases, no direction attribute is present in the SDP if the

interopSdpDirectionAttributeEnable variable is set to disable (see Direction Attribute on
SDP Direction Attribute Level

Standards Supported

Protocol (SDP)
You can define the preferred location where the stream direction attribute is set.
Table 307: SDP Direction Attribute Level
Parameter
MediaOrSessionLevel
Description
If every media have the same direction, the stream direction attribute is
only present at session level.
Otherwise, the stream direction attribute is only present at media level.
MediaAndSessionLevel
If every media have the same direction, the stream direction attribute is
present both at session level and media level.
Otherwise, the stream direction attribute is only present at media level.
To define the SDP direction attribute level:

1.
In the sipEpMIB, set the InteropSdpDirectionAttributeLevel variable to the proper value.

sipEp.InteropSdpDirectionAttributeLevel="Value"

Table 308: SDP Direction Attribute Level
Value
Meaning
100
MediaOrSessionLevel
200
MediaAndSessionLevel
Local Ring Behaviour on Provisional Response

You can set the Mediatrix unit so that it starts or not the local ring upon receiving a 18x Provisional response
without SDP.
This setting does not affect the behaviour when the 18x Provisional response contains SDP, which allows
establishing an early media session before the call is answered.
328
This variable does not affect the behaviour in case the '18x Provisional' response contains SDP, in which case
the media stream, if present, is played.
Figure 144: Local Ring Behaviour
Parameter
Disable
Description
The local ring is not started on a '18x Provisional' response without SDP, with one
exception: the '180 Ringing' without SDP will start the local ring if the media stream is
not already established.
LocalRingWhenNo : The local ring is started on any '18x Provisional' response without SDP if the media
EstablishedMediaS stream is not already established.
tream
LocalRingAlways
The local ring is always started on any '18x Provisional' response without SDP.
To define the local ring behaviour on provisional response:

1.
In the sipEpMIB, set the interopLocalRingOnProvisionalResponse variable to the proper value.

sipEp.interopLocalRingOnProvisionalResponse="Value"

Figure 145: Local Ring Values
Value
Meaning
disable
LocalRingWhenNoEstablishedMediaStream
LocalRingAlways
Session ID and Session Version Number in the Origin Field of the SDP
You can define the maximum length of the session ID and the session version number in the origin line (o=)
of the SDP. This allows the Mediatrix unit to be compatible with 3rd party vendor equipment.
Table 309: Maximum Length Parameters
Length
Description
max-32bits
The session ID and the session version number are represented with a 32 bit integer.
They have a maximum length of 10 digits.
max-64bits
The session ID and the session version number are represented with a 64 bit integer.
They have a maximum length of 20 digits. This is the default value.
To set the maximum length of the session ID and the session version number:
1.
In the sipEpMIB, set the interopSdpOriginLineSessionIDAndVersionMaxLength variable with

the proper length.
sipEp.interopSdpOriginLineSessionIdAndVersionMaxLength="Value"
329

Table 310: Maximum Length Values
Value
Meaning
100
max-32bits
200
max-64bits
Register Home Domain Override

By default, the address-of-record in the To header uses the value set in the Proxy Host field of the SIP/
Configuration page for the host/port part. See SIP Servers Configuration on page 284 for more details. You
can override this value if required.
To override the register home domain value:

1.
In the sipEpMIB, set the interopRegisterHomeDomainOverride variable with the override home
domain value.
sipEp.interopRegisterHomeDomainOverride="IP_Address"
The address of record in the register will use this string instead of the home domain proxy. If the
variable is empty, the value of the Proxy Host field is used.
The host is also overridden in the From and Call-Id headers since they match the To header.
DNS SRV Record Lock

Standards Supported
RFC 3263 - Session Initiation Protocol (SIP): Locating SIP

Servers
You can configure the Mediatrix unit to always use the same DNS SRV record for a SIP call ID. As a result, a
call or registration always uses the same destination until the destination is unreachable or the unit receives
a different DNS SRV result.
Table 311: DNS SRV Record Lock Parameters
Length
Description
disable
The behaviour follows RFC 3263.
enable
All messages during a call or registration use the same SRV record.
To enable the DNS SRV record lock feature:

1.
In the sipEpMIB, set the interopLockDnsSrvRecordPerCallEnable variable to enable.

sipEp.interopLockDnsSrvRecordPerCallEnable="Value"

Figure 146: DNS SRV Record Lock Values
Value Meaning
330
disable
enable
Listening for Early RTP

Note: This feature applies to the following models:
Mediatrix 3716 / 3731 / 3732 / 3741 / 3742
Mediatrix LP Series
Mediatrix C7 Series
You can set the Mediatrix unit so that it listens for RTP before the reception of a response with SDP. This
feature only applies to calls initiated from analog endpoints (FXS/FXO) with non-secure RTP.
Table 312: Early RTP Parameters
Length
Description
enable
The RTP port is opened after the initial INVITE has been sent, without waiting for a provisional or
final response with SDP to be received. No local ring is generated. This conforms to section 5.1 of
RFC 3264.
disable
The RTP port is opened only after a response with SDP is received.
Warning: Do not enable this feature unless the server supports early RTP (or early media). Failing so
prevents any ringing to be heard for outgoing calls.
To enable the Early RTP feature:

1.
In the sipEpMIB, set the InteropListenForEarlyRtpEnable variable to enable.

sipEp.InteropListenForEarlyRtpEnable="Value"

Figure 147: Early RTP Values
Value Meaning
0
disable
enable
Resolve Route Header

The Mediatrix unit has a parameter that allows you to resolve the FQDN in the top-most route header of
outgoing packets.
Table 313: Resolve Route Header Parameters
Length
Description
enable
The FQDN in the top-most route header is replaced by the IP address of the packet's destination
if the FQDN matches the gateway's configured outbound proxy.
disable
The route header is not modified.
331
To resolve the route header:

1.
In the sipEpMIB, set the InteropResolveRouteHeaderEnable variable with the proper value.
sipEp.interopResolveRouteHeaderEnable="Value"

Figure 148: Resolve Route Header Values
Value Meaning
0
disable
enable
ACK Branch Matching

You can configure the method used to match incoming ACK SIP packets.
Table 314: ACK Branch Matching Parameters
Parameter
Rfc3261
Description
Follows the method described in RFC 3261 (section 8.1.1.7). The branch value in the topmost
via of the ACK request to a 2XX response MUST be different than the one of the INVITE.
Rfc3261Wi Follows the method described in RFC 3261 (section 8.1.1.7) but enables the handling of ACK
thoutAck
requests (for 2XX responses) that have the same branch value in the topmost via as the
INVITE.
To set ACK branch matching:

1.
In the sipEpMIB, set the interopAckBranchMatching variable with the proper value.
sipEp.interopAckBranchMatching="Value"

Figure 149: ACK Branch Matching Values
Value
Meaning
100
Rfc3261
200
Rfc3261WithoutAck
Ignore Require Header

You can define whether or not the Require Header must be ignored when processing the incoming SIP Client
requests (INVITE, re-INVITE, Bye, etc.).
Table 315: Ignore Require Header Parameters
Parameter
Enable
332
Description
The Require Header is ignored and no validation about these options-tags is performed.
Table 315: Ignore Require Header Parameters (Continued)

Parameter
Disable
Description
The Require Header options-tags are validated and, when an option-tag is not supported, a
420 (Bad Extension) response is sent.
The supported options-tags are:
* 100rel
* replaces
* timer
To set whether or not to ignore the Require header:

1.
In the sipEpMIB, set the interopIgnoreRequireHeaderEnable variable with the proper value.
sipEp.interopIgnoreRequireHeaderEnable="Value"

Figure 150: Ignore Require Header Values
Value Meaning
0
disable
enable
Reject Code for Unsupported SDP Offer

You can define the rejection code used when an offer is received with invalid or unsupported SDP Offer. RFC
3261 recommends using the error code 488 'Not Acceptable Here'.
Table 316: Reject Code for Unsupported SDP Offer Parameters
Parameter
Description
UnsupportedMediaType The SIP error code 415 'Unsupported Media Type' is returned if the ContentType is invalid; the payload is missing or the SDP content is invalid.
NotAcceptableHere
The SIP error code 488 'Not Acceptable Here' is returned if the SDP content is
invalid.
To set the reject code:

1.
In the sipEpMIB, set the InteropRejectCodeForUnsupportedSdpOffer variable with the proper

value.
sipEp.InteropRejectCodeForUnsupportedSdpOffer="Value"

Figure 151: Reject Code Values
Value
Meaning
415
UnsupportedMediaType
488
NotAcceptableHere
333
SIP User-Agent Header Format

You can define the text to display in the SIP User-Agent header. You can use macros to include information
specific to the unit.
You can also define the same information in the HTTP User-Agent header. See HTTP User-Agent Header
Format on page 43 for more details.
To set the SIP User-Agent header format:

1.
In the sipEpMIB, set the User-Agent header format in the interopUaHeaderFormat variable.
sipEp.interopUaHeaderFormat="Value"
where Value may contain any text, as well as one or more of the following macros:
Table 317: Macros Supported
Macro
Description
%version%
Application version.
%mac%
MAC address.
%product% Product name.

%profile%
Profile.
%%
Insert the % character.
For instance, the default value is:

%product%/v%version% %profile%
SIP INFO Without Content Answer

You can define the response of the Mediatrix unit to a received SIP INFO with no message body for an existing
call.
RFC 2976 recommends that a 200 OK response MUST be sent for an INFO request with no message body if
the INFO request was successfully received for an existing call.
Table 318: Reject Code for Unsupported SDP Offer Parameters
Parameter
Description
UnsupportedMediaType The unit responds with the SIP error code 415 'Unsupported Media Type'.
Ok
The unit responds with a 200 OK.
To define the SIP INFO Without Content Answer behaviour:

1.
In the sipEpMIB, set the interopSipInfoWithoutContentAnswer variable with the proper

behaviour.
sipEp.interopSipInfoWithoutContentAnswer="Value"

Table 319: SIP INFO Values
Value
334
Meaning
200
Ok
415
UnsupportedMediaType
Keep Alive Option Format

You can configure the Keep Alive OPTION requests format.
Table 320: Keep Alive Option Format Parameters
Parameter
Description
ShortFrom
The unit sends the OPTION request with the standard format with only the units
IP address in the from header. This is the default.
FullFrom
The unit sends the OPTION request with the standard format with the first
registered username and IP address in the from header.
Note: The SipEp service must be restarted to apply a new username to the Keep Alive.
To set the keep alive option format:

1.
In the sipEpMIB, locate the InteropGroup folder.
2.
Set the InteropKeepAliveOptionFormat variable with the proper value.

sipEp.InteropKeepAliveOptionFormat="Value"

Figure 152: Keep Alive Option Format Values
Value
Meaning
100
ShortFrom
200
FullFrom
Unsupported Content-Type
You can define the behaviour of the Mediatrix unit upon reception of a SIP packet containing multiple
unsupported Content-Type in the payload.
Table 321: Unsupported Content-Type Parameters
Parameter
Description
Reject
Unsupported Content-Type are rejected.
Allow
Unsupported Content-Type are allowed and ignored if at least one Content-Type

is supported.
Ignore
Unsupported Content-Type are ignored.
Note: When ignored, unsupported Content-Type are treated as if they were not present in the packet.
To define the unsupported Content-Type behaviour:

1.
In the sipEpMIB, set the interopUnsupportedContentType variable with the proper behaviour.
sipEp.interopUnsupportedContentType="Value"
335

Table 322: Unsupported Content-Type Values
Value Meaning
336
100
Reject
200
Allow
300
Ignore
H A P T E R
30
Miscellaneous SIP Parameters

This chapter describes miscellaneous SIP parameters you can set:
SIP penalty box parameters

How to override the default mapping of error causes defined in RFC 3398.
Additional Headers
PRACK
Session Refresh
SIP Gateway Configuration
SIP Blind Transfer Method
Diversion Configuration
DNS Configuration
Event Handling Configuration
Messaging Subscription
SIP Penalty Box

The penalty box feature is used when a given host FQDN resolves to a non-responding address. When the
address times out, it is put into the penalty box for a given amount of time. During that time, the address in
question is considered as non-responding for all requests.
This feature is most useful when using DNS requests returning multiple or varying server addresses. It makes
sure that, when a host is down, users wait a minimal amount of time before trying a secondary host.
When enabled, this feature takes effect immediately on the next call attempt.
The penalty box feature is applied only when using UDP or TCP connections established with a FQDN. A
similar penalty box feature for the TLS persistent connections is available via the TLS Persistent Retry Interval
parameter. See SIP Transport Type on page 307 for more details.
Note: The Penality Box feature is disabled when a gateway of type "endpoint" is configured in the gateway.
Penalty Box vs Transport Types

Media5 recommends to use this feature with care when supporting multiple transports (see Chapter 28 - SIP
Transport Parameters on page 307 for more details) or you may experience unwanted behaviours.
When the Mediatrix unit must send a packet, it retrieves the destination from the packet. If the destination
address does not specify a transport to use and does not have a DNS SRV entry that configures which
transport to use, then the Mediatrix unit tries all transports it supports, starting with UDP. If this fails, it tries
with TCP. The unit begins with UDP because all SIP implementations must support this transport, while the
mandatory support of TCP was only introduced in RFC 3261.
Note: It is not the destination itself that is placed in the penalty box, but the combination of address, port
and transport. When a host is in the penalty box, it is never used to try to connect to a remote host unless it
is the last choice for the Mediatrix unit and there are no more options to try after this host.
Lets say for instance that the Mediatrix unit supports both the UDP and TCP transports. It tries to reach
endpoint B for which the destination address does not specify a transport and there is no DNS SRV entry to
specify which transports to use in which order. It turns out that this endpoint B is also down. In this case, the
337
Chapter 30 - Miscellaneous SIP Parameters
SIP Penalty Box
Mediatrix unit first tries to contact endpoint B via UDP. After a timeout period, UDP is placed in the penalty
box and the unit then tries to contact endpoint B via TCP. This fails as well and TCP is also placed in the
penalty box.
Now, lets assume endpoint B comes back to life and the Mediatrix unit tries again to contact it before UDP
and TCP are released from the penalty box. First, the unit tries UDP, but it is currently in the penalty box and
there is another transport left to try. The Mediatrix unit skips over UDP and tries the next target, which is TCP.
Again, TCP is still in the penalty box, but this time, it is the last target the Mediatrix unit can try, so penalty box
or not, TCP is used all the same to try to contact endpoint B.
There is a problem if endpoint B only supports UDP (RFC 2543-based implementation). Endpoint B is up,
but the Mediatrix unit still cannot contact it: with UDP and TCP in the penalty box, the unit only tries to contact
endpoint B via its last choice, which is TCP.
The same scenario would not have any problem if the penalty box feature was disabled. Another option is to
disable TCP in the Mediatrix unit, which makes UDP the only possible choice for the unit and forces to use
UDP even if it is in the penalty box.
You must fully understand the above problem before configuring this feature. Mixing endpoints that do not
support the same set of transports with this feature enabled can lead to the above problems, so it is suggested
to either properly configure SRV records for the hosts that can be reached or be sure that all hosts on the
network support the same transport set before enabling this feature.
Penalty Box Configuration

The following steps describe how to configure the penalty box feature.
To set the SIP penalty box parameters:

1.
In the web interface, click the SIP link, then the Misc sub-link.
Figure 153: SIP Configuration Misc Web Page
2
3
2.
In the Penalty Box section, enable the SIP penalty box feature by selecting Enable in the Penalty
Box Activation drop-down menu.
The penalty box is always active. This means that even if the feature is disabled, IP addresses are
marked as invalid, but they are still tried. This has the advantage that when the feature is enabled,
IP addresses that were already marked as invalid are instantly put into the penalty box.
3.
Set the amount of time, in seconds, that a host spends in the penalty box in the Penalty Box Time
field.
Changing the value does not affect IP addresses that are already in the penalty box. It only affects
new entries in the penalty box.
4.
338
Error Mapping
Error Mapping
Standards Supported
RFC 3398: Integrated Services Digital Network (ISDN) User

Part (ISUP) to Session Initiation Protocol (SIP) Mappinga
a. Only the ISDN to SIP error mapping is supported.
You can override the default mapping of error causes defined in RFC 3398.The web interface offers two
sections:
The SIP To Cause Error Mapping section allows you to override the default mapping for SIP
code to ISDN cause.
The Cause To SIP Error Mapping section allows you to override the default mapping for ISDN
cause to SIP code.
The following standard SIP codes are available:

400: Bad Request
414: Request-URI too long
485: Ambiguous
401: Unauthorized
415: Unsupported media type
486: Busy here
402: Payment required
416: Unsupported URI Scheme
500: Server internal error
403: Forbidden
420: Bad extension
501: Not implemented
404: Not found
421: Extension Required
502: Bad gateway
405: Method not allowed
423: Interval Too Brief
503: Service unavailable
406: Not acceptable
480: Temporarily unavailable
504: Server time-out
407: Proxy authentication

required
481: Call/Transaction Does not

Exist
504: Version Not Supported
408: Request timeout
482: Loop Detected
600: Busy everywhere
410: Gone
483: Too many hops
603: Decline
413: Request Entity too long
484: Address incomplete
604: Does not exist anywhere
513: Message Too Large
You can also map any other custom code between 400 and 699.
339
Error Mapping
The following standard ISDN cause numbers specified in Q.931 are available:
Normal event:
Service or option not implemented:
1: Unassigned (unallocated) number.
65: Bearer capability not implemented.
2: No route to specified transit network.
66: Channel type not implemented.
3: No route to destination.
69: Requested facility not implemented.
6: Channel unacceptable.
70: Only restricted digital information bearer.
7: Call awarded and being delivered in an

established channel.
79: Service or option not implemented,

unspecified.
17: User busy.

18: No user responding.
Invalid Message
19: User alerting, no answer.
81: Invalid call reference value.
20: Subscriber absent.
82: Identified channel does not exist.
21: Call rejected.

22: Number changed.
83: A suspended call exists, but this call identity

does not.
23: Redirection to new destination.
84: Call identity in use.
26: Non-selected user clearing.
85: No call suspended.
27: Destination out of order.
86: Call having the requested call identity has been

cleared.
28: Invalid number format (incomplete number).

29: Facility rejected.
30: Response to STATUS ENQUIRY.
31: Normal, unspecified.
Resource unavailable:
34: No circuit/channel available.
38: Network out of order.
41: Temporary failure.
42: Switching equipment congestion.
43: Access information discarded.
44: Requested circuit/channel not available.
47: Resource unavailable, unspecified.
87: user not member of CUG.

88: Incompatible destination.
91: Invalid transit network selection.
95: Invalid message, unspecified.
Protocol error
96: Mandatory information element is missing.
97: Message type non-existent or not
implemented.
98: Message not compatible with call state or
message type non-existent or not implemented.
99: Information element non-existent or not
implemented.
100: Invalid information element contents.
Service or option not available:
101: Message not compatible with call state.
55: Incoming calls barred within CUG.
102: Recovery on time expiry.
57: Bearer capability not authorized.
111: Protocol error, unspecified.
58: Bearer capability not presently available.
Interworking
63: Service or option not available, unspecified.
127: Interworking, unspecified
You can also map any other custom code between 1 and 127.
SIP to Cause Error Mapping

This section describes how to override the default mapping of ISDN error causes.
340
Error Mapping
To override the default mapping of ISDN error causes:

1.
In the SIP To Cause Error Mapping section of the Misc page, click the
button to add a new row.
Figure 154: SIP To Cause Error Mapping Section

1
This brings you to the Configure New SIP To Cause Error Mapping panel.
2.
Enter the SIP code in the SIP Code field, then the corresponding ISDN cause number in the Cause
column.
You can use the Suggestion columns drop-down menu to select between available code values.
Figure 155: Configure New SIP To Cause Error Mapping Panel
3.
Click Submit.
This brings you back to the main Misc web page.
You can delete an existing row by clicking the
button.
You can modify the Cause value by typing a new code in the field. See SIP To Cause Default Error
Mapping on page 341 for the default mappings as per RFC 3398.
4.
SIP To Cause Default Error Mapping

Table 323 lists the default mappings as per RFC 3398.
Table 323: SIP To Cause Default Error Mapping
SIP Response Received
Cause Value
400 Bad Request
41
Temporary Failure
401 Unauthorized
21
Call rejected
402 Payment required
21
Call rejected
403 Forbidden
21
Call rejected
404 Not found
Unallocated number
405 Method not allowed
63
Service or option unavailable
406 Not acceptable
79
Service/option not implemented
407 Proxy authentication required
21
Call rejected
408 Request timeout
102 Recovery on timer expiry
410 Gone
22
413 Request Entity too long
127 Interworking
414 Request-URI too long
127 Interworking
Number changed (w/o diagnostic)
341
Error Mapping
Table 323: SIP To Cause Default Error Mapping (Continued)

SIP Response Received
Cause Value
415 Unsupported media type
79
Service/option not implemented
416 Unsupported URI Scheme
127 Interworking
420 Bad extension
127 Interworking
421 Extension Required
127 Interworking
423 Interval Too Brief
127 Interworking
480 Temporarily unavailable
18
No user responding
481 Call/Transaction Does not Exist
41
Temporary Failure
482 Loop Detected
25
Exchange - routing error
483 Too many hops
25
Exchange - routing error
484 Address incomplete
28
Invalid Number Format
485 Ambiguous
Unallocated number
486 Busy here
17
User busy
500 Server internal error
41
Temporary failure
501 Not implemented
79
Not implemented, unspecified
502 Bad gateway
38
Network out of order
503 Service unavailable
41
Temporary failure
504 Server time-out
102 Recovery on timer expiry
504 Version Not Supported
127 Interworking
513 Message Too Large
127 Interworking
600 Busy everywhere
17
User busy
603 Decline
21
Call rejected
604 Does not exist anywhere
Unallocated number
Cause to SIP Error Mapping

This section describes how to override the default mapping of SIP codes.
To override the default mapping of SIP codes:

1.
In the Cause To SIP Error Mapping section of the Misc page, click the
button to add a new row.
Figure 156: Cause To SIP Error Mapping Section

1
This brings you to the Configure New Cause To SIP Error Mapping panel.
2.
Enter the ISDN cause number in the Cause column, then the corresponding SIP code in the SIP
Code field.
You can use the Suggestion columns drop-down menu to select between available code values.
342
Error Mapping
Figure 157: Configure New Cause To SIP Error Mapping Panel
3.
Click Submit.
This brings you back to the main Misc web page.
You can delete an existing row by clicking the
button.
You can modify the SIP Code value by typing a new code in the field. See Cause To SIP Default
Error Mapping on page 343 for the default mappings as per RFC 3398.
4.
Cause To SIP Default Error Mapping

Table 324 lists the default mappings as per RFC 3398.
Table 324: Cause To SIP Default Error Mapping
ISUP Cause Value
SIP Response
Normal Event
unallocated number
404 Not Found
no route to network
404 Not Found
no route to destination
404 Not Found
16
normal call clearing
---
17
user busy
486 Busy Here
18
no user responding
408 Request Timeout
19
no answer from the user
20
subscriber absent
21
call rejected
403 Forbidden
22
number changed (w/o diagnostic)
410 Gone
22
number changed (w/ diagnostic)
301 Moved Permanently
23
redirection to new destination
410 Gone
26
non-selected user clearing
404 Not Found
27
destination out of order
502 Bad Gateway
28
address incomplete
484 Address incomplete
29
facility rejected
501 Not implemented
31
normal unspecified
BYE or CANCEL
Resource Unavailable
34
no circuit available
38
network out of order
343
Error Mapping
Table 324: Cause To SIP Default Error Mapping (Continued)

ISUP Cause Value
SIP Response
41
temporary failure
42
switching equipment congestion
47
resource unavailable

Service or Option not Available
55
incoming calls barred within CUG
403 Forbidden
57
bearer capability not authorized
403 Forbidden
58
bearer capability not presently available 503 Service unavailable

Service or Option not Implemented
65
bearer capability not implemented
488 Not Acceptable Here
70
only restricted digital available
488 Not Acceptable Here
79
service or option not implemented
501 Not implemented
Invalid message
87
user not member of CUG
403 Forbidden
88
incompatible destination

Protocol error
102 recovery of timer expiry
504 Gateway timeout
111
protocol error
Interworking
127 interworking unspecified
344
Additional Headers
Additional Headers
You can define whether or not the Mediatrix unit uses additional SIP headers.
To use additional SIP headers:

1.
In the Additional Headers section of the Misc page, select the method to use in the Reason Header
Support drop-down menu.
Figure 158: Reason Header Section
1
2
Table 325: Reason Header Support Parameters

Parameter
2.
Description
None
Silently ignores any incoming reason headers and does not

send the reason header.
SendQ850
Silently ignores incoming reason codes and sends the SIP

reason code when the original Q.850 code is available. The
reason code sent is not affected by the entries in the Error
Mapping SIP To Cause table.
ReceiveQ850
Uses the incoming Q.850 reason cause header. When

received, the reason code supersedes any entrie s in the
Error Mapping SIP To Cause table.
SendReceiveQ850
Uses the incoming Q.850 reason cause header and sends

the SIP reason code when the original Q.850 code is
available. When received, the reason code supersedes any
entries in the Error Mapping SIP To Cause table. The
reason code sent is not affected by the entries in the Error
Mapping SIP To Cause table.
Select how the Referred-By header is used when participating in a transfer in the Referred-By
Table 326: Referred-By Support Parameters
Parameter
None
Description
When acting as the transferor (sending the REFER), the
REFER does not contain a Referred-By header.
When acting as the transferee (receiving the REFER and
sending the INVITE to the target), the Referred-By header is
not copied from the REFER to the INVITE.
HeaderOnly
When acting as the transferor (sending the REFER), the

Referred-By header contains the SIP URI of the transferor.
When acting as the transferee (receiving the REFER and
sending the INVITE to the target), the Referred-By header is
copied from the REFER to the INVITE.
3.
4.
Set the interval, in seconds, at which SIP Keep Alive requests using SIP OPTIONS or Ping are sent
to verify the server status in the Keep Alive Interval field.
345
PRACK
PRACK
Standards Supported
RFC 3262: Reliability of Provisional Responses in the

Session Initiation Protocol (SIP)
RFC 3311: The Session Initiation Protocol (SIP) UPDATE

Methoda
a. Only support receiving UPDATE. Sending an UPDATE is not supported.
The Mediatrix unit supports reliable provisional responses (PRACK) as per RFC 3262. You can define this
support when acting as a user agent client and when acting as a user agent server.
The Mediatrix unit supports the UPDATE as per RFC 3311; however, its support is limited to reception.
To define the PRACK support:

1.
In the PRACK section of the Misc page, define the support of RFC 3262 (PRACK) when acting as
a user agent server in the UAS PRACK Support drop-down menu.
Figure 159: PRACK Section
1
2
Table 327: PRACK User Agent Server Parameters

Parameter
Description
Unsupported
The option tag 100rel is ignored if present in the Supported

or Required header of received initial INVITEs and
provisional responses are not sent reliably as per RFC 3261.
Supported
If the option tag 100rel is present in the Supported or

Required header of initial received INVITEs, provisional
responses are sent reliably as per RFC 3262 by adding the
option tag 100rel to the Require header.
Receiving an UPDATE request to negotiate early media is supported only if you have selected
Supported.
2.
Define the support of RFC 3262 (PRACK) when acting as user agent client in the UAC PRACK
Table 328: PRACK User Agent Client Parameters
Parameter
346
Description
Unsupported
The option tag 100rel is not added in the Supported or

Required header of sent INVITEs as per RFC 3261. If the
provisional response contains a Require header field with
the option tag 100rel, the indication is ignored and no
PRACK are sent.
Supported
The option tag 100rel is added to the Supported header of

sent initial INVITEs as per RFC 3262. If the received
the option tag 100rel, the response is sent reliably using
the PRACK method.
Session Refresh
Table 328: PRACK User Agent Client Parameters (Continued)

Parameter
Description
Required
3.
The option tag 100rel is added to the Require header of

sent initial INVITEs as per RFC 3262. If the received
the option tag 100rel, the response is sent reliably using
the PRACK method.
Forked Provisional Responses Behaviour


by using the CLI
You can configure the unit's behaviour when receiving forked provisional answers. This configuration has no
effect if the UAC PRACK Support drop-down menu is set to a value other than Unsupported.
Table 329: Forked Provisional Responses Behaviour Parameters
Value
Description
InterpretFirst
Only the first provisional answer is interpreted. Following responses do not change the
state of the call and the SDP is ignored if present.
InterpretAll
Each forked provisional response received by the unit is interpreted replacing the previous
one. If the response contains SDP, it replaces previous answers if any.
To set the forked provisional responses behaviour:

1.
In the sipEpMIB, define the behaviour in the interopForkedProvisionalResponsesBehavior

variable.
sipEp.interopForkedProvisionalResponsesBehavior=[value]

Table 330: Forked Provisional Responses Behaviour Values
Value
Meaning
100
InterpretFirst
200
InterpretAll
Session Refresh
This section allows you to define session refresh and session timers parameters. Session timers apply to the
whole unit.
347
Session Refresh
To set Session Refresh information:

1.
In the Session Refresh section of the Misc page, define whether to enable or disable the session
expiration services in the Session Refresh Timer Enable drop-down menu.
Figure 160: Session Refresh Section
1
2
1
2
Disabling this service is not recommended since it will make 'dead' calls impossible to detect.
See Background Information on page 348 for more details.
2.
Set the session timer minimum expiration delay, in seconds, in the Minimum Expiration Delay (s)
field.
This is the minimum value, in seconds, for the periodical session refreshes. It must be equal to or
smaller than the maximum value. This value is reflected in the Min-SE header.
The Min-SE value is a threshold under which proxies and user agents on the signalling path are not
allowed to go. Increasing the minimum helps to reduce network traffic, but also makes dead calls
longer to detect.
3.
Set the session timer maximum expiration delay, in seconds, in the Maximum Expiration Delay (s)
field.
This is the suggested maximum time, in seconds, for the periodical session refreshes. It must be
equal to or greater than the minimum value. This value is reflected in the Session-Expires header.
Increasing the maximum helps to reduce network traffic, but also makes dead calls longer to
detect.
Note: When the Maximum Expiration Delay value is lower than the Minimum Expiration Delay value, the
minimum and maximum expiration delay values in INVITE packets are the same as the value set in the
Minimum Expiration Delay field.
4.
Select the method used for sending Session Refresh Requests in the Use UPDATE for Session
Refresh parameter.
Table 331: UPDATE for Session Refresh Parameters
Parameter
Description
ReInvite
Session Refresh Requests are sent with the INVITE method.
Update
Session Refresh Requests are sent with the UPDATE

method.
Session Refresh Requests can be received via both methods, regardless of how this parameter is
configured.
5.
Background Information
The following explains how the session timers are used.
What is the session timer extension?

The session timer extension allows detecting the premature end of a call caused by a network problem or a
peers failure by resending a refresh request at every n seconds. This refresh request is either an reINVITE or
an UPDATE, according to the configuration of the Session Refresh Request Method parameter (see PRACK
on page 346).
348
A successful response (200 OK) to this refresh request indicates that the peer is still alive and reachable. A
timeout to this refresh request may mean that there are problems in the signalling path or that the peer is no
longer available. In that case, the call is shut down by using normal SIP means.
SDP in Session Timer reINVITEs or UPDATEs

The reINVITE is sent with the last SDP that was negotiated. Receiving a session timer reINVITE should not
modify the connection characteristics.
If the reINVITE method is used, it is sent with the last SDP that was negotiated. Reception of a session timer
reINVITE should not modify the connection characteristics. If the UPDATE method is used, it is sent without
any SDP offer. REMPLACER
Relation Between Minimum and Maximum Values

A user agent that receives a Session-Expires header whose value is smaller than the minimum it is willing to
accept replies a 422 Timer too low to the INVITE and terminates the call. The phone does not ring.
It is up to the caller to decide what to do when it receives a 422 to its INVITE. The Mediatrix unit will
automatically retry the INVITE, with a Session-Expires value equal to the minimum value that the user agent
server was ready to accept (located in the Min-SE header). This means that the maximum value as set in the
Mediatrix unit might not be followed. This has the advantageous effect of establishing the call even if the two
endpoints have conflicting values. The Mediatrix unit will also keep retrying as long as it gets 422 answers with
different Min-SE values.
Who Refreshes the Session?

Sending a session timer reINVITE or UPDATE is referred to as refreshing the session. Normally, the user
agent server that receives the INVITE has the last word on who refreshes. The Mediatrix unit always lets the
user agent client (caller) perform the refreshes if the caller supports session timers. In the case where the
caller does not support session timers, the Mediatrix unit assumes the role of the refresher.

You can define whether or not to override the SIP domain used.
To set the SIP domain override:

1.
In the SIP Gateway Configuration section of the Misc page, define whether or not to override the
SIP domain used in the SIP Domain field.
If not empty, it overrides the home domain proxy (Proxy Host field of the Servers sub-page SIP
Default Servers section SIP Servers Configuration on page 284) in the address of record and the
request-URI. When it overrides the home domain proxy in the request-URI, the request-URI also
contains a maddr parameter with the resolved home domain proxy to make sure the requests are
routable.
Figure 161: SIP Gateway Configuration Section
1
2
2.
349

You can set the SIP transfer method when an endpoint is acting as the transferor in a blind transfer scenario.
To set the SIP blind transfer method:

1.
In the SIP Transfer section of the Misc page, set the Blind Transfer Method.
Figure 162: SIP Transfer Section
1
Table 332: SIP Blind Transfer Method Parameters

Parameter
2.
Description
Semi Attended
When blind transfer is invoked by the transferor, the device

sends immediately a REFER (it does not wait for the
reception of the 200OK response). This allows the call
transfer to be executed before the transfer-target answers.
The transferee and the target are then connected together
early and the transferee can hear the ringback from the
target until the target answers.
Semi Attended Confirmed
When blind transfer is invoked by the transferor, the device

waits for reception of the 200 OK from the transfer-target
before sending a REFER to the transferee.
Semi Attended Cancelled
This method is similar to the Semi Attended Transfer

method except that the INVITE sent to the transfer-target is
cancelled when the blind transfer is invoked before
receiving a 200OK (INVITE). In case where the transferor
receives a 200OK (INVITE) from the transfer-target before
receiving of a 487 Request Terminated, the transfer stays
ongoing and it behaves as a Semi Attended Confirmed
Transfer.
Diversion Configuration
You can define call diversion parameters.
Note: The Diversion feature is not available in the NI2 and QSIG signalling protocols. See PRI
Configuration on page 186 for more details on how to configure the signalling protocol.
To set the call diversion parameters:

1.
In the Diversion section of the Misc page, set the Methcd drop-down menu with the SIP method
used to receive/send call diversion information in an INVITE.
The gateways available are those defined in SIP Gateways Configuration on page 279.
350
DNS Configuration
Figure 163: Diversion Configuration Section

1
Table 333: Diversion Parameters

Parameter
2.
Description
None
No diversion information is sent in SIP messages.
Diversion Header
The SIP gateway supports the SIP header 'Diversion' (RFC

5806) in received and sent INVITEs, as well as in 302
messages.
DNS Configuration
You can define DNS-related parameters.
To set the DNS-related parameters:

1.
In the DNS section of the Misc page, set the Supported DNS Queries drop-down menu with the type
of DNS queries that the SipEp service supports and uses.
Figure 164: DNS Configuration Section
1
Table 334: DNS Parameters

Parameter
2.
Description
Address
Sends only Address requests (type A).
SRV
Sends a Service request (type SRV) first and then Address

requests (type A) if needed.
NAPTR
Sends a Naming Authority Pointer request (type NAPTR)

first and then Service requests (type SRV) or Address
requests (type A) as needed.

The Mediatrix unit supports receiving event handling Notifications to start a remote reboot or a sync of
configuration for specific endpoint(s). The event handling Notifications "reboot" or "check-sync" is not
specified in an Allow-Events header. The Mediatrix unit supports the Notify without subscription.
It is recommended to use these event handling notifications only when the SIP transport is secure (TLS) or
when the firewall filters the requests sent to the unit.
351
To set the event handling parameters:

1.
In the Event Handling section of the Misc page, set the Reboot column of each available gateway
to define whether or not the SIP gateway can start a remote reboot via a SIP NOTIFY Event.
This specifies whether a remote reboot via a SIP NOTIFY message event is supported or not for a
specific SIP gateway.
Figure 165: Event Handling Parameters
1
Table 335: Reboot Event Handling Parameters

Parameter
2.
Description
Rejected
The "reboot" notification is rejected on reception.
Restart
When receiving a "reboot" notification, a restart of the unit is

done.
Set the CheckSync column of each available gateway to define whether or not the SIP gateway can
transfer and run a configuration file via a SIP NOTIFY Event.
This specifies whether a transfer script via a SIP NOTIFY message event is supported or not for a
Table 336: CheckSync Event Handling Parameters
Parameter
3.
352
Description
Rejected
The "check-sync" notification is rejected on reception.
TransferScript
When receiving a "check-sync" notification, the

Conf.ConfiguredScriptsTransferAndRun command is
executed.
The Mediatrix unit allows you to add the username in the Request-URI of SUBSCRIBEs it sends.
To set the messaging subscription:

1.
In the Messaging Subscription section of the Misc page, set the Username in Request-URI dropdown menu, set whether or not the unit adds the username in the request URI of MWI SUBSCRIBE
requests.
Figure 166: Messaging Subscription Parameters
1
Table 337: Messaging Subscription Parameters

Parameter
2.
Description
Enable
The unit adds the username in the Request-URI of sent MWI

SUBSCRIBE requests.
Disable
No username in Request-URI of MWI SUBSCRIBE

requests sent by the unit.
Advice of Charge Configuration

The Mediatrix unit allows you to configure the Advice Of Charge (AOC) to send the current charge (D)uring a
call via an AOC-D message or the total charge at the (E)nd of call via an AOC-E message.
Note: The AOC feature is not available LP/4100/C7 Series models.
To set the AOC parameters:

1.
In the AOC section of the Misc page, set the AOC-D Support and AOC-E Support parameters for
each available gateway to define if and how AOC-D and AOC-E messages are sent.
Figure 167: AOC-D and AOC-E Configuration Parameters
1
Table 338: AOC-D and AOC-E Configuration Parameters

Parameter
2.
Description
Disabled
No AOC information is sent. Received AOC information is

discarded.
Transparent
AOC information is forwarded to the peer interface if AOC

353
Additional DNS Parameters
Additional DNS Parameters


by using the CLI
DNS Failure Concealment

You can configure the way failed DNS queries are handled.
Table 339: DNS Failure Concealment Parameters
Parameter
Description
None
When a DNS query times out or returns an error, the SIP transaction fails.
OnNoResol
ution
When a DNS query times out or returns an error, the result from the last successful query for
the same FQDN is used.
To set the DNS failure concealment parameter:

1.
In the sipMIB, locate the DnsGroup folder.
2.
Set the DNS failure concealment configuration in the DnsFailureConcealment variable.

sip.DnsFailureConcealment="Value"

Table 340: DNS Failure Concealment Values
Value
Meaning
100
None
300
OnNoResolution
Note: This variable applies only to gateway type Endpoint; it has no effect on Trunk gateways, and
therefore, DNS failure concealment is always considered to be none.
354
Media Parameters
H A P T E R
31
Voice & Fax Codecs

Configuration
This chapter describes the voice and fax codec configuration parameters.
Codec descriptions.
How to enable and disable the codecs.
How to set the individual codecs parameters.
Standards Supported
RFC 3550: RTP: A Transport Protocol for Real-Time

Applications
RFC 3551: RTP Profile for Audio and Video Conferences with
Minimal Control
Codec Descriptions
The Mediatrix unit supports several voice and fax codecs. It also supports unicast applications, but not
multicast ones. All voice transport is done over UDP.
All the endpoints of the Mediatrix unit can simultaneously use the same codec (for instance, G.711 PCMA), or
a mix of any of the supported codecs. Set and enable these codecs for each endpoint.
Table 341: Codecs Comparison
Compression
Voice Quality
G.711
None
Excellent
G.723.1a
Highest
Good
G.726
Medium
Fair
G.729a/ab
High
Fair/Good
a. This codec is not available on the Mediatrix C7 Series and

4102S models.
G.711 A-Law and -Law

Standards Supported
ITU-T Recommendation G.711
The audio data is encoded as 8 bits per sample, after logarithmic scaling.
Table 342: G.711 Features
Feature
Packetization time
Description
Range of 10 ms to 30 ms with increments of 10 ms. See G.711 Codec
Parameters on page 365 for more details.
For the reception, the range is extended from 10 ms to 100 ms with
increments of 1 ms only if the stream is not encrypted (SRTP).
Voice Activity Detection (VAD)
Two levels of detection are available: transparent or conservative. See

Generic Voice Activity Detection (VAD) on page 364 for more details.
357
Codec Descriptions
Table 342: G.711 Features (Continued)

Feature
Description
Comfort noise
Uses custom comfort noise as defined in RFC 3389.
Available for voice
Yes
Available for fax
Yes
Available for modem
Yes
G.723.1
Standards Supported
ITU-T Recommendation G.723.1a
a. This codec is not available on the Mediatrix C7 Series and 4102S models.
Dual-rate speech coder for multimedia communications transmitting at 5.3 kbit/s and 6.3 kbit/s. This
Recommendation specifies a coded representation that can be used to compress the speech signal
component of multi-media services at a very low bit rate. The audio is encoded in 30 ms frames.
A G.723.1 frame can be one of three sizes: 24 octets (6.3 kb/s frame), 20 octets (5.3 kb/s frame), or 4 octets.
These 4-octet frames are called SID frames (Silence Insertion Descriptor) and are used to specify comfort
noise parameters.
Table 343: G.723.1 Features
Feature
Packetization time
Description
Supports the annex A, which is the built-in support of VAD in G.723.1.
Payload type
Available for voice
Yes
Available for fax
No
Available for modem
No
G.726
Standards Supported
ITU-T Recommendation G.726: 40, 32, 24, 16 kbit/s adaptive

differential pulse code modulation (ADPCM)
Algorithm recommended for conversion of a single 64 kbit/s A-law or U-law PCM channel encoded at 8000
samples/s to and from a 40, 32, 24, or 16 kbit/s channel. The conversion is applied to the PCM stream using
an Adaptive Differential Pulse Code Modulation (ADPCM) transcoding technique.
Feature
Packetization time
Description
Range of 10 ms to 30 ms with increments of 10 ms. See G.726 Codecs
358
Two levels of detection are available: transparent or conservative. See

Generic Voice Activity Detection (VAD) on page 364 for more details.
Codec Descriptions
Table 344: G.726 Features (Continued)

Feature
Description
Comfort noise
Uses custom comfort noise as defined in RFC 3389.
Payload type
Configurable as per G.726 Codecs Parameters on page 367.
Available for voice
Yes
Available for fax
Yes (32 kbps and 40 kbps)
Available for modem
Yes (32 kbps and 40 kbps)
G.729
Standards Supported
Coding of speech at 8 kbit/s using conjugate structure-algebraic code excited linear prediction (CS-ACELP).
For all data rates, the sampling frequency (and RTP timestamp clock rate) is 8000 Hz.
A voice activity detector (VAD) and comfort noise generator (CNG) algorithm in Annex B of G.729 is
recommended for digital simultaneous voice and data applications; they can be used in conjunction with G.729
or G.729 Annex A. A G.729 or G.729 Annex A frame contains 10 octets, while the G.729 Annex B comfort
noise frame occupies 2 octets.
The Mediatrix unit supports G.729A and G.729AB for encoding and G.729, G.729A and G.729AB for
decoding.
Feature
Packetization time
Description
For reception, the range is extended from 10 ms to 100 ms with
Supports the annex B, which is the built-in support of VAD in G.729. See
G.729 Codec Parameters on page 369 for more details.
Payload type
18
Available for voice
Yes
Available for fax
No
Available for modem
No
359
Codec Descriptions
Clear Mode
Standards Supported
RFC 4040: RTP Payload Format for a 64 kbit/s Transparent

Call
The Clear Mode codec is similar to the G.711 codec but without any modification of the 64 kbit/s payload (no
encoding or decoding). The Clear Mode codec thus does not have echo cancellation and a fix jitter buffer.
Clear Mode is a method to carry 64 kbit/s channel data transparently in RTP packets. This codec always uses
the RTP transport.
Table 346: Clear Mode Features
Feature
Packetization time
Description
Range of 10 ms to 30 ms with increments of 10 ms. See Clear Mode
Codec Parameters on page 371 for more details.
N/A
Comfort noise
N/A
Payload type
Configurable as per Clear Mode Codec Parameters on page 371.
Available for voice
Yes
Available for fax
Yes
Available for modem
Yes
Clear Channel
Standards Supported

Call
The Clear Channel codec is similar to the G.711 codec but without any modification of the 64 kbit/s payload
(no encoding or decoding). The Clear Channel codec thus does not have echo cancellation and a fix jitter
buffer. Clear Channel is a method to carry 64 kbit/s channel data transparently in RTP packets. The Clear
Channel codec follows the specification of RFC 4040 and uses the X-CLEAR-CHANNEL mime type instead
of the CLEARMODE mime type.
This codec always uses the RTP transport.
Table 347: Clear Channel Features
Feature
Packetization time
Description
Range of 10 ms to 30 ms with increments of 10 ms. See Clear Channel
Codec Parameters on page 372 for more details.
360
N/A
Comfort noise
N/A
Payload type
Configurable as per Clear Channel Codec Parameters on page 372.
Available for voice
Yes
Available for fax
Yes
Available for modem
Yes
Codec Descriptions
X-CCD Clear Channel

Standards Supported

Call
The Clear Channel codec is similar to the G.711 codec but without any modification of the 64 kbit/s payload
(no encoding or decoding). The X-CCD Clear Channel codec thus does not have echo cancellation and a fix
jitter buffer. The X-CCD Clear Channel is a method to carry 64 kbit/s channel data transparently in RTP
packets. The Clear Channel codec follows the specification of RFC 4040 and uses the X-CCD mime type
instead of the CLEARMODE mime type.
This codec always uses the RTP transport.
Table 348: X-CCD Clear Channel Features
Feature
Description
Packetization time
Range of 10 ms to 100 ms with increments of 1 ms. See X-CCD Clear

Channel Codec Parameters on page 374 for more details.
N/A
Comfort noise
N/A
Payload type
Configurable as per X-CCD Clear Channel Codec Parameters on

page 374.
Available for voice
Yes
Available for fax
Yes
Available for modem
Yes
T.38
Standards Supported
ITU-T Recommendation T.38 version 0
T.38 fax relay is a real-time fax transmission; that is, two fax machines communicating with each other as if
there were a direct phone line between the two. T.38 is called a fax relay, which means that instead of sending
inband fax signals, which implies a loss of signal quality, it sends those fax signals out-of-band in a T.38
payload, so that the remote end can reproduce the signal locally.
Table 349: T.38 Features
Feature
Description
Packetization time
N/A
N/A
Payload type
N/A
Available for voice
No
Available for fax
Yes
Available for modem
No
T.38 is an unsecure protocol, thus will not be used along with secure RTP (SRTP), unless the Allow Unsecure
T.38 with Secure RTP parameter has been set to Enable. See Chapter 32 - Security on page 381 for more
details.
361
Codec Parameters
Codec Parameters
The Codec section allows you to enable or disable the codecs of the Mediatrix unit, as well as access the
codec-specific parameters.
Standards Supported
draft-choudhuri-sip-info-digit-00
ITU-T Recommendation Q.24: Multifrequency push-button

signal reception
RFC 2833: RTP Payload for DTMF Digits, Telephony Tones

and Telephony Signals
RFC 1890: RTP Profile for Audio and Video Conferences with
Minimal Control
Specific configurations that override the default configurations. You can define specific
configurations for each endpoint in your Mediatrix unit.
To enable or disable the codecs:

1.
In the web interface, click the Telephony link, then the CODECS sub-link.
Figure 168: Telephony Codecs Web Page
2
3
2.
Select to which endpoint (interface) you want to apply the changes in the Select Endpoint dropdown menu at the top of the window.
You have the choice between Default and the interfaces of your Mediatrix unit. The number of
interfaces available vary depending on the Mediatrix unit model you have.
You can also perform this operation in the codec-specific pages.
3.
Select whether or not you want to override one or more of the available default codecs parameters
in the Endpoint Specific column of the corresponding codec(s).
This column is available only in the specific endpoints configuration.
4.
Enable one or more codecs for voice transmission by selecting Enable in the Voice column of the
corresponding codec(s).
This indicates if the codec can be selected for voice transmission. If enabled, this codec is listed as
supported for this specific endpoint. Otherwise, it is ignored.
362
Codec vs. Bearer Capabilities Mapping

5.
Enable one or more codecs for data transmission by selecting Enable in the Data column of the
corresponding codec(s).
This indicates if the codec can be selected for data transmission. If enabled, this codec is listed as
6.
Click the
button to access the corresponding codec-specific parameters.
These parameters are described in the following sections.

7.
Codec vs. Bearer Capabilities Mapping

The Codec vs. Bearer Capabilities Mapping section allows you to select the codec to prioritize or select in the
outgoing INVITE when the incoming SETUPs ITC (Information Transfer Capability) matches the configured
one. On the other hand, you can also select the ITC value to set in the outgoing SETUPs bearer capabilities
when the incoming INVITE's codec matches the configured one.
Depending on the mapping type, a codec is prioritized or selected. Prioritized means it is the first in the list of
offered codecs when initiating a call on the IP side and Selected means it is the only codec offered.
This parameter is available only on ISDN interfaces.
You can define up to three mappings.
Note: You can also modify the selection of the preferred codec in the incoming SDP by using the the interop
parameter Codec vs Beareer Capabilities Mapping Preferred Codec Choice in the SIP > Interop > SDP
section. See SDP Interop on page 318 for more details.
To set the codec vs. bearer capabilities mapping

1.
Define if the outgoing codecs priority should reflect the incoming ITC and vice versa in the Enable
column drop-down menu.
Figure 169: Codec vs. Bearer Capabilities Mapping Section
1
Table 350: Outgoing Codec Priority

Parameter
Disable
Enable
Description
The mapping is not applied:
The codecs order in the outgoing INVITE follows the codec priority.
The ITC in the outgoing SETUP is the default one (3.1 kHz) unless a
mapping in the Call Routing table modifies it.
If the ITC value in the incoming SETUP matches the value of the corresponding
ITC drop-down menu, the first codec in the outgoing INVITE is the one set in the
CODEC drop-down menu.
If the first codec in the incoming INVITE matches the value set in the CODEC
drop-down menu, the ITC value in the outgoing SETUP is the one set in the ITC
drop-down menu.
2.
Select a codec in the CODEC column drop-down menu.

363
Generic Voice Activity Detection (VAD)
This is the codec to be prioritized or selected in an outgoing INVITE when the incoming SETUP's
ITC matches the value set in the corresponding ITC drop-down menu. This codec is also checked
against an incoming INVITE's priority codec. If it matches, then the outgoing SETUP's ITC is set to
the value in the corresponding ITC drop-down menu.
See Step 3 for a description of prioritization versus selection of a codec.
3.
Set the mapping type of the codec in the Mapping Type drop-down value.
Table 351: Codec Mapping Type
Parameter
4.
Description
Prioritize
The codec is set on top of the list in an outgoing INVITE when the incoming
SETUP's ITC matches the value set in the corresponding ITC drop-down menu.
Select
The codec is the only one offered in an outgoing INVITE when the incoming
SETUP's ITC matches the value set in the corresponding ITC drop-down menu.
Select an ITC value in the ITC column drop-down menu.

This is the ITC value to be set in the outgoing SETUP when the incoming INVITE's priority codec
matches the value of the corresponding CODEC drop-down menu. This value is also checked
against an incoming SETUP's bearer capabilities. If it matches, then the outgoing INVITE's
prioritized or selected codec is set to the value of the corresponding CODEC drop-down menu.
Table 352: Information Transfer Capability Values
Value
Description
speech
Voice terminals (telephones).
unrestricted
Unrestricted digital information (64 kbps).
3.1Khz
Transparent 3.1 kHz audio channel.
For an incoming ISDN call using UDI (Unrestricted Digital), the Codec vs Bearer Capabilities
Mapping entries with the ITC set to Unrestricted is used only if the codec is Clear Mode, Clear
Channel, XCCD, G.711 a-law or G.711 u-law. G.711 a-law and G.711 u-law are also used only if
they match the ISDN port Preferred Encoding Scheme value (see PRI Configuration on page 186
and BRI Configuration on page 197 for more details).
See Step 3 for a description of prioritization versus selection of a codec.
5.

VAD defines how the Mediatrix unit sends information pertaining to silence. This allows the unit to detect when
the user talks, thus avoiding to send silent RTP packets. This saves on network resources. However, VAD
may affect packets that are not really silent (for instance, cut sounds that are too low). VAD can thus slightly
affect the voice quality.
To set the generic Voice Activity Detection (VAD)

1.
In the Generic Voice Activity Detection (VAD) section, select whether or not you want to override
the VAD parameters set in the Default configuration in the Endpoint Specific drop-down menu.
This menu is available only in the specific endpoints configuration.
Figure 170: Generic Voice Activity Detection (VAD) Section
1
364
G.711 Codec Parameters
2.
Enable the G.711 and G.726 Voice Activity Detection (VAD) by selecting the proper setting in the
Enable (G711 and G726) drop-down menu.
Table 353: G.711/G.726 VAD Settings
Setting
Description
Disable
VAD is not used.
Transparent
VAD is enabled. It has low sensitivity to silence periods.
Conservative
VAD is enabled. It has normal sensitivity to silence periods.
The difference between transparent and conservative is how aggressive the algorithm considers
something as an inactive voice and how fast it stops the voice stream. A setting of conservative is
a little bit more aggressive to react to silence compared to a setting of transparent.
3.

The following are the G.711 codec parameters you can set. There are two sections for G.711:
G.711 a-law
G.711 u-law
These sections use the same parameters, so only one of them is described below.
To set the G.711 codec parameters:

1.
In the CODEC section of the CODECS page, click the

G.711 codec to access the codec-specific parameters.
button at the right of the corresponding
2.
Figure 171: G.711 a-law Section
3
4
5
6
7
8
3.
Select whether or not you want to override the G.711 parameters set in the Default configuration in
the Use Endpoint Specific drop-down menu.
You can also perform this operation in the main CODEC section.
4.
Enable the G.711 codec for voice transmission by selecting Enable in the Voice Transmission dropdown menu.
365
5.
Set the default priority for voice in the Voice Priority field.
This sets the priority between different codecs. Codecs with a higher priority are used first, a priority
of 0 being the lowest priority. For instance, a codec with priority 3 is used before a codec with priority
2. The maximum priority is 10.
The Mediatrix unit uses an internal order for codecs with the same priority.
Note: The codec used is also related to the SIP negotiation. The priority order affects the SIP negotiation,
which decides on the codec to use.
6.
Enable the G.711 codec for data transmission by selecting Enable in the Data Transmission dropdown menu.
7.
Set the default priority for data in the Data Priority field.
8.
Select the minimum and maximum packetization time values for the codec in the Minimum
Packetization Time and Maximum Packetization Time drop-down menus.
The packetization time (also called packetization period or ptime) is the duration, in ms, of the voice
packet. The range is from 10 ms to 30 ms with increments of 10 ms.
For the reception, the range is extended from 10 ms to 100 ms with increments of 1 ms only if the
stream is not encrypted (SRTP).
9.

You can also access the specific parameters of another codec by selecting the codec in the Select
CODEC drop-down menu at the top of the page.

The following are the G.723 codec parameters you can set.
Note that the G.723 codec is not available on the Mediatrix C7 Series and 4102S models.

1.

to access the codec-specific parameters.
button at the right of the G.723 codec
2.
366
G.726 Codecs Parameters
Figure 172: G.723 Section
3
4
5
3.
4.
5.
6.
Select the G.723 bit rate in the Bit Rate drop-down menu.
7.
53 Kbs
63 Kbs
kstream is not encrypted (SRTP).
8.


The following are the G.726 codecs parameters you can set. There are four sections for G.726:
367
G.726 16 Kbps
G.726 24 Kbps
G.726 32 Kbps
G.726 40 Kbps
These sections offer almost the same parameters, except that you cannot use the G.726 16 Kbps and
G.726 24 Kbps codecs for fax transmission.
To set the G.726 codecs parameters:

1.

2.
3
4
5
6
7
3.
4.
Enable the corresponding G.726 codec for voice transmission by selecting Enable in the Voice
Transmission drop-down menu.
5.
6.
Enable the codec for data transmission by selecting Enable in the Data Transmission drop-down
menu.
368
This menu is not available for the G.726 16 Kbps and G.726 24 Kbps codecs.
7.
This field is not available for the G.726 16 Kbps and G.726 24 Kbps codecs.
8.
Set the G.726 actual RTP dynamic payload type used in an initial offer in the Payload Type field.
The payload types available are as per RFC 3551. The values range from 96 to 127. The default
values are as follows:
Table 354: G.726 Default Payload Type
Codec
9.
Default Value
G.726 (16 kbps)
97
G.726 (24 kbps)
98
G.726 (32 kbps)
99
G.726 (40 kbps)
100
Select the minimum and maximum packetization time values for the G.726 codec in the Minimum
10.


The following are the G.729 codec parameters you can set.

1.

button at the right of the G.729 codec
2.
369
3
4
5
6
7
3.
In the G.729 section, select whether or not you want to override the G.729 parameters set in the
Default configuration in the Use Endpoint Specific drop-down menu.
4.
5.
6.
For reception, the range is extended from 10 ms to 100 ms with increments of 10 ms only if the
7.
Select the G.729 Voice Activity Detection (VAD) in the Built-in Voice Activity Detection (VAD) dropdown menu.
Table 355: G.729 VAD
Parameter
Description
Disable
G.729 uses annex A only.
Enable
G.729 annex A is used with annex B. Speech frames are only sent during talkspurts
(periods of audio activity). During silence periods, no speech frames are sent, but
Comfort Noise (CN) packets containing information about background noise may
be sent in accordance with annex B of G.729.
VAD defines how the Mediatrix unit sends information pertaining to silence. This allows the unit to
detect when the user talks, thus avoiding to send silent RTP packets. This saves on network
resources. However, VAD may affect packets that are not really silent (for instance, cut sounds that
are too low). VAD can thus slightly affect the voice quality.
370
Clear Mode Codec Parameters
G.729 has a built-in VAD in its Annex B version. It is recommended for digital simultaneous voice
and data applications and can be used in conjunction with G.729 or G.729 Annex A. A G.729 or
G.729 Annex A frame contains 10 octets, while the G.729 Annex B frame occupies 2 octets. The
CN packets are sent in accordance with annex B of G.729.
8.

Clear Mode Codec Parameters

The following are the Clear Mode codec parameters you can set.
To set the Clear Mode codec parameters:

1.

codec to access the codec-specific parameters.
button at the right of the Clear Mode
2.
Figure 175: Clear Mode Section
3
4
5
6
7
3.
Select whether or not you want to override the Clear Mode parameters set in the Default
configuration in the Use Endpoint Specific drop-down menu.
4.
Enable the Clear Mode codec for voice transmission by selecting Enable in the Voice Transmission
drop-down menu.
5.
371
Clear Channel Codec Parameters
6.
Enable the Clear Mode codec for data transmission by selecting Enable in the Data Transmission
drop-down menu.
7.
8.
Set the Clear Mode RTP dynamic payload type used in an initial offer in the Payload Type field.
value is 125.
9.
10.


The following are the Clear Channel codec parameters you can set.
To set the Clear Channel codec parameters:

1.

codec to access the codec-specific parameters.
button at the right of the Clear Channel
2.
372
Figure 176: Clear Channel Section
3
4
5
6
7
3.
Select whether or not you want to override the Clear Channel parameters set in the Default
4.
Enable the Clear Channel codec for voice transmission by selecting Enable in the Voice
5.
6.
Enable the Clear Channel codec for data transmission by selecting Enable in the Data
7.
8.
Set the Clear Channel RTP dynamic payload type used in an initial offer in the Payload Type field.
value is 125.
9.
373
X-CCD Clear Channel Codec Parameters
10.

X-CCD Clear Channel Codec Parameters

The following are the X-CCD Clear Channel codec parameters you can set.
To set the Clear Channel codec parameters:

1.

button at the right of the X CCD codec
2.
Figure 177: X CCD Section
3
4
5
6
7
3.
Select whether or not you want to override the X CCD parameters set in the Default configuration
in the Endpoint Specific drop-down menu.
4.
Enable the X CCD codec for voice transmission by selecting Enable in the Voice Transmission
drop-down menu.
5.
374
Fax Parameters
6.
Enable the X CCD codec for data transmission by selecting Enable in the Data Transmission dropdown menu.
7.
8.
Set the X CCD RTP dynamic payload type used in an initial offer in the Payload Type field.
value is 125.
9.
10.

Fax Parameters
The Mediatrix unit handles G3 fax transmissions at speeds up to 14.4 kbps. Automatic fax mode detection is
standard on all endpoints. Real-Time Fax Over UDP with the T.38 protocol stack is also available.
A fax call works much like a regular voice call, with the following differences:
1.
The fax codec may be re-negotiated by using a re-INVITE.
2.
The goal of the re-INVITE is to allow both user agents to agree on a fax codec, which is either:
3.
a.
Clear channel (G.711 or G.726) without Echo Cancellation nor Silence Suppression
(automatically disabled).
b.
T.38.
Upon fax termination, if the call is not BYE, the previous voice codec is recovered with another reINVITE.
All endpoints of the Mediatrix unit can simultaneously use the same codec (for instance, T.38), or a mix of any
of the supported codecs. Set and enable these codecs for each endpoint.
Clear Channel Fax

The Mediatrix unit can send faxes in clear channel. The following is a clear channel fax call flow:
375
Fax Parameters
Figure 178: Clear Channel Fax Call Flow

INVITE
[]
m=audio 5006 RTP/AVP 18 0 13
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
[]
m=audio 5004 RTP/AVP 18 0 13
a=rtpmap:18 G729/8000
200 OK
Ringing/Trying
ACK
RTP=G.729 (Voice Call)
Fax Tone Detected

User
Agent
#1
RTP=PCMU (Echo Cancellation + Silence Suppression = disabled)
User
Agent
#2
No re-INVITE!!
There is no need for a re-INVITE since the far end already supports the
data codec (PCMU).
When your SDP capabilities are inserted in a SIP packet, it implies that
you can receive any of these capabilities at any given time without notice.
In this case, both ends should switch to clear channel automatically upon
detection of the fax transmission.
Fax is terminated
BYE
200 OK
DSP Limitation
The Mediatrix unit currently suffers from a limitation of its DSP. Because of this limitation, the voice does not
switch back to the original negotiated codec after a clear channel fax is performed.
The Mediatrix unit cannot detect the end of a clear channel fax, which means that the unit cannot switch back
to the original negotiated codec if this codec was not a clear channel codec, e.g., a session established in
G.729.
When the unit detects a fax, it automatically switches to a negotiated clear channel codec such as PCMU (if
there is no T.38 or if T.38 negotiation failed). Once the fax is terminated, the Mediatrix unit is not notified by
the DSP. The unit thus stays in the clear channel codec and does not switch back to G.729.
T.38 Fax
The Mediatrix unit can send faxes in T.38 mode over UDP. T.38 is used for fax if both units are T.38 capable;
otherwise, transmission in clear channel over G.711 as defined is used (if G.711 -law and/or G.711 A-law are
enabled). If no clear channel codecs are enabled and the other endpoint is not T.38 capable, the fax
transmission fails.
Caution: The Mediatrix unit opens the T.38 channel only after receiving the 200 OK message from the
peer. This means that the Mediatrix unit cannot receive T.38 packets before receiving the 200 OK. Based
on RFC 3264, the T.38 channel should be opened as soon as the unit sends the INVITE message.
The quality of T.38 fax transmissions depends upon the system configuration, type of call control system used,
type of Mediatrix units deployed, as well as the model of fax machines used. Should some of these conditions
be unsatisfactory, performance of T.38 fax transmissions may vary and be reduced below expectations.
Note: Media5 recommends not to use a fax that does not send a CNG tone. If you use such a fax to send
a fax communication to the public network, this might result in a communication failure.
The following is a T.38 fax call flow:
376
Fax Parameters
Figure 179: T.38 Fax Call Flow

INVITE
[]
m=audio 5004 RTP/AVP 0 18 4 8 13
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
[]
m=audio 5006 RTP/AVP 0 18 4 8 13
a=rtpmap:18 G729/8000
Ringing/Trying
200 OK
ACK
Fax Tone Detected

INVITE
User
Agent
#1
[]
m=image 6006 udptl t38
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxVersion:0
a=T38FaxUdpEC:t38UDPRedundancy
200 OK
Trying
[]
m=image 6006 udptl t38
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxVersion:0
a=T38FaxUdpEC:t38UDPRedundancy
User
Agent
#2
ACK
Fax is terminated
[]
m=audio 5004 RTP/AVP 0 18 4 8 13
a=rtpmap:18 G729/8000
INVITE
200 OK
Trying
[]
m=audio 5006 RTP/AVP 0 18 4 8 13
a=rtpmap:18 G729/8000
ACK
BYE
200 OK
T.38 Parameters Configuration

The following are the T.38 codec parameters you can set.
To set the T.38 codec parameters:

1.

2.
Figure 180: T.38 Section
2
4
6
8
5
7
9
10
3.
In the T.38 section, select whether or not you want to override the T.38 parameters set in the Default
377
Fax Parameters

4.
Enable the T.38 codec by selecting Enable in the Enable drop-down menu.
5.
Set the default priority for fax in the Priority field.

Note: Currently, the only T.38 priority accepted is 10. Priority between 1 and 9 is refused.
6.
Set the number of redundancy packets sent with the current packet in the Redundancy Level field.
This is the standard redundancy offered by T.38. Available values range from 1 to 5. Please see
step 7 for additional reliability options for T.38.
7.
Set the T.38 input signal detection threshold in the Detection Threshold drop-down menu.
Lowering the threshold allows detecting lower amplitude fax signals. The following values are
available:
8.
Default: (-26 dB)
Low: (-31 dB)
Lowest: (-43 dB)
For additional reliability, define the number of times T.38 packets are retransmitted in the Frame
Redundancy Level field.
This field is available only in the default endpoint configuration.
This only applies to the T.38 packets where the PrimaryUDPTL contains the following T.38 data
type:
9.
HDLC_SIG_END,
HDLC_FCS_OK_SIG_END,
HDLC_FCS_BAD_SIG_END and
T4_NON_ECM_SIG_END
Define whether or not the Mediatrix unit sends no-signal packets during a T.38 fax transmission in
the No Signal drop-down menu.
This menu is available only in the default endpoint configuration.
When enabled, the unit ensures that, during a T.38 fax transmission, data is sent out at least every
time the No Signal Timeout delay expires. The Mediatrix unit sends no-signal packets if no
meaningful data have been sent for a user-specified period of time.
10.
Set the period, in seconds, at which no-signal packets are sent during a T.38 transmission in the No
Signal Timeout field.
This field is available only in the default endpoint configuration.
No-signal packets are sent out if there are no valid data to send.
11.

378
Fax Parameters
Data Codec Selection Procedure

The Mediatrix unit follows a procedure when selecting data codec. This procedure is the default behaviour of
the Mediatrix unit. Some interop variables may modify this procedure. Tones are detected on the analog ports
only.
379
Fax Parameters
Figure 181: Data Codec Selection Procedure

Voice /
Voiceband
data call
Complete fax/modem codec

selection procedure .
Tones are detected on the telephony ports except for
the CED, which can also be detected on the IP side
CNG tone
detected?
CED tone
detected?
No
No
V.21 tone
detected?
No
Continue call
Yes
Yes
Evaluate
BehaviorOnCed
ToneDetection
Stop sending
voice codec. Start
buffering T.38
packets
Faxmode
Yes
Passthrough
T.38 is
enabled?
Yes
Send a re-INVITE
for T.38
Current voice
codec is data capable?
No
Yes
Continue call with

current codec
No
T.38 accepted
by peer?
Evaluate
BehaviorOnT38Invite
NotAccepted.Behavior
according to error
code
No
List of
negotiated
codecs
contains a
data-capable
codec?
Yes
ReInviteForClear
ChannelOnly (default)
Start sending T.38

packets
At least one data

codec is enabled
on the device ?
Continue T.38 fax
DropCall
Switch to highest
negotiated priority
data codec and
continue call
No
Switch to highest
configured priority G.711
data codec
(PCMU if all disabled )
ReInviteOn
NoNegotiated
DataCodec
No
Yes
Yes
ReEstablish
Audio
Evaluate
interopB ehavior
OnMachine
Detection
ReInviteOn
FaxT38Only
Continue call
UsePrevious
MediaNegiciation
Send SIP reINVITE for datacapable codecs
Terminate call
No
Disable fax/modem detection

and send SIP re-INVITE for
voice -capable codecs
New codec
accepted by
peer ?
Yes
Restore the previouly
used voice codec and
continue call
380
Switch to
selected data
codec and
continue call
H A P T E R
32
Security
This chapter describes how to properly configure the security parameters of the Mediatrix unit.
Standards Supported
RFC 3711: The Secure Real-time Transport Protocol (SRTP)

(Supports only the AES-CM encryption)
RFC 3830: MIKEY: Multimedia Internet KEYing (Compliant

for method Pre-Shared Key only)
RFC 4567: Key Management Extensions for Session

Description Protocol (SDP) and Real Time Streaming
Protocol (RTSP)
RFC 4568: SDES: Security Descriptions for Media Streams
Introduction
You can define security features on the Mediatrix unit. This section applies to media security parameters.
Applying security on the Mediatrix unit involves several steps:
Properly set the time on the Mediatrix unit by configuring a valid SNTP server (SNTP
Configuration on page 95) and time zone (Time Configuration on page 96).
Transfer a valid CA certificate into the Mediatrix unit (Chapter 47 - Certificates Management
on page 569).
Use secure signalling by enabling the TLS transport protocol (Chapter 28 - SIP Transport
Parameters on page 307).
Caution: If you enable Secure RTP (SRTP) on at least one line, it is acceptable to have the secure SIP
transport (TLS) disabled for testing purposes. However, you must never use this configuration in a
production environment, since an attacker could easily break it. Enabling TLS for SIP Transport is strongly
recommended and is usually mandatory for security interoperability with third-party equipments.
Caution: When using a codec other than G.711, enabling Secure RTP (SRTP) has an impact on the
Mediatrix units overall performance as SRTP requires CPU power. The more lines use SRTP, the more
overall performance is affected. This is especially true with the Mediatrix 4116, LP16, 4124 and LP24
models. This could mean that a user picking up a telephone on these models may not have a dial tone due
to lack of resources. See also DSP Limitation on page 433 for more details on resources limitations with
SRTP and conferences.
Use secure media by:
Defining the SRTP/ SRTCP base port (Base Ports Configuration on page 401).
Setting the RTP secure mode to Secure or Secure with fallback (this section).
Security Parameters
The Security section allows you to secure the RTP stream (media) of the Mediatrix unit.
Since the SRTP encryption and authentication needs more processing, the number of calls that the Mediatrix
unit can handle simultaneously may be reduced, depending of the codecs enabled. You could set the
Mediatrix unit not to impact the number of simultaneous calls by enabling only G.711 codecs and disabling
every other voice or data codec, even T.38.
381
Chapter 32 - Security
Security Parameters
The Mediatrix unit supports the MIKEY protocol using pre-shared keys (MIKEY-PS) or the SDES protocol for
negotiating SRTP keys.
To set the RTP stream security parameters:

1.
In the web interface, click the Media link, then the Security sub-link.
Figure 182: Media Security Web Page
4
6
2.
3.
Select whether or not you want to override one or more of the available default security parameters
in the Endpoint Specific drop-down menu.
4.
In the Security section of the Security page, select the RTP payload mode in the Mode drop-down
menu.
The unit relies on these modes when negotiating an audio stream.
Table 356: Default RTP Mode
Mode
Description
Unsecure
The Mediatrix unit supports only unsecure RTP. It rejects secure RTP
offers it receives.
Secure
The Mediatrix unit supports only secure RTP. It rejects unsecure RTP
offers it receives.
Secure with fallback
The Mediatrix unit supports both secure and unsecure RTP. It prioritizes
secure RTP but permits unsecure RTP fallback when the remote peer
does not support security.
The TLS SIP transport must usually be enabled for secure audio negotiation via SDP (refer to the
Caution box above). See Chapter 28 - SIP Transport Parameters on page 307 for more details.
The RTP mode is reflected in the SIP/SDP payload, with a RTP/AVP for unsecure RTP, and a RTP/
SAVP for secure RTP.
The following basic rules apply when sending units capabilities via SDP:
382
When the RTP mode is set to Unsecure, the Mediatrix unit offers/answers with only one
active RTP/AVP audio stream. Any other audio stream present in the offer is disabled
in the answer.
When the RTP mode is set to Secure, the Mediatrix unit offers/answers with only one
active RTP/SAVP audio stream. Any other audio stream present in the offer is disabled
in the answer.
Security Parameters
5.
When the RTP mode is set to Secure with fallback, the Mediatrix unit offers one RTP/
AVP and one RTP/SAVP audio streams. The unit answers with only the most secure
stream.
If the remote unit answers to an offer with both RTP/AVP and RTP/SAVP streams
enabled, a new offer is sent with only RTP/SAVP enabled.
Select the key management protocol for SRTP in the Key Management drop-down menu.
Table 357: Key Management Protocol
Protocol
Description
Mikey
Use MIKEY (Multimedia Internet KEYing).
Sdes
Use SDES (Security DEScriptions).
This parameter has no effect if the Mode parameter is set to Unsecure.

If the unit receives an offer with both MIKEY and SDES, only the configured key management
protocol is kept.
6.
Select the encryption type to be used with SRTP in the Encryption drop-down menu.
Encryption
Description
Null
No encryption. It is ignored for the Sdes Key Management as

defined in Step 3. Use only for debug.
AesCm128
AES (Advanced Encryption Standard) Counter Mode 128 bits.
This parameter has no effect if the Mode parameter is set to Unsecure.

7.
Select whether or not to enable T.38 even if the call has been established previously in SRTP in the
Allow Unsecure T.38 with Secure RTP drop-down menu.
Mode
Description
Disable
T.38 is disabled for SRTP calls.
Enable
T.38 is enabled for SRTP calls.
Caution: Enabling this parameter opens a security hole,

because T.38 is an unsecure protocol.
This menu is available only in the default configuration.
Note that this parameter has no effect if the Mode parameter is set to Unsecure.
8.
Enforcing Symmetric RTP


by using the CLI
For each bi-directional RTP streams, you can define whether or not to enforce that incoming RTP packets are
from the same source as the destination of outgoing RTP packets.
Enforcing symmetric RTP may prevent legitimate RTP streams coming from a media server from being
processed, for example: Music and conferencing servers.
383
Chapter 32 - Security
Security Parameters

Table 360: Enforce Symmetric RTP Parameters
Parameter
Description
disable
Accept packets from all sources. This is the default value.
enable
Silently discard incoming RTP packets with source address and port differing from the
destination address and port of outgoing packets.
To enforce symmetric RTP:

1.
In the miptMIB, set the enforceSymmetricRtpEnable variable with the proper behaviour.
mipt.enforceSymmetricRtpEnable="Value"

Figure 183: Symmetric RTP Values
Value Meaning
384
disable
enable
H A P T E R
33
RTP Statistics Configuration

The Mediatrix unit collects meaningful statistics that can be read via the web interface. This chapter describes
how to read and configure the RTP statistics.
Note that the RTP statistics are also available via SNMP and CLI.
Statistics Displayed
The Mediatrix unit collects two types of statistics:
statistics for the last 10 connections

statistics for the last 10 collection periods
The Connection Statistics section displays the statistics for the last 10 connections. You can use the Display
All button to display more information or the Display Overview button to display less information.
The Connection Period Statistics section displays the statistics for the last 10 periods. The period duration is
defined in the Statistics Configuration section. You can use the Display All button to display more information
or the Display Overview button to display less information.
Figure 184: Telephony RTP Stats Web Page

Table 361: Statistics Displayed
Statistic
Octets Tx
Connection Statistics
Number of octets transmitted during the
connection.
Collection Period Statistics

Number of octets transmitted during the
collection period. This value is obtained by
cumulating the octets transmitted in all
connections that were active during the
collection period.
385
Chapter 33 - RTP Statistics Configuration
Statistics Displayed
Table 361: Statistics Displayed (Continued)

Statistic
386
Octets Rx
Number of octets received during the

connection.
Number of octets received during the

cumulating the octets received in all
collection period.
Packets Tx
Number of packets transmitted during the

connection.
Number of packets transmitted during the

cumulating the packets transmitted in all
collection period.
Packets Rx
Number of packets received during the

connection.
Number of packets received during the

cumulating the packets received in all
collection period.
Packets Lost
Number of packets lost during the

connection. This value is obtained by
substracting the expected number of
packets based on the sequence number
from the number of packets received.
Number of packets lost during the collection

period. This value is obtained by cumulating
the packets lost in all connections that were
active during the collection period.
Min. Jitter
Minimum interarrival time, in ms, during the

connection. All RTP packets belonging to
the connection and received at the RTP
level are considered in the calculation.
Minimum interarrival time, in ms, during the

collection period. This value is the lowest
interarrival jitter for all connections that were
Max. Jitter
Maximum interarrival time, in ms, during the

Maximum interarrival time, in ms, during the

collection period. This value is the highest
interarrival jitter for all connections that were
Avg. Jitter
Average interarrival time, in ms, during the

Average interarrival time, in ms, during the

collection period. This value is the weighted
average of the interarrival jitter for all
collection period. For each connection, the
total jitter of packets received during the
collection period and the total number of
packets received during the collection period
are used in the weighted average
calculation.
Min. Latency
Minimum latency, in ms, during the

connection. The latency value is computed
as one half of the round-trip time, as
measured through RTCP.
Minimum latency, in ms, during the

collection period. This value is the lowest
latency for all connections that were active
during the collection period.
Max. Latency Maximum latency, in ms, during the

Maximum latency, in ms, during the

collection period. This value is the highest
latency for all connections that were active
Statistics Configuration
Table 361: Statistics Displayed (Continued)

Statistic
Avg. Latency
Average latency, in ms, during the

Average latency, in ms, during the collection

period. This value is the weighted average
of the latency for all connections that were
active during the collection period. For each
connection, the total latency of packets
received during the collection period and the
total number of packets received during the
collection period are used in the weighted
average calculation.
Statistics Configuration
You can define how to collect the statistics. The statistics are sent as syslog messages, so you must properly
set the syslog information before setting the statistics. You must set the Media IP Transport (MIPT) service to
the Info or Debug level. See Syslog Daemon Configuration on page 73 for more details on how to configure
the Syslog.
To configure how to collect statistics:

1.
In the web interface, click the Telephony link, then the RTP Stats sub-link.
Figure 185: Telephony RTP stats Web Page
2.
Set the Collection Period field with the collection period duration in minutes.
Putting a value of 0 disables the collection period statistics feature.
3.
Set the End-of-Connection Notification drop-down menu with the proper behaviour.
Table 362: End-of-Connection Notification
Parameter
Description
Enable
Notifications are generated.
Disable
Notifications are not generated.

387
4.
Channel Statistics
Set the End-of-Period Notification drop-down menu with the proper behaviour.
Table 363: End-of-Period Notification
Parameter
5.
Description
Enable
Notifications are generated.
Disable
Notifications are not generated.
To save your settings, click Submit.
To save your settings and reset the statistics of the current period., click Submit &
Reset Current Collection Period Statistics.
The previous periods are left unchanged.
Channel Statistics
This section describes how to access data available only in the MIB parameters of the Mediatrix unit. You can
display these parameters as follows:

by using the CLI
The channel statistics are cumulated RTP statistics for all calls using a specific channel of a telephony
interface. Statistics are updated at the end of each call.
The statistics are associated to the channel in use at the end of the call. In some cases, such as in hold/resume
scenarios, the channel assignment may change during a call. This can result in discrepancies between the
RTP statistics and the actual usage of the telephony interface.
The following are the channel statistics the Mediatrix unit keeps.
Table 364: Channel Statistics
MIB Variable
388
Statistics Description
PacketsSent
Number of packets transmitted on the channel since

service start. This value is obtained by cumulating the
packets transmitted in all the connections that ended
PacketsReceived
Number of packets received on the channel since

packets received in all the connections that ended
BytesSent
Number of bytes transmitted on the channel since

bytes transmitted in all the connections that ended
BytesReceived
Number of bytes received on the channel since service

start. This value is obtained by cumulating the bytes
received in all the connections that ended during the
collection period.
AverageReceiveInterarr
ivalJitter
Average interarrival time, in microseconds, for the

channel since service start. This value is based on the
average interarrival jitter of each call ended during the
collection period. The value is weighted by the duration
of the calls.
Channel Statistics
To display channel statistics:

1.
In the miptMIB, go to the ChannelStatistics table.

You can also use the following line in the CLI:
get mipt.channelStatistics
To reset channel statistics values to zero:

1.
In the miptMIB, set ChannelStatistics.Reset to Reset for the endpoint to reset.

set mipt.channelStatistics.Reset=Reset
2.
In the miptMIB, set ChannelStatistics[EpChannelId=channelStatisticsEpChannelId].Reset

to Reset to reset only one specific endpoint.
where:
channelStatisticsEpChannelId is the string that identifies the combination of an

endpoint and a channel. The endpoint name is the same as the EpId used to refer to
endpoints in other tables. On endpoints with multiple channels, the channel number
must be appended at the end of the endpoint name, separated with a dash.
set mipt.channelStatistics[EpChannelId=channelStatisticsEpChannelId].Reset=Reset
Examples:
Slot3/E1T1-12 refers to endpoint Slot3/E1T1, channel 12.
Phone-Fax1 refers to FXS endpoint Phone-Fax1 on a 4102s.
Port06 refers to FXS endpoint Port06 on 4108/4116/4124.
No channel number is appended to FXS endpoint strings because FXS lines do not support multiple
channels.
389
390
Channel Statistics
H A P T E R
34
Miscellaneous Media Parameters

This chapter describes how to configure parameters that apply to all codecs.
Standards Supported
draft-choudhuri-sip-info-digit-00.txt
Jitter Buffer Configuration

DTMF Transport Configuration
Machine Detection Configuration
Base Ports Configuration

The Jitter Buffer section allows you to configure parameters to reduce jitter buffer issues.
To set the jitter buffer parameters:

1.
In the web interface, click the Media link, then the Misc sub-link.
Figure 186: Media Misc Web Page
2
3
4
5
6
7
8
9
10
2.
3.
In the Jitter Buffer section, if you have selected a specific endpoint, select whether or not you want
to override the jitter buffer parameters set in the Default configuration in the Endpoint Specific dropdown menu.
4.
Select the jitter buffer level in the Level drop-down menu.
391
Jitter is an abrupt and unwanted variation of one or more signal characteristics, such as the interval
between successive pulses or the frequency or phase of successive cycles. An adaptive jitter buffer
usually consists of an elastic buffer in which the signal is temporarily stored and then retransmitted
at a rate based on the average rate of the incoming signal.
Table 365: Jitter Buffer Levels
Level
Description
Optimize Latency The jitter buffer is set to the lowest effective value to minimize the latency.
Voice cut can be heard if the network is not optimal. The predefined values
are as follows:
Normal
Fax / Modem
5.
Minimum value: 10 ms
Maximum value: 40 ms
The jitter buffer tries to find a good compromise between the latency and the
voice quality. This setting is recommended in private networks. The
predefined values are as follows:
Optimize Quality
Custom
The jitter buffer is set to a high value to minimize the voice cuts at the cost of
high latency. This setting is recommended in public networks. The
predefined values are as follows:
The jitter buffer is set to maximum. The Fax/Modem transmission is very

sensitive to voice cuts but not to latency, so the fax has a better chance of
success with a high buffer. The predefined values are as follows:
The jitter buffer uses the configuration of the Minimum and Maximum
variables (Steps 4 and 5).
If you have selected the Custom level, define the target jitter buffer length in the Minimum field of
the Voice Call part.
The adaptive jitter buffer attempts to hold packets to the minimal holding time. This is the minimal
delay the jitter buffer adds to the system. The minimal jitter buffer is in ms and must be equal to or
smaller than the maximal jitter buffer.
Values range from 0 ms to 135 ms. The default value is 30 ms. You can change values by
increments of 1 ms, but Media5 recommends to use multiples of 5 ms. The minimal jitter buffer
should be a multiple of ptime.
It is best not to set the minimal jitter value below the default value. Setting a minimal jitter buffer
below 5 ms could cause an error. Jitter buffer adaptation behaviour varies from one codec to
another. See About Changing Jitter Buffer Values on page 393 for more details.
6.
If you have selected the Custom level, define the maximum jitter buffer length in the Maximum field
of the Voice Call part.
This is the highest delay the jitter buffer is allowed to introduce. The jitter buffer length is in ms and
must be equal to or greater than the minimum jitter buffer.
Values range from 0 ms to 135 ms. The default value is 125 ms. You can change values by
increments of 1 ms, but Media5 recommends to use multiples of 5 ms. The maximal jitter buffer
should be a multiple of ptime.
The maximum jitter buffer value should be equal to the minimum jitter buffer value + 4 times the
ptime value. Lets say for instance that:
392
Minimum jitter buffer value is 30 ms
Ptime value is 20 ms
The maximum jitter buffer value should be: 30 + 4x20 = 110 ms
7.
If you have selected the Custom level, define the voiceband data custom jitter buffer type in the
Playout Type drop-down menu of the Data Call part.
This is the algorithm to use for managing the jitter buffer during a call. The Nominal field value
serves as the delay at the beginning of the call and might be adapted afterwards based on the
selected algorithm.
Table 366: Voiceband Data Custom Jitter Buffer Type
Level
8.
Description
Adaptive
Immediately
The nominal delay varies based on the estimated packet jitter. Playout
adjustment is done immediately when the actual delay goes out of bounds of
a small window around the moving nominal delay.
Adaptive Silence
The nominal delay varies based on the estimated packet jitter. Playout
adjustment is done based on the actual delay going out of bounds of a small
window around the moving nominal delay. The adjustment is deferred until
silence is detected (either from playout buffer underflow or by analysis of
packet content). Playout adjustment is also done when overflow or
underflow events occur.
Fixed
The nominal delay is fixed to the value of the Nominal field value and does
not change thereafter. Playout adjustment is done when overflow or
underflow events occur.
If you have selected the Custom level, define the voiceband data jitter buffer minimal length (in
milliseconds) in the Minimum field of the Data Call part.
The voiceband data jitter buffer minimal length is the delay the jitter buffer tries to maintain. The
minimal jitter buffer MUST be equal to or smaller than the voiceband data maximal jitter buffer.
The minimal jitter buffer should be a multiple of ptime.
This value is not available when the Playout Type drop-down menu is set to Fixed.
9.
If you have selected the Custom level, define the voiceband data custom jitter buffer nominal length
in the Nominal field of the Data Call part.
The jitter buffer nominal length (in milliseconds) is the delay the jitter buffer uses when a call begins.
The delay then varies depending on the type of jitter buffer.
In adaptive mode, the nominal jitter buffer should be equal to (voice band data minimal jitter buffer
+ voice band data maximal jitter buffer) / 2.
10.
If you have selected the Custom level, define the default voiceband data custom jitter buffer
maximal length in the Maximum field of the Data Call part.
The jitter buffer maximal length (in milliseconds) is the highest delay the jitter buffer is allowed to
introduce. The maximal jitter buffer MUST be equal to or greater than the minimal jitter buffer.
The maximal jitter buffer should be a multiple of ptime.
The maximal jitter buffer should be equal to or greater than voiceband data minimal jitter buffer + (4
* ptime) in adaptive mode.
See About Changing Jitter Buffer Values on page 393 for more details.
11.
About Changing Jitter Buffer Values

Media5 recommends to avoid changing the target and maximum jitter buffer values unless experiencing or
strongly expecting one of the following symptoms:
If the voice is scattered, try to increase the maximum jitter buffer value.
If the delay in the voice path (end to end) is too long, you can lower the target jitter value, but
393
ONLY if the end-to-end delay measured matches the target jitter value.
For instance, if the target jitter value is 50 ms, the maximum jitter is 300 ms and the delay measured
is 260 ms, it would serve nothing to reduce the target jitter. However, if the target jitter value is
100 ms and the measured delay is between 100 ms and 110 ms, then you can lower the target jitter
from 100 ms to 30 ms.
Starting a Call in Voiceband Data Mode


by using the CLI
You can define whether or not a call should be started in voiceband data mode.
Table 367: Voiceband Data Mode Parameters
Parameter
Description
Disable
The call is started in voice mode. A fax/modem tone detection triggers a transition from voice
to voiceband data according to the configuration in the Machine Detection Group
(Miscellaneous Media Parameters on page 391).
Enable
The call is started in voiceband data mode.
To start a call in voiceband data mode:

1.
In the telIfMIB, set the voiceband data mode in the InteropStartCallInVbdEnable variable.
telIf.InteropStartCallInVbdEnable="Value"

Table 368: Voiceband Data Mode Values
Value
Method
Disable
Enable

The DTMF Transport section allows you to set the DTMF transport parameters of the Mediatrix unit.
To set DTMF transport parameters:

1.
2.
In the DTMF Transport section of the Misc page, select whether or not you want to override the
DTMF transport parameters set in the Default configuration in the Endpoint Specific drop-down
menu.
394
Figure 187: DTMF Transport Section

2
4
3.
Select the DTMF transport type in the Transport Method drop-down menu.
The following choices are available:
Table 369: DTMF Transport Type Parameters
Transport Parameter
4.
Description
In-band
The DTMFs are transmitted like the voice in the RTP

stream.
Out-of-band using RTP
The DTMFs are transmitted as per RFC 2833. This

parameter also works with SRTP.
Out-of-band using SIP
The DTMFs are transmitted as per draft-choudhuri-sip-infodigit-00.
Signaling protocol Dependant
The signalling protocol has the control to select the DTMF

transport mode. The SDP body includes both RFC 2833
and draft-choudhuri-sip-info-digit-00 in that order of
preference.
If you have selected the Out-of-band using SIP transport method, select the method used to
transport DTMFs out-of-band over the SIP protocol in the SIP Transport Method drop-down menu.
Table 370: DTMF Out-of-Band Transport Methods
Method
Description
draftChoudhuriSipInfoDigit00 Transmits DTMFs by using the method defined in draftchoudhuri-sip-info-digit-00. Only the unsolicited-digit part is
supported.
DTMF out-of-band
Certain compression codecs such as G.723.1 and G.729 effectively distort voice because they lose
information from the incoming voice stream during the compression and decompression phases. For
normal speech this is insignificant and becomes unimportant. In the case of pure tones (such as DTMF)
this distortion means the receiver may no longer recognize the tones. The solution is to send this
information as a separate packet to the other endpoint, which then plays the DTMF sequence back by regenerating the true tones. Such a mechanism is known as out-of-band DTMF. The Mediatrix unit receives
and sends out-of-band DTMFs as per ITU Q.24. DTMFs supported are 0-9, A-D, *, #.
395
Table 370: DTMF Out-of-Band Transport Methods (Continued)

Method
Info DTMF Relay
Description
Transmits DTMFs by using a custom method. This custom
method requires no SDP negotiation and assumes that the other
peer uses the same method.
It uses a SIP INFO message with a content of type application/
dtmf-relay. The body of the message contains the DTMF
transmitted and the duration of the DTMF:
Signal= 1
Duration= 160
When transmitting, the duration is the one set in the

interopDtmfTransportDuration variable (see DTMF
Transport over the SIP Protocol on page 396).
When receiving, the duration of the DTMF received is not used
and the DTMF is played for 100 ms.
DTMFs are transmitted one at a time.
Available digits are 0123456789ABCD*#. The Mediatrix unit
also supports the ,;p characters when receiving DTMFs.
5.
If you have selected the Out-of-band using RTP transport method, set the payload type in the
Payload Type field.
You can determine the actual RTP dynamic payload type used for the telephone-event in an initial
offer. The payload types available are as per RFC 1890. Available values range from 96 to 127.
6.
DTMF Transport over the SIP Protocol


by using the CLI
You can set the DTMF duration sent in the INFO message when using the Info DTMF Relay method to
transmit DTMFs (see Miscellaneous Media Parameters on page 391, Step 8 for more details).
To set the DTMF duration sent in the INFO message:

1.
In the sipEpMIB, set the DTMF duration sent in the INFO message when using the infoDtmfRelay
method to transmit DTMFs in the interopDtmfTransportDuration variable.
sipEp.interopDtmfTransportDuration="Value"
This value is expressed in milliseconds (ms). The default value is 100 ms.
DTMF Detection

by using the CLI
The default DTMF detection parameters of the Mediatrix unit may sometimes not be enough to properly detect
the DTMFs. This section describes how to set additional DTMF detection parameters.
396
DTMF Frequencies
The DTMF keypad is laid out in a 4x4 matrix, with each row representing a low frequency, and each column
representing a high frequency. For example, pressing a single key (such as '1') sends a sinusoidal tone of the
two frequencies (697 Hz and 1209 Hz). When the unit is configured to send DTMFs out-of-band, its DSP
detects these DTMFs, removes them from the RTP stream, and sends them out-of-band.
Table 371: DTMF Keypad Frequencies
Low/High (Hz)
1209
1336
1477
1633
697
770
852
941
DTMF Detection Configuration

Below is a frequency spectrum analysis of a DTMF (9) with the Frequency in Hertz on the x axis and the Power
in dBm on the y axis. The low and high frequencies of the DTMF are in red and you can clearly see that they
are the most powerful frequencies in the signal.
Figure 188: DTMF Detection Example
To configure the DTMF detection:

1.
In the telIfMIB, define how the Rise Time criteria should be configured for DTMF detection in the
interopDtmfDetectionRiseTimeCriteria variable.
sipEp.interopDtmfDetectionRiseTimeCriteria="Value"
397

Table 372: DTMF Detection Values
Value
100
Method
CheckSr
Enables the Step Rise criteria and disables the Confirm DTMF SNR
criteria.
The Step Rise criteria compares the current frame energy to the high
frequency power of the previous frame. If the current frame energy is high
enough, then it passes the test, further validating the DTMF.
Disabling the Step Rise criteria may result in deteriorated talk-off
performance, but increases the detection of malformed DTMF.
200
ConfirmSnr Enable the Confirm DTMF SNR criteria and disable the Step Rise criteria.
The Confirm DTMF SNR criteria is an additional Signal-to-noise ratio test
performed before a confirmed DTMF report is sent to finally validate the
DTMF.
2.
Set the interopDtmfDetectionPositiveTwist variable.

sipEp.interopDtmfDetectionPositiveTwist="Value"
When the high-group frequency of a DTMF is more powerful than the low-group frequency, the
difference between the high-group frequency absolute power and the low-group frequency absolute
power must be smaller than or equal to the value set in this variable. Otherwise, the DTMF is not
detected.
Raising this value increases the sensitivity of DTMF detection. Raising this value too high may also
cause false detections of DTMFs.
Using the Payload Type Found in the Answer


by using the CLI
The default behaviour when sending an initial offer that contains an RFC 2833 payload type is to keep using
that payload type even if the response comes back with a different one. You can set the Mediatrix unit to rather
use the payload type found in the answer.
This feature is effective only if the Transport Method drop-down menu is set to Out-of-band using RTP (see
Miscellaneous Media Parameters on page 391 for more details).
Table 373: Payload Type in Answer
Parameter
Description
disable
Keep using the initial payload type. This is the default value.
enable
Use the RFC 2833 payload type found in the received answer.
To use the payload type found in the answer:

1.
In the sipEpMIB, set the interopUseDtmfPayloadTypeFoundInAnswer variable with the proper

behaviour.
sipEp.interopUseDtmfPayloadTypeFoundInAnswer="Value"
398

Figure 189: Payload Type Values
Value Meaning
0
disable
enable
Quantity of initial packets sent to transmit a DTMF Out-of-Band using RTP


by using the CLI
You can specify the quantity of packets sent at the beginning of an Out-of-Band DTMF using RTP. This
variable also specifies the quantity of terminating packets that are sent at the end of the DTMF transmission.
Note that this variable has an effect only if the Transport Method drop-down menu is set to Out-of-band
using RTP (see Miscellaneous Media Parameters on page 391 for more details).
To set the initial quantity of RTP packets:

1.
In the miptMIB, set the InteropDtmfRtpInitialPacketQty variable with the proper quantity.
mipt.interopDtmfRtpInitialPacketQty="Value"
where Value may be between 1 and 3.

The Machine Detection section allows you to set the tone detection parameters of the Mediatrix unit.
To set Machine detection parameters:

1.
2.
In the Machine Detection section of the Misc page, select whether or not you want to override the
machine detection parameters set in the Default configuration in the Endpoint Specific drop-down
menu.
Figure 190: Machine Detection Section
2
4
3
5
399
3.
Select whether or not you want to enable fax calling tone (CNG tone) detection in the CNG Tone
Detection drop-down menu.
Table 374: CNG Tone Detection Settings
Setting
Enable
Description
Upon recognition of the CNG tone, the unit switches the communication from
voice mode to fax mode and the CNG is transferred by using the preferred fax
codec.
Note: This option allows for quicker fax detection, but it also increases the risk
of false detection.
Disable
The CNG tone does not trigger a transition from voice to data and the CNG is
transferred in the voice channel.
Note: With this option, faxes are detected later, but the risk of false detection
is reduced.
4.
Select whether or not you want to enable CED tone detection in the CED Tone Detection drop-down
menu.
Table 375: CNG CED Detection Settings
Setting
5.
Description
Enable
Upon recognition of the CED tone, the unit behaves as defined in the
Behavior on CED Tone Detection parameter Step 6).
Disable
The CED tone does not trigger a transition to fax or voiceband data mode.
The CED is transferred in the voice channel.
Select whether or not you want to enable fax V.21 modulation detection in the V.21 Modulation
Table 376: V.21 Modulation Detection Settings
Setting
6.
Description
Enable
Upon recognition of the V.21 modulation tone, the unit switches the
communication from voice mode to fax mode and the signal is transferred by
using the preferred fax codec.
Disable
The V.21 modulation does not trigger a transition from voice to data and the
signal is transferred in the voice channel.
Define the behaviour of the unit upon detection of a CED tone in the Behavior on CED Tone
Table 377: CED Tone Detection Settings
Setting
Description
Passthrough
The CED tone triggers a transition from voice to voice band data and is
transferred in the voice channel.. Use this setting when any kind of analog
device (i.e.: telephone, fax or modem) can be connected to this port.
Fax Mode
Upon detection of a CED tone, the unit switches the communication from
voice mode to fax mode and the CED is transferred by using the preferred fax
codec. Only a fax can then be connected to this port.
Note: This parameter has no effect if the CED Tone Detection parameter is set to Disabled.
7.
400


The Base Ports section allows you to set the ports that the Mediatrix unit uses for different transports.
This section is available only in the default endpoint configuration.
To set base ports parameters:

1.
2.
In the Base Ports section of the Misc page, set the UDP port number you want to use as RTP/RTCP
base port in the RTP field.
The RTP/RTCP ports are allocated starting from this base port.
RTP ports number are even and RTCP ports number are odd.
The default RTP/RTCP base port is 5004. For instance, assuming that the base port is defined on
5004, if there is currently no ongoing call and there is an incoming or outgoing call, the unit uses the
RTP/RTCP ports 5004 and 5005.
Figure 191: Base Ports Section
2
4
3.
Set the UDP port number you want to use as SRTP/SRTCP base port in the SRTP field.
The SRTP/SRTCP ports are allocated starting from this base port.
SRTP ports number are even and SRTCP ports number are odd.
The default SRTP/SRTCP base port is 5004. For instance, assuming that the base port is defined
on 5004, if there is currently no ongoing call and there is an incoming or outgoing call, the unit uses
the SRTP/SRTCP ports 5004 and 5005.
Using the same base port for RTP/RTCP and SRTP/SRTCP does not conflict.
Note that if the media transport is set to Secure with fallback (Chapter 32 - Security on page 381),
both RTP and SRTP base ports are used at the same time when initiating an outgoing call. If there
is currently no call and the default base ports are used, the RTP port is 5004 and the SRTP port is
the next available port starting from the base port, which is 5006.
4.
Set the port number you want to use as T.38 base port in the T.38 field.
The T.38 ports are allocated starting from this base port.
The default T.38 base port is 6004. For instance, assuming that the base port is defined on 6004 if
there is currently no ongoing call and there is an incoming or outgoing call, the unit uses the T.38
port 6005.
5.
401
402
H A P T E R
35
DTMF Maps Configuration

This chapter describes how to configure and use the DTMF maps of the Mediatrix unit.
Standards Supported
RFC 2705: Media Gateway Control Protocol (MGCP) Version

1.0, section 3.4 (Formal syntax description of the protocol).
DTMF maps syntax.

General DTMF maps parameters.
Allowed DTMF maps parameters.
Refused DTMF maps parameters.
Introduction
A DTMF map (also called digit map or dial map) allows you to compare the number users just dialed to a string
of arguments. If they match, users can make the call. If not, users cannot make the call and get an error signal.
It is thus essential to define very precisely a DTMF map before actually implementing it, or your users may
encounter calling problems.
Because the Mediatrix unit cannot predict how many digits it needs to accumulate before transmission, you
could use the DTMF map, for instance, to determine exactly when there are enough digits entered from the
user to place a call.
Syntax
The permitted DTMF map syntax is taken from the core MGCP specification, RFC 2705, section 3.4:
DigitMap = DigitString / '(' DigitStringList ')'
DigitStringList = DigitString 0*( '|' DigitString )
DigitString = 1*(DigitStringElement)
DigitStringElement = DigitPosition ['.']
DigitPosition = DigitMapLetter / DigitMapRange
DigitMapLetter = DIGIT / '#' / '*' / 'A' / 'B' / 'C' / 'D' / 'T'
DigitMapRange = 'x' / '[' 1*DigitLetter ']'
DigitLetter ::= *((DIGIT '-' DIGIT ) / DigitMapLetter)
Where x means any digit and . means any number of.

For instance, using the telephone on your desk, you can dial the following numbers:
Table 378: Number Examples
Number
Description
Local operator
00
Long distance operator
xxxx
Local extension number
8xxxxxxx
Local number
#xxxxxxx
Shortcut to local number at other corporate sites
91xxxxxxxxxx
Long distance numbers
405
Chapter 35 - DTMF Maps Configuration
Syntax
Table 378: Number Examples (Continued)

Number
Description
9011 + up to 15 digits
International number
The solution to this problem is to load the Mediatrix unit with a DTMF map that corresponds to the dial plan.
A Mediatrix unit that detects digits or timers applies the current dial string to the DTMF map, attempting a
match to each regular expression in the DTMF map in lexical order.
If the result is under-qualified (partially matches at least one entry in the DTMF map), waits for
more digits.
If the result matches, dials the number.

If the result is over-qualified (i.e., no further digits could possibly produce a match), sends a fast
busy signal.
Special Characters
DTMF maps use specific characters and digits in a particular syntax.
Table 379: DTMF Map Characters
Character
Use
Digits (0, 1, 2... 9) Indicates specific digits in a telephone number expression.

T
The Timer indicates that if users have not dialed a digit for the time defined, it is likely
that they have finished dialing and the SIP Server can make the call.
Matches any digit, excluding # and *.
Indicates a choice of matching expressions (OR).
Matches an arbitrary number of occurrences of the preceding digit, including 0.
Indicates the start of a range of characters.
Indicates the end of a range of characters.
How to Use a DTMF Map

Lets say you are in an office and you want to call a co-workers 3-digits extension. You could build a DTMF
map that says after the user has entered 3 digits, make the call. The DTMF map could look as follows:
xxx
You could refine this DTMF map by including a range of digits. For instance, you know that all extensions in
your company either begin with 2, 3, or 4. The corresponding DTMF map could look as follows:
[2-4]xx
If the number you dial begins with anything other than 2, 3, or 4, the call is not placed and you get a busy signal.
Combining Several Expressions

You can combine two or more expressions in the same DTMF map by using the | operator, which is equal to
OR.
Lets say you want to specify a choice: the DTMF map is to check if the number is internal (extension), or
external (a local call). Assuming that you must first dial 9 to make an external call, you could define a DTMF
map as follows:
([2-4]xx|9[2-9]xxxxxx)
The DTMF map checks if:
the number begins with 2, 3, or 4 and

the number has 3 digits
If not, it checks if:

406
Syntax
the number begins with 9 and

the second digit is any digit between 2 and 9 and
the number has 7 digits
Note: Enclose the DTMF map in parenthesis when using the | option.
Using the # and * Characters

It may sometimes be required that users dial the # or * to make calls. This can be easily incorporated in a
DTMF map:
xxxxxxx#
xxxxxxx*
The # or * character could indicate users must dial the # or * character at the end of their number to
indicate it is complete. You can specify to remove the # or * found at the end of a dialed number. See
General DTMF Maps Parameters on page 408.
Using the Timer

The Timer indicates that if users have not dialed a digit for the time defined, it is likely that they have finished
dialing and the Mediatrix unit can make the call. A DTMF map for this could be:
[2-9]xxxxxxT
Note: When making the actual call and dialing the number, the Mediatrix unit automatically removes the T
found at the end of a dialed number, if there is one (after a match). This character is for indication purposes
only.
See General DTMF Maps Parameters on page 408 for more details.
Calls Outside the Country

If your users are making calls outside their country, it may sometimes be hard to determine exactly the number
of digits they must enter. You could devise a DTMF map that takes this problem into account:
001x.T
In this example, the DTMF map looks for a number that begins with 001, and then any number of digits after
that (x.).
Example
Table 378 on page 405 outlined various call types one could make. All these possibilities could be covered in
one DTMF map:
(0T|00T|[1-7]xxx|8xxxxxxx|#xxxxxxx|91xxxxxxxxxx|9011x.T)
Validating a DTMF Map

The Mediatrix unit validates the DTMF map as you are entering it and it forbids any invalid value.
407
General DTMF Maps Parameters
General DTMF Maps Parameters

The following are the general DTMF maps parameters you can set.
To set the general DTMF map parameters:

1.
In the web interface, click the Telephony link, then the DTMF Maps sub-link.
Figure 192: Telephony DTMF Maps Web Page
2.
In the General Configuration section, define the time, in milliseconds (ms), between the start of the
dial tone and the receiver off-hook tone, if no DTMF is detected, in the First DTMF Timeout field.
Values range from 1000 ms to 180000 ms. The default value is 20000 ms.
If you want to set a different First DTMF Timeout value for one or more endpoints, click the Edit
Endpoints button (see Configuring Timeouts per Endpoint on page 409 for more details).
3.
Define the value, in milliseconds (ms), of the T digit in the Inter Digit Timeout field.
The T digit expresses a time lapse between the detection of two DTMFs. Values range from 500
ms to 10000 ms. The default value is 4000 ms.
If you want to set a different Inter Digit Timeout value for one or more endpoints, click the Edit
4.
Define the total time, in milliseconds (ms), the user has to dial the DTMF sequence in the
Completion Timeout field.
The timer starts when the dial tone is played. When the timer expires, the receiver off-hook tone is
played. Values range from 1000 ms to 180000 ms. The default value is 60000 ms.
If you want to set a different Completion Timeout value for one or more endpoints, click the Edit
5.
In the DTMF Maps Digit Detection (FXO/FXS) drop-down menu, define when a digit is processed
through the DTMF maps.
This parameters is available only when the unit has FXS or FXO ports.
Table 380: DTMF Maps Digit Detection Parameters
Parameter
6.
408
Description
When
Pressed
Digits are processed as soon as they are pressed. This can lead to a digit leak in
the RTP at the beginning of a call if the voice stream is established before the last
digit is released.
When
Released
Digits are processed only when released. This option increases the delay needed
to match a dialed string to a DTMF map. There is also an impact on the First DTMF
Timeout, Inter Digit Timeout and Completion Timeout parameters since the timers
are stopped at the end of a digit instead of the beginning.
Allowed DTMF Maps
Configuring Timeouts per Endpoint

You can set a different timeout value for one or more endpoints.
To set a different value per endpoint:

1.
In the General Configuration section of the DTMF Maps page, click the Edit Endpoints button.
The following window is displayed:
Figure 193: DTMF Map Timeout Section
2.
Set the Override drop-down menu for the endpoint you want to set to Enable.
3.
Change the value of one or more timeouts as required.
4.
Repeat for each endpoint that you want to modify.
5.
Click Submit when finished.
Allowed DTMF Maps

You can create/edit ten DTMF maps for the Mediatrix unit. DTMF map rules are checked sequentially. If a
telephone number potentially matches two of the rules, the first rule encountered is applied.
To set up DTMF maps:

1.
In the DTMF Map drop-down menu at the top of the window, select Allowed.
The Allowed DTMF Map section displays.
2.
In the Allowed DTMF Map section Enable column, enable one or more DTMF maps by selecting
the corresponding Enable choice.
Figure 194: Allowed DTMF Map Section
2
409
3.
Allowed DTMF Maps
Select the entity to which apply the allowed DTMF map in the Apply to column.
Table 381: DTMF Map Entity
Parameter
4.
Description
Unit
The DTMF map entry applies to the unit.
Endpoint
The DTMF map applies to a specific endpoint. The endpoint is specified in the
Endpoint column of the same row.
Enter a string that identifies an endpoint in other tables in the Endpoint column.
This field is available only if you have selected the Endpoint entity in the previous step for the
specific row.
You can specify more than one endpoint. In that case, the endpoints are separated with a comma
(,). You can use the Suggestions columns drop-down menu to select between suggested values, if
any.
5.
Define the DTMF map string that is considered valid when dialed in the DTMF Map column.
The string must use the syntax described in DTMF Maps Configuration on page 405. A DTMF map
string may have a maximum of 64 characters.
6.
Enter the DTMF transformation to apply to the signalled DTMFs before using it as call destination
in the Transformation column.
The following are the rules you must follow; x represents the signalled number.
Add before x the DTMF to prefix or/and after x the suffix to add. Characters
0123456789*# ABCD are allowed.
Use a sequence of DTMFs between {} to remove a prefix/suffix from the dialed

number if present. Use before x to remove a prefix and after x to remove a suffix.
Characters 0123456789*#ABCD are allowed.
Use a number between () to remove a number of DTMFs. Use before x to remove

DTMFs at the beginning of the number and after x to remove DTMFs at the end.
Characters 0123456789 are allowed.
The transformations are applied in order from left to right.

The following table gives an example with 18195551111# as signalled number.
Table 382: DTMF Map Transformation Examples
Action
7.
Transformation
Result
Add the prefix 0 to the dialed number
0x
018195551111#
Remove the suffix # from the dialed

number
x{#}
18195551111
Remove the first four DTMFs from the dialed

number
(4)x
5551111#
Remove the international code and

termination and replace the area code by
another one
(1){819}514x{#}
5145551111
Replace the signalled DTMFs by 3332222
3332222
3332222
Define the target to use when the DTMF map matches in the Target column.
This allows associating a target (FQDN) with a DTMF map. This defines a destination address to
use when the DTMF map matches. This address is used as destination for the INVITEs in place of
the home domain proxy. This is useful for such features as the speed dial and emergency call.
The default target is used when the value is empty.
The dialed DTMFs are not used if the target contains a user name.
410
Refused DTMF Maps
8.
Enable/Disable the emergency process of the call in the Emergency column.
Disable: The call is processed normally.
Enable: The call is processed as emergency.
The Emergency Call service (also called urgent gateway) allows a 911-style service. It allows a
user to dial a special DTMF map resulting in a message being sent to a specified urgent gateway,
bypassing any other intermediaries.
If enabled, whenever the user dials the specified DTMF map, a message is sent to the target
address.
9.
Refused DTMF Maps

A refused DTMF map forbids to call specific numbers; for instance, you want to accept all 1-8xx numbers
except 1-801. You can create/edit ten refused DTMF maps for the Mediatrix unit.
A refused DTMF map applies before an allowed DTMF map.
To set up refused DTMF maps:

1.
In the DTMF Map drop-down menu at the top of the window, select Refused.
The Refused DTMF Map section displays.
2.
In the Refused DTMF Map section Enable column, enable one or more DTMF maps by selecting
the corresponding Enable choice.
Figure 195: Refused DTMF Map Section
2
3.
Select the entity to which apply the refused DTMF map in the Apply to column.
Table 383: DTMF Map Entity
Parameter
4.
Description
Unit
The DTMF map entry applies to the unit.
Endpoint
The DTMF map applies to a specific endpoint. The endpoint is specified in the
Endpoint column of the same row.
Enter a string that identifies an endpoint in other tables in the Endpoint column.
This field is available only if you have selected the Endpoint entity in the previous step for the
specific row.
411
Refused DTMF Maps
You can specify more than one endpoint. In that case, the endpoints are separated with a comma
(,). You can use the Suggestions columns drop-down menu to select between suggested values, if
any.
5.
Define the DTMF map string that is considered valid when dialed in the DTMF Map column.
The string must use the syntax described in DTMF Maps Configuration on page 405. A DTMF map
string may have a maximum of 64 characters.
6.
412
H A P T E R
36
Call Forward Configuration

This chapter describes how to set three types of Call Forward:
On Busy
On No Answer
Unconditional

Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
Call Forward On Busy

You can automatically forward the incoming calls of your users to a pre-determined target if they are already
on the line. The user does not have any feedback that a call was forwarded.
You can enable the Call Forward On Busy feature in two ways:
By allowing the user to configure the call forward activation and its destination via the handset
(Steps 4-6).
By manually enabling the service (Steps 7-8).
413
Chapter 36 - Call Forward Configuration
To set the Call Forward On Busy feature:

1.
In the web interface, click the Telephony link, then the Call Forward sub-link.
Figure 196: Telephony Call Forward Web Page
2
3
4
5
7
6
8
2.
Select to which endpoint you want to apply the changes in the Select Endpoint drop-down menu at
You have the choice between Default and all FXS endpoints your Mediatrix unit has.
3.
In the Call Forward On Busy section, define whether or not you want to override the Call Forward
On Busy parameters set in the Default configuration in the Endpoint Specific drop-down menu.
4.
Enable the Call forward configuration via handset service by setting the Allow Activation via
Handset drop-down menu to Enable.
You also need to configure the activation and deactivation DTMF maps (steps 5 and 6).
If you select Disable, this does not disable the call forward, but prevents the user from activating or
deactivating the call forward service. The user will not be able to use the digits used to activate and
deactivate the call forward service.
5.
Define the digits that users must dial to start the service in the DTMF Map Activation field.
This field is available only in the Default configuration.
For instance, you could decide to put *72 as the sequence to activate the service. This sequence
must be unique and follow the syntax for DTMF maps (see Chapter 35 - DTMF Maps Configuration
on page 405). Dialing this DTMF map does not have any effect unless the services status is
enabled.
The activating sequence is set for all the endpoints of the Mediatrix unit. You cannot have a different
sequence for each endpoint.
6.
Define the digits that users must dial to stop the service in the DTMF Map Deactivation field.
For instance, you could decide to put *73 as the sequence to deactivate the service. This
sequence must be unique and follow the syntax for DTMF maps (see Chapter 35 - DTMF Maps
Configuration on page 405). Dialing this DTMF map does not have any effect unless the services
status is enabled.
The deactivating sequence is set for all the endpoints of the Mediatrix unit. You cannot have a
different sequence for each endpoint.
7.
Set the call forward service in the Activation field to Inactive or Active.
Table 384: Activation State
State
Inactive
414
Description
The call forward service is not available on the telephone connected to the specific
endpoint. A call to this endpoint is not forwarded if the endpoint is busy.
Table 384: Activation State (Continued)

State
Active
Description
The call forward service is available on the telephone connected to the specific
endpoint. A call to the endpoint is forwarded to the specified destination if the
endpoint is busy. You must define the call forward destination in the Forwarding
Address field (Step 8). The call forward service behaves as if it is inactive if the
Forwarding Address is empty.
To let the user activate or deactivate this service with his or her handset, see steps 4, 5, and 6. In
that case, the field is automatically updated to reflect the activation status.
8.
Define the address to which forward incoming calls in the Forwarding Address field.
Accepted formats are:
telephone numbers (5551111)
SIP URLs such as scheme:user@host. For instance, sip:user@foo.com.
This string is used literally, so cosmetic symbols (such as the dash in 555-xxxx) should not be
present.
9.
Configuring Call Forward on Busy via Handset

The following is the procedure to use this service on the users telephone.
To forward calls:
1.
Take the receiver off-hook.
2.
Wait for the dial tone.
3.
Dial the sequence implemented to activate the call forward on busy service.
This sequence could be something like *72.
4.
Wait for the stutter dial tone (three beeps) followed by the dial tone.
5.
Dial the number to which you want to forward your calls. Dial any access code if required.
6.
Wait for three beeps followed by a silent pause.

The call forward is established.
7.
Hang up your telephone.
To cancel the call forward:

1.
2.
3.
Dial the sequence implemented to deactivate the call forward on busy service.
4.
Wait for the transfer tone (three beeps) followed by the dial tone.
The call forward is cancelled.
5.
415
Call Forward On No Answer

You can forward the incoming calls of your users to a pre-determined target if they do not answer their
telephone before a specific amount of time. The user does not have any feedback that a call was forwarded.
(Steps 3-5).
To set the Call Forward On No Answer feature:

1.
2.
In the Call Forward On No Answer section, define whether or not you want to override the Call
Forward On No Answer parameters set in the Default configuration in the Endpoint Specific dropdown menu.
Figure 197: Telephony Call Forward on No Answer section
2
3
4
6
8
3.
5
7
4.
enabled.
5.
status is enabled.
6.
416
Define the time, in milliseconds, the telephone keeps ringing before the call forwarding activates in
the Timeout field.
7.
Set the status of the service in the Activation field to Inactive or Active.
State
Description
Inactive
Active
8.
present.
9.
Configuring Call Forward on Answer via Handset

To forward calls:
1.
2.
3.
Dial the sequence implemented to activate the call forward on no answer service.
4.
5.
6.

7.

1.
2.
3.
Dial the sequence implemented to deactivate the call forward on no answer service.
4.
5.
417
Call Forward Unconditional

The Call Forward Unconditional feature allows users to forward all of their calls to another extension or line.
(Steps 3-5).
To set the Call Forward Unconditional feature:

1.
2.
In the Unconditional section, define if you want to override the Call Forward Unconditional
parameters set in the Default configuration in the Endpoint Specific drop-down menu.
Figure 198: Telephony Call Forward Unconditional Section
2
3
4
5
6
7
3.
4.
enabled.
5.
status is enabled.
418
6.
Set the status of the service in the Activation field to Inactive or Active.
State
Description
Inactive
Active
7.
present.
8.
Configuring Call Forward on Unconditional via Handset

When forwarding calls outside the system, a brief ring is heard on the telephone to remind the user that the
call forward service is active. The user can still make calls from the telephone.
To forward calls:
1.
2.
3.
Dial the sequence implemented to activate the call forward unconditional service.
4.
5.
6.

7.
To check if the call forward has been properly established:

1.
2.
3.
Dial your extension or telephone number.

The call is forwarded to the desired telephone number.
4.
419

1.
2.
3.
Dial the sequence implemented to deactivate the call forward unconditional service.
4.
5.
420
H A P T E R
37
Telephony Services
Configuration
This chapter describes how to set the following subscriber services:
Hook Flash Processing

Automatic call
Call completion
Delayed Hotline
Call Transfer
Call Waiting
Conference
Direct IP address call
Hold
Second call
Message Waiting Indicator
Some of the subscriber services are not supported on all Mediatrix unit models, so your specific model may
not have all subscriber services listed in this chapter.
Standards Supported
RFC 2976: The SIP INFO Method
The General Configuration sub-section of the Services Configuration section allows you to define the Hook
Flash Processing feature.
Note: Performing a flash hook and pressing the flash button means the same thing. However, not all
telephone models have a flash button.
421
To set general services parameters:

1.
In the web interface, click the Telephony link, then the Services sub-link.
Figure 199: Telephony Services Web Page
3
4
2.
3.
In the General Configuration sub-section, define whether or not you want to override the general
services parameters set in the Default configuration in the Endpoint Specific drop-down menu.
4.
Select how to process hook-flash detection in the Hook Flash Processing drop-down menu.
Hook flash processing allows hook flash signals to be transported over the IP network allowing to
use advanced telephony services. Users normally press the flash button of the telephone during
a call in progress to put this call on hold, transfer it, or even initiate a conference call.
You can define whether these subscriber services are handled by the unit or delegated to a remote
party. If services are to be handled by a remote party, a SIP INFO message is sent to transmit the
user's intention.
Note: The hook-flash processing attribute is not negotiated in SDP.

Table 387: Hook Flash Settings
Setting
Definition
Process Locally
The hook-flash is processed locally. The actual behaviour of the flash

button depends on which endpoint services are enabled for this
endpoint.
Transmit Using
Signaling Protocol
The hook-flash is processed by a remote party. The hook-flash event is

carried by a signaling protocol message. The actual behaviour of the
flash button depends on the remote party.
The hook-flash event is relayed as a SIP INFO message as described
in RFC 2976.
5.
422
Automatic Call
The automatic call feature allows you to define a telephone number that is automatically dialed when taking
the handset off hook.
When this service is enabled, the second line service is disabled but the call waiting feature is still functional.
The user can still accept incoming calls.
To set the automatic call feature:

1.
2.
In the Automatic Call sub-section, define whether or not you want to override the automatic call
Figure 200: Telephony Automatic Call Section
2
3
4
3.
Enable the service by setting the Automatic Call Activation drop-down menu to Enable.
4.
Define the string to dial when the handset is taken off hook in the Automatic Call Target field.
present.
5.
423
Call Completion
Standards Supported
RFC 4235: An INVITE-Initiated Dialog Event Package for the

Session Initiation Protocol (SIP)a
draft draft-poetzl-bliss-call-completion-00b
a. Implemented in client mode only and used for the call completion.
b. Implement the solution 1 in 5.1.
Note: This section applies only to the following models:

Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
The call completion service allows you to configure the Completion of Calls on No Reply (CCNR) and
Completion of Calls to Busy Subscriber (CCBS) features.
CCBS allows a caller to establish a call with a busy callee as soon as this callee is available to take the call.
It is implemented by monitoring the activity of a UA and look for the busy-to-idle state transition pattern.
CCNR allows a caller to establish a call with an idle callee right after this callee uses his phone. It is
implemented by monitoring the activity of a UA and look for the idle-busy-idle state transition pattern.
The information about the call completion is not kept after a restart of the EpServ service. This includes the
call completion activation in the Pots service and the call completion monitoring in the SipEp service.
To set the call completion feature:

1.
2.
In the Call Completion sub-section, define whether or not you want to override the call completion
Figure 201: Telephony Call Completion Section
3
4
5
7
6
8
9
10
11
12
13
3.
Enable or disable the (CCBS) service by selecting the proper value in the Allow CCBS Activation
Via Handset drop-down menu.
424
4.
If the CCBS service is enabled, define the digits that users must dial to start the service in the CCBS
DTMF Map Activation field.
You can use the same code in the CCNR DTMF Map Activation field.
enabled.
5.
Enable or disable the (CCNR) service by selecting the proper value in the Allow CCNR Activation
Via Handset drop-down menu.
6.
If the CCNR service is enabled, define the digits that users must dial to start the service in the CCNR
DTMF Map Activation field.
You can use the same code in the CCBS DTMF Map Activation field.
enabled.
7.
Define the digits that users must dial to stop the CCBS and CCNR services in the DTMF Map
Deactivation field.
For instance, you could decide to put *94 as the sequence to deactivate the services. This
status is enabled.
8.
Define the delay, in minutes, after the call completion activation to automatically deactivate the call
completion if the call is not completed in the Expiration Timeout field.
9.
Select the call completion method to detect that the call completion destination is ready to complete
the call in the Method drop-down menu.
Table 388: Call Completion Method Parameters
Method
Monitoring Only
Desciption
The call completion only uses the monitoring method to detect that the
destination is ready to complete the call.
Monitoring And Polling The call completion only uses the monitoring method to detect that the
destination is ready to complete the call. The polling mechanism is used
if the call completion destination cannot be monitored.
425
The monitoring method consists of using the protocol signalling to detect the destination state
without using the call. When the destination is ready to complete the call, the local user is notified
that the call is ready to be completed and the call to the destination is initiated when the user is ready
to initiate the call.
The polling method consists of using periodic calls to the call completion destination until the
destination responds with a ringing or connect. Upon receiving these responses, the local user is
notified that the call is ready to be completed.
The polling mechanism can only be used for call completion to busy subscriber (CCBS).
The retransmission of the polling mechanism is configurable with
DefaultCallCompletionPollingInterval.
10.
Enable or disable the call completion auto reactivation in the Auto Reactivate drop-down menu.
When enabled, the call completion busy subscriber is automatically activated if the call initiated by
a call completion busy subscriber or call completion no response fails because of a busy
destination.
11.
Define the minimal delay to wait, in seconds, before executing a call completion after its activation
in the Auto Reactivate Delay field.
This delay only applies to call completion activated via the call completion auto reactivation feature
(See Step 9).
Media5 recommends to set a delay when the method to monitor the target state is based on the
target calls instead of its ability to answer a call.
If the timeout is set to 0 and the target is off hook, the FXS endpoint always rings to notify that the
call completion is ready to be completed. However the call is always busy and thus reactivated
without the possibility for the user to cancel the call completion. The call completion will continue
until the ringing or call completion timeout or if the target became ready to receive call.
12.
Define how the call completion service needs to interpret the reception of a progress message with
early media in the Early Media Behaviour drop-down menu.
Table 389: Call Completion Early Media Behaviour Parameters
Parameter
Description
None
The progress message with early media is not considered as a busy or a ringing
response.
CCBS
The progress message with early media is interpreted as a busy response and the
CCBS can be activated on the call.
CCNR
The progress message with early media is interpreted as a ringing response and
the CCNR can be activated on the call.

13.
Define the delay, in seconds, between the calls to the call completion target used for the polling
mechanism in the Polling Interval field.
This parameter is used only if the Default Call Completion Method drop-down menu is set to
Monitoring And Polling.
14.
Special SIP Configuration

If you are using an Asterisk IP PBX, it returns the error code 503 instead of 486 for a busy destination when
the call limit is reached. The following error mapping can be required:
426
1.
Go to the page SIP > Misc.
2.
Insert a new mapping (with the plus button) in the SIP To Cause Error Mapping section.
3.
Set the SIP code to 503 Service Unavailable and the cause to 17 User busy.
4.
Click Submit.
Using the Call Completion Services

The following are the various procedures to use these services on the users telephone.
To start the CCBS (procedure 1)

The call has reached a busy destination and the busy tone is played.
1.
Dial the sequence implemented to enable the CCBS.

The confirmation tone is played.
2.
Hang up the telephone.

Alternatively, you can use procedure 2.
To start the CCBS (procedure 2)

The call has reached a busy destination and the busy tone is played.
1.
2.

The dial tone is played
3.
Dial the sequence implemented to enable the CCBS.

4.

Alternatively, you can use procedure 1.
To start the CCNR

The call has reached a destination but the call is still not yet established. A ring back or welcome message is
generally played at this moment.
1.
2.

3.
Dial the sequence implemented to enable the CCNR.

4.
To stop the CCBS or CCNR

1.

2.
Dial the sequence implemented to disable the CCBS and CCNR.

427
3.
Note: The CCBS and CCNR cannot be started to complete a second call.
When the call completion target is ready to receive a call:

1.
The telephone rings with the distinctive ringing Bellcore-dr2 (0.8 On 0.4 Off, 0.8 On 4.0 Off).
2.

The call is initiated to the call completion destination.
Call Transfer
Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
The Call Transfer service offers two ways to transfer calls:
Blind Transfer
Attended Transfer
To enable the Call Transfer services:

1.
2.
In the Call Transfer sub-section, define whether or not you want to override the call transfer
Figure 202: Telephony Call Transfer Web Page
2
3
4
3.
Enable the Blind Transfer service by setting the Blind Transfer Activation drop-down menu to
Enable.
The blind call transfer service is sometimes called Transfer without Consultation or Unattended
Transfer. It allows a user to transfer a call on hold to a still ringing (unanswered) call. The individual
at the other extension or telephone number does not need to answer to complete the transfer.
The call hold and second call services must be enabled for this service to work. See Call Hold on
page 439 and Second Call on page 440.
4.
428
Enable the Attended Transfer service by setting the Attended Transfer Activation drop-down menu
to Enable.
The attended call transfer service is sometimes called Transfer with Consultation. It allows a user
to transfer a call on hold to an active call. The individual at the other extension or telephone number
must answer to complete the transfer.
The call hold and second call services must be enabled for this service to work. See Call Hold on
page 439 and Second Call on page 440.
5.
Using Blind Call Transfer

To configure the SIP Blind Transfer Method, see SIP Blind Transfer Method on page 350.
To transfer a current call blind:

1.
Perform a Flash-Hook by pressing the Flash button on your analog telephone.

This puts the call on hold.
2.
Wait for the transfer tone (three beeps).
3.
Dial the number to which you want to transfer the call.
4.
Wait for the ringback tone, then hang up your telephone.

The call is transferred.
Once the transfer is executed, the remaining calls (call on hold and ringing call with third party) are
then connected together. The call on hold is automatically unheld and hears the ringback tone
provided by the third party's ringing.
You can also wait for the third party to answer if you want. In this case, the call transfer becomes
attended.
If you want to get back to the first call (the call on hold), you must perform a Flash-Hook.
You are back with the first call and the third party is released.
Using Attended Call Transfer

To transfer a current call attended:

1.

This puts the call on hold.
2.
Wait for the transfer tone (three beeps).
3.
Dial the number to which you want to transfer the call.

The third party answers.
4.

The call is transferred.
5.
If you want to get back to the first call (the call on hold), you must perform a Flash-Hook before the
target answers.
You are back with the first call and the third party is released.
Note: If the number to which you want to transfer the call is busy or does not answer, perform a Flash-Hook.
The busy tone or ring tone is cancelled and you are back with the first call.
429
Call Waiting
Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
The call waiting tone indicates to an already active call that a new call is waiting on the second line.
Your users can activate/deactivate the call waiting tone for their current call. This is especially useful when
transmitting faxes. The user that is about to send a fax can thus deactivate the call waiting tone to ensure that
the fax transmission will not be disrupted by an unwanted second call. When the fax transmission is completed
and the line is on-hook, the call waiting tone is automatically reactivated.
To set the Call Waiting services:

1.
2.
In the Call Waiting sub-section, define whether or not you want to override the call waiting
Figure 203: Call Waiting Section
2
3
4
3.
Enable the service by setting the Call Waiting Activation drop-down menu to Enable.
This permanently activates the call waiting tone. When receiving new calls during an already active
call, a special tone is heard to indicate that a call is waiting on the second line. The user can then
answer that call by using the flash button. The user can switch between the two active calls by
using the flash button.
The call hold service must be enabled for this service to work. See Call Hold on page 439.
If the user is exclusively using faxes, select Disable to permanently disable the call waiting tone.
4.
Define the digits that users must dial to disable the Call Waiting tone in the Cancel DTMF Map field.
This allows a user who has call waiting enabled to disable that service on the next call only. If, for
any reason, the user wishes to undo the cancel, unhook and re-hook the telephone to reset the
service.
For instance, you could decide to put *76 as the sequence to disable the call waiting tone. This
status is enabled.
430
5.
Using Call Waiting

The call waiting feature alerts the user if he or she is already on the telephone and a second call happens. A
beep (the call waiting tone) is heard and repeated every ten seconds to indicate there is a second incoming
call.
To put the current call on hold:

1.

This puts the call on hold and the second line is automatically connected to your line.
2.
Answer the call on the second line.
To switch from one line to the other:

1.
Perform a Flash-Hook each time you want to switch between lines.
To terminate the first call before answering the second call:

1.
2.
Wait for the telephone to ring.
3.
Answer the telephone.

The second call is on the line.
Removing the Call Waiting Tone

You can temporarily deactivate the call waiting tone indicating a call is waiting. This is especially useful when
transmitting faxes. If you are about to send a fax, you can thus deactivate the call waiting tone to ensure that
the fax transmission is not disrupted by an unwanted second call. When the fax transmission is completed and
the line is on-hook, the call waiting tone is automatically reactivated.
To deactivate the call waiting tone:

1.
2.
3.
Dial the sequence implemented to deactivate the call waiting tone.

4.
The call waiting tone is disabled.
IMS-3GPP Communication Waiting

Upon receipt of a SIP INVITE with multipart/mixed content where a valid IMS communication waiting indicator
is correctly specified such as in this example:
INVITE sip:...
[...]
Content-Type: multipart/mixed;boundary=boundary1
[...]
--boundary1
Content-Type: application/vnd.3gpp.cw+xml
Content-Disposition: render;handling=optional
<?xml version="1.0"?>
<ims-cw xmlns="urn:3gpp:ns:cw:1.0">
<communication-waiting-indication/>
431
</ims-cw>
--boundary1
Content-Type: application/sdp
[...]
--boundary1--
The 180 Ringing response to this may contain a special header :

Alert-Info: <urn:alert:service:call-waiting>
that is appended if all of the following are true :

1.
The INVITE contained the <communication-waiting-indication/> 3GPP option.
2.
The destination endpoint supports call waiting.
3.
The call waiting feature is enabled for this endpoint.
4.
The endpoint is currently in an active state (not ringing, not on hold, not on hook).
There are no variables to control this behaviour, it is always activated.
This header could be used by the server to notify the 2nd caller that the destination is currently busy
in a call but was notified of this new incoming call.
432
Conference
Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
The Conference Call service allows a user to link two or more calls together to form a single conversation,
called a conference.
Only 3-way conferences are currently supported.

A participant of the conference can put the conference on hold and attempt other calls. This
participant may then rejoin the conference at a later time by unholding it. The participant who
initiated the conference cannot put it on hold.
You must enable the call hold, second call and attended call transfer services for this service to work. See
Call Hold on page 439, Second Call on page 440, and Call Transfer on page 428.
The following is a conference call flow example:
Figure 204: Conference Call Flow
User
Agent
#2
(B)
INVITE (G.729)
Trying/ Ringing/200 OK
ACK
User
Agent
#1
(A)
User
Agent
#3
(C)
Flash Hook
INVITE (HOLD)
Trying /200 OK
ACK
INVITE (G.729 )
Trying/Ringing/200 OK
ACK
Flash Hook
Trying200 OK
ACK
INVITE (UNHOLD-G.729 )
Trying/200 OK
ACK
3-way Conference Call Established
DSP Limitation
The Mediatrix 4108, 4116, 4124, C7, LP16 and LP24 models currently suffer from a limitation of their DSPs.
When using a codec other than G.711, enabling Secure RTP (SRTP) and/or using conferences has an impact
on the Mediatrix units overall performance as SRTP and conferences require CPU power. That is the reason
why there is a limitation on the lines that can be used simultaneously, depending on the codecs enabled and
SRTP. This could mean that a user picking up a telephone on these models may not have a dial tone due to
lack of resources in order to not affect the quality of ongoing calls. See Security on page 201 for more details
on SRTP limitations.
The DSPs offer channels as resources to the Mediatrix unit. The Mediatrix unit is limited to two conferences
per DSP.
Please note that:
433
One FXS line requires one channel.

Each conference requires one additional channel
The Mediatrix 4108/C7 has one DSP
The Mediatrix 4116/LP16 have two DSPs
The Mediatrix 4124/LP24 have three DSPs
A total of eight channels per DSP are available when using unsecure communication, to be used between the
FXS lines and up to two conferences.
A total of six channels per DSP are available when using SRTP, to be used between the FXS lines and up to
two conferences.
Enabling the Conference Call Feature

You must enable this service before your users can use it.
To enable the Conference service:

1.
2.
In the Conference sub-section, define whether or not you want to override the conference
Figure 205: Conference Section
2
3
3.
Enable the service by setting the Conference Activation drop-down menu to Enable.
4.
Using an External Server for the Conference

Standards Supported
RFC 4579: Session Initiation Protocol (SIP) - Call Control Conferencing for User Agentsa
a. Partially compliant. Only call flows of sections 5.4 and 5.6 are supported. RFC 4575 is not supported.

by using the CLI
The Mediatrix unit can use an external server to mix the media of the conference. This conference type
requires the configuration of an external server. Using this type of conference does not affect the number of
simultaneous calls supported. You can use this feature only if the Conference service is enabled (see
Enabling the Conference Call Feature on page 434 for more details).
434
To use a server-based conference:

1.
In the EpServMIB, specify how to manage the conference by setting the defaultConferenceType
variable to the proper value.
This configuration only applies to a conference initiated by one of the unit's endpoint.
EpServ.defaultConferenceType="Value"

Table 390: Conference Type Parameters
Value
Parameter
Description
100
Local
The media of the conference is locally mixed by the unit. This

conference type does not require any special support of the call peer
or server. Using this type of conference can reduce the number of
simultaneous calls supported.
200
ConferenceSer The unit uses an external server to mix the media of the conference.
ver
This conference type requires the configuration of an external server
(See Step 3). Using this type of conference does not affect the number
of simultaneous calls supported.
In Local mode, the number of participants is limited to the unit's model capacity. In
ConferenceServer mode, the number of participants is limited by the server's capacity.
2.
If you want to set a different conference type for one or more endpoints, set the following variables:
epSpecificConferenceEnableConfig variable for the specific endpoint you want to
epSpecificConferenceType variable for the specific endpoint you want to configure to
the proper value.

EpServ.epSpecificConference.EnableConfig[Id="Specific_Endpoint"]="1"
EpServ.epSpecificConference.Type[Id="Specific_Endpoint"]="Type"
where:
3.
Specific_Endpoint is the number of the endpoint you want to configure.
Value is the type as defined in Step 1.
If you have set the Conference type to ConferenceServer, in the SipEpMIB, set the
defaultConferenceType variable with the URI used in the request-URI of the INVITE sent to the
conference server as defined in RFC 4579.

SipEp.DefaultStaticConferenceServerUri="URI"
4.
If you want to set a different URI for one or more endpoints, set the following variables:
GwSpecificConferenceEnableConfig variable for the specific endpoint you want to
GwSpecificConferenceServerUri variable for the specific endpoint you want to

EpServ.GwSpecificConference.EnableConfig[Id="Specific_Endpoint"]="1"
EpServ.GwSpecificConference.ServerUri[Id="Specific_Endpoint"]="URIValue"
where:
435
URIValue is the URI you want to use.
Managing a Conference Call

If you are on the telephone with one person and want to conference with a third one, you can do so. In the
following examples, lets assume that:
A is the conference initiator.
E is a fifth person that C wants to add to the conference in conferenceServer conference

type.
B is the person called on the first line.

C is the person called on the second line.
D is a fourth person that A wants to add to the conference in conferenceServer conference
type.
To initiate a three-way conference (A and B already connected):

1.
A performs a Flash-Hook.
This puts B on hold and the second line is automatically connected. A hears a dial tone.
2.
A dials Cs number.
A and C are now connected.
3.
A performs another Flash-Hook.

The call on hold (B) is reactivated. A is now conferencing with B and C.
B (or C) hangs up during the conference:

1.
B (or C) hangs up during the conference.

The conference is terminated, but the call between A and C (or B) is not affected and they are
still connected.
A (conference initiator) hangs up during the conference:

1.
A hangs up.
The conference is terminated, both call "C" and "B" are also terminated.
A wants to add a fourth member to the conference:

This is available only in the conferenceServer conference type.
1.
A performs a Flash-Hook.
A hears a dial tone. The second line is automatically connected. B and C are still in conference.
2.
A dials Ds number.
A and D are now connected.
3.
A performs another Flash-Hook.

A is now conferencing with B, C, and D.
"C" wants to add a fifth member to the conference:

This is available only in the conferenceServer conference type.
1.
"C" performs a Flash-Hook.

"C" hears a dial tone. The second line is automatically connected. "A ", "B " and "D " are still in
conference.
2.
"C" dials "E's" number.

"C" and "E" are now connected.
3.
436
"C" performs another Flash-Hook.

"E" is now conferencing with "A ", "B", "C", and "D".
Delayed Hot Line

Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
The delayed hot line feature (also called warm line) is used to make an automatic call to a specified address
on the two following conditions:
When the user picks up the phone but does not dial any digit. The configured destination is
automatically called upon picking up the phone and after waiting for the configurable number
of seconds without dialling.
When the user starts dialing but does not complete a valid number before the timeout set in the
Delayed Hotline Condition drop-down menu expires.
The condition on which the delayed hotline is activated is configurable. This feature thus places an automatic
call whenever the Delayed Hotline Condition timeout expires. It could be used as an alternative to the
emergency number (for instance, the 911 number in North America).
To configure the basic delayed hot line feature:

1.
2.
In the Delayed Hotline sub-section, define whether or not you want to override the delayed hotline
Figure 206: Delayed Hotline Section
2
3
3.
Enable the service by setting the Delayed Hotline Activation drop-down menu to Enable.
When the feature is disabled, a user picking up the phone but not pressing any telephone keys
hears the Receiver Off-Hook tone after the amount of time specified in the
digitMapTimeoutFirstDigit variable.
4.
437
To configure the delayed hotline activation condition:

1.
In the Delayed Hotline sub-section, select the condition(s) that activate the delayed hotline in the
Delayed Hotline Condition drop-down menu.
Table 391: Delayed Hotline Conditions

Parameter
Description
FirstDtmfTimeout
The delayed hotline is activated when the timeout configured

in the First DTMF Timeout field of the Telephony > DTMF
Maps page elapses (General DTMF Maps Parameters on
page 408).
InterDtmfOrCompletionTimeout The delayed hotline is activated when the timeout configured

in the Completion Timeout field of the Telephony > DTMF
Maps page elapses or when the DTMFs collection fails
because the Inter DTMF Timeout parameter elapses
(General DTMF Maps Parameters on page 408).
AnyTimeout
2.
The delayed hotline is activated when the timeout configured

in the Completion Timeout field of the Telephony > DTMF
Maps page elapses and when the DTMFs collection fails
because the Inter DTMF Timeout parameter elapses
(General DTMF Maps Parameters on page 408).
To configure the delayed hotline target:

1.
In the Delayed Hotline sub-section, set the destination (address or telephone number) that is
automatically called in the Delayed Hotline field.
present.
2.
Direct IP Address Call

The IP address call service allows a user to dial an IP address without the help of a SIP server. Using this
method bypasses any server configuration of your unit.
The user can dial an IP address and enter an optional telephone number. Note that the optional telephone
number is matched by using the same digit maps as a normal call.
The IP address call method can be used when a SCN user wants to reach a LAN endpoint.
438
To set the direct IP call feature:

1.
This menu is available only in the default endpoints configuration.
2.
Enable the service by setting the Direct IP Address Call drop-down menu to Enable.
Figure 209: Telephony Direct IP Address Call Section
2
Dialing an IP Address
To make an IP address call:
1.
Dial ** (IP address prefix).
2.
Dial the numerical digits of the IP address and use the * for the . of the IP address.
3.
Dial * to terminate the IP address if you do not need to specify a phone number.
For instance, lets say you want to reach a one-line access device or another LAN endpoint such as
an IP Phone with the IP address 192.168.0.23. You must then dial the following digits:
**192*168*0*23*
4.
If you need to specify the phone number of a specific line, dial # to terminate the IP address.
5.
Dial the telephone number of the specific line you want to reach.
For example, lets say you want to reach the telephone connected to Line 2 of the Mediatrix unit with
the IP address 192.168.0.23. The phone number assigned to Line 2 of this Mediatrix unit is 1234.
You must then dial the following digits:
**192*168*0*23#1234
In this case, the Mediatrix unit sends an INVITE 1234@192.168.0.23.
Call Hold
Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
The Call Hold service allows the user to temporarily put an existing call on hold, usually by using the flash
button of the telephone. The user can resume the call in the same way.
You must enable this service for the following services to work properly:
Call Waiting
Second Call
Blind Transfer
Attended Transfer
Conference
439
To enable the Call Hold service:

1.
2.
In the Hold sub-section, define whether or not you want to override the call hold parameters set in
the Default configuration in the Endpoint Specific drop-down menu.
Figure 210: Hold Section
2
3
3.
Enable the service by setting the Hold Activation drop-down menu to Enable.
4.
Using Call Hold

To put the current call on hold:

1.

This puts the call on hold. You can resume the call in the same way.
Second Call
Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
The Second Call service allows a user with an active call to put the call on hold, and then initiate a new call on
a second line. This service is most useful with the transfer and conference services.
The call hold service must be enabled for this service to work. See Call Hold on page 439.
You must enable this service for the following services to work properly:
Blind Transfer
Attended Transfer
Conference
To enable the Second Call service:

1.
2.
440
In the Second Call sub-section, define whether or not you want to override the second call

Figure 211: Second Call Section
2
3
3.
Enable the service by setting the Second Call Activation drop-down menu to Enable.
4.
Using Second Call

To use the second call service:

1.

This puts the call on hold and the second line is automatically connected to your line.
2.
Initiate the second call.

Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
Standards Supported
RFC 3265: Session Initiation Protocol (SIP)-Specific Event

Notificationa
RFC 3842: The Message Summary and Message Waiting

Indication Event Package for the Session Initiation Protocol
(SIP)b
a. Supports receiving blind NOTIFY without subscribing. Sending blind NOTIFY is not supported.
b. Supports receiving blind NOTIFY without subscribing. Sending blind NOTIFY is not supported.

by using the CLI
The Message Waiting Indicator (MWI) service alerts the user when new messages have been recorded on a
voice mailbox. It is enabled by default.
After the message is recorded, the server sends a message (SIP NOTIFY request) to the Mediatrix unit listing
how many new and old messages are available. The Mediatrix unit alerts the user of the new message in two
different ways:
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The telephones LED blinks (if present). A FSK signal is sent on the FXS line.
A message waiting stutter dial tone replaces the normal dial tone when the user picks up the
FXS line.
Note: The message waiting state does not affect the Second Call feature. When in an active call,
performing a flash-hook to get access to the second line plays the usual dial tone.
The Mediatrix unit supports to receive SIP MWI notifications via SIP NOTIFY requests as defined in RFC 3842
but with the following limitations/diversions:
In addition to the SIP event string "message-summary" (RFC 3842), the string "simplemessage-summary" is accepted. The significations of those strings are identical.
In addition to the SIP content type string "simple-message-summary" (RFC 3842), the string
"message-summary" is accepted. The significations of those strings are identical.
Support of message-summary is not advertised in the SIP REGISTER.
Note that received SIP NOTIFY with an event different than "message-summary" or "simple-messagesummary" is not interpreted as a valid MWI notification.
To disable the Message Waiting Indicator service:

1.
In the potsMIB, set the fxsDefaultMessageWaitingIndicatorActivation variable to the proper

value.
pots.fxsDefaultMessageWaitingIndicatorActivation="100"
If you want to reactivate the feature, use the following:

pots.fxsDefaultMessageWaitingIndicatoreActivation="Value"

Table 392: Message Waiting Indicator Parameters
Value
2.
Parameter
Description
100
Disabled
The user is not alerted of messages awaiting attention.
200
Tone
When messages are awaiting attention, the user is alerted by a

message waiting tone when picking up the handset.
300
Visual
When messages are awaiting attention, the user is alerted by a Visual

Message Waiting Indicator such as a blinking LED on the phone.
400
ToneAndVisual When messages are awaiting attention, the user is alerted by a Visual
Message Waiting Indicator such as a blinking LED on the phone, and
a message waiting tone when picking up the handset.
If you want to set a different activation for one or more endpoints, set the following variables:
fxsSpecificMessageWaitingIndicatorEnableConfig variable for the specific
endpoint you want to configure to enable.
fxsSpecificMessageWaitingIndicatorActivation variable for the specific endpoint
you want to configure to the proper value.

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Distinctive Call Waiting Tone
pots.fxsSpecificMessageWaitingIndicator.EnableConfig[Id="Specific_Endpoint"]="1"
pots.fxsSpecificMessageWaitingIndicator.Activation[Id="Specific_Endpoint"]="Valu
e"
where:
Value is the activation as defined in Step 1.
Visual Message Waiting Indicator Type

You can configure how the Visual Message Waiting Indicator is sent on FXS lines.
To configure the visual message waiting indicator type:

1.
In the potsMIB, set the fxsDefaultVisualMessageWaitingIndicatorType variable to the proper

value.
pots.fxsDefaultVisualMessageWaitingIndicatorType="Value"

Table 393: Visual Message Waiting Indicator Type Parameters
Value
Parameter
Description
100
Fsk
A FSK signal is sent to activate the VMWI on the phone.
200
FskAndVoltage Both FSK signal and high voltage signal are used to activate the VMWI
on the phone.
Note: This parameter applies only to the following models:
Mediatrix LP Series
Distinctive Call Waiting Tone

The distinctive call waiting tone configuration allows the administrator to modify the pattern of the tone.
Two variables are used:
ToneId: Allows the identification of the distinctive call waiting tone. If the distinctive ring callproperty matches the ToneId, the distinctive tone will be used.
Pattern: Describes the tone pattern.
A tone pattern contains:

1.
Frequencies
Up to 4 frequencies (f1 to f4) each with a power level can be defined. At least one frequency/power
pair must be defined. Frequency range is from 10 to 4000 Hz and Power level range is from -99 to
3 dbm.
The syntax is: f1=<frequency>:<power>
2.
States
Up to 8 states (s1 to s8) can be defined, each with an action, a set of frequencies, a duration and a
next state. At least one state must be described if the tone-pattern is not empty.
The action can be 'on', 'off' or 'CID' (for call waiting tones).
The duration of the state is from 10 to 56000 ms.
The tone is continuous if no time is specified.
443
Call Statistics
The syntax is: s1=<action>:<frequency>:...:<frequency>:<duration>:<end-of-loop-indicator>:<next state>

3.
Loops
A set of states can be enclosed in a loop.
The starting state of a loop is marked with a loop counter (l=), the range is from 2 to
128.
The ending state of a loop is marked with an end-of-loop indicator (l).
The syntax is: l=<loop count>,<state definition>,...,<state definition (with end-of-loopindicator)>,<state definition>...
Examples:
Germany dialtone (continuous): "f1=350:-17,f2=440:-17,s1=on:f1:f2"
North America Recall dialtone (3 quick tones followed by a continuous tone): "f1=350:17,f2=440:-17,l=3,s1=on:f1:f2:100:s2,s2=off:100:l:s1,s3=on:f1:f2"
Australia ring back tone (on 400ms, off 200 ms, on 400 ms and off 2000 ms and
replay): "f1=425:-17,f2=400:-5,f3=450:5,s1=on:f1:f2:f3:400:s2,s2=off:200:s3,s3=on:f1:f2:f3:400:s4,s4=off:2000:s1"
Only two frequencies can be used by the Call Waiting tone.

The parameters can be set :

by using the CLI
To set the distinctive call waiting tone:

1.
2.
In the Tellf MIB set :
Tellf.DistinctiveCallWaitingPattern variable in the CallWaitingToneGroup table
Tellf.DistinctiveCallWaitingRingId variable in the CallWaitingToneGroup table.
or
Use the CLI or a configuration script:
Tellf.DistinctveCallWaiting[Index=value].Pattern=value
Tellf.DistinctveCallWaiting[Index=value].ToneId=value
Index value can vary from 1 to 4.
Call Statistics
This section describes how to access data available only in the MIB parameters of the Mediatrix unit. You can
display these parameters as follows:

by using the CLI
The following are the call statistics the Mediatrix unit keeps. Statistics are updated at the end of each call.
Table 394: Call Statistics
MIB Variable
IncomingCallsReceived
444
Number of incoming IP calls received on the endpoint
since service start.
Call Statistics
Table 394: Call Statistics

MIB Variable
IncomingCallsAnswered
Number of incoming IP calls answered on the endpoint
IncomingCallsConnected Number of incoming IP calls that successfully completed

call setup signaling on the endpoint since service start.
IncomingCallsFailed
Number of incoming IP calls that failed to complete call

setup signaling on the endpoint since service start.
OutgoingCallsAttempted
Number of outgoing IP calls attempted for the endpoint

OutgoingCallsAnswered
Number of outgoing IP calls answered by the called

party for the endpoint since service start.
OutgoingCallsConnected Number of outgoing IP calls that successfully completed

call setup signaling for the endpoint since service start.
OutgoingCallsFailed
Number of outgoing IP calls that failed to complete call

setup signaling for the endpoint since service start.
CallsDropped
Number of IP calls, on the endpoint since service start,

that were successfully connected (incoming or
outgoing), but dropped unexpectedly while in progress
without explicit user termination.
TotalCallTime
Cumulative duration of all IP calls on the endpoint since

service start, in seconds.
To display call statistics:

1.
In the epServMIB, go to the CallStatistics table.

get epServ.callStatistics
To reset call statistics values to zero:

1.
In the epServMIB, set callStatistics.Reset to Reset for the endpoint to reset.

set epServ.callStatistics.Reset=Reset
2.
In the epServMIB, set callStatistics[EplId=callStatisticsEpId].Reset to Reset to reset

only one specific endpoint.
where:
callStatisticsEpId is the string that identifies the combination of an endpoint and a

channel. The endpoint name is the same as the EpId used to refer to endpoints in other
tables. On endpoints with multiple channels, the channel number must be appended at
the end of the endpoint name, separated with a dash.
set epServ.callStatistics[EplId=callStatisticsEpId].Reset=Reset
Examples:
Slot3/E1T1-12 refers to endpoint Slot3/E1T1, channel 12.
Phone-Fax1 refers to FXS endpoint Phone-Fax1 on a 4102s.
Port06 refers to FXS endpoint Port06 on 4108/4116/4124.
No channel number is appended to FXS endpoint strings because FXS lines do not support multiple
channels.
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Default Outbound Priority Call Routing
Default Outbound Priority Call Routing


by using the CLI
You can define how to route priority calls including emergency calls.
To set the default outbound priority call routing:

1.
In the sipEpMIB, set the defaultOutboundPriorityCallRouting variable to the proper value.

sipEp.defaultOutboundPriorityCallRouting="Value"

Table 395: Default Outbound Priority Call Routing Parameters
Value
446
Parameter
Description
100
Normal
Sends the call using normal SIP call routing to the outbound proxy (if
defined) and to the target host (usually the SIP server).
200
SkipOutbound
Proxy
Sends the call directly to the configured server skipping the outbound
proxy.
H A P T E R
38
Tone Customization Parameters

Configuration
This chapter describes how to override the pattern for a specific tone defined for the selected country (see
Appendix A - Country-Specific Parameters on page 621 for more details). It covers the following topics:
Current Tone Definition

Tone Override
Current Tone Definition

The Tone Customization page allows you to both see the current definition and override the pattern of the
following tones:
Busy
Call Waiting
Confirmation
Congestion
Dial
Hold
Intercept
Message Waiting
Preemption
Reorder
Ringback
Receiver Off Hook (ROH)
Special Information Tone (SIT)
Stutter
This includes the number of frequencies used, the tone value in Hertz (Hz), its power in dBm, as well as the
states configured.
To see the current definition of a tone:

1.
In the web interface, click the Telephony link, then the Tone Customization sub-link.
Figure 212: Telephony Tone Customization Web Page
2.
Select the proper tone to see in the Select Tone drop-down menu at the top of the window.
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Chapter 38 - Tone Customization Parameters Configuration
Tone Override
The Current Tone Definition and Current Tone States sections describe the current definition of the
selected tone.
Tone Override
You can override the pattern for a specific tone. This is done in two sections:
Table 397: Tone Override Sections
State
Description
Overridden Tone Definition Allows you to define up to four frequencies (F1 to F4). You must enter at least
one frequency.
Overridden Tone States
Description of the tone state. You can define up to eight states. You must
enter at least one state.
To override the pattern of a tone:

1.
Select which tone you want to override in the Override Current Tone Values drop-down menu.
Figure 213: Tone Override Sections
1
2
2.
You can use the current values of the selected tone as a starting point for your
customization by clicking the Copy Current Tone Definition to Overridden button.
You can clear all override fields by clicking the Reset Overridden Values button.
In the Overridden Tone Definition section, define the value of the proper Frequency used in the
corresponding Value field.
The value is in Hz. The range is from 10 Hz to 4000 Hz.
Note: You can use only two frequencies for the Call Waiting tone.
3.
Define the power level of the proper Frequency in dBm in the corresponding Power field.
The range is from -99 dBm to 3 dBm.
4.
If applicable, enter a value for the loop counter in the Loop Count field.
The range is from 2 to 128. This value will be used in Step 8.
Note: You can use only one loop count for the Call Waiting tone.
5.
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In the Overridden Tone States section, set the corresponding On/Off drop-down menu with the
proper value for each state.
Tone Override
On means the corresponding state plays a tone.
Off means the corresponding state does not play a tone.
CID means the moment where the Caller-ID will be sent to the analog port. This options
is available only for the Call Waiting tone.
You may also want to perform the following operations:
6.
To add a state, click the
button at the bottom of the Overridden Tone States section.
To remove a state, click the

button at the bottom of the Overridden Tone States
section. This removes the last state in the list.
For the On states, select the frequency to play in the corresponding Frequencies column.
The frequencies defined in the Overridden Tone Definition section are listed as clickable buttons.
You can use from one to four frequencies. A blue button indicates that the frequency is selected.
7.
Set the corresponding Duration field with the number of times, in ms, to perform the action of the
state.
The range is from 10 ms to 56000 ms. The tone stays indefinitely in the state (continuous) if no time
is specified.
8.
In the corresponding Loop drop-down menu, select whether or not to stop looping between states
after a number of loops defined in Step 4.
When the number of loops is reached, the next state is s(n+1) for the state s(n) instead of the state
defined in the Next State drop-down menu.
9.
In the corresponding Next State drop-down menu, select the next tone state to use when the time
has elapsed.
This value is not available if the Duration field is empty.
10.
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Chapter 38 - Tone Customization Parameters Configuration
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Tone Override
H A P T E R
39
Music on Hold Parameters

Configuration
This chapter describes how to configure the Music on Hold (MoH) parameters.
MP3 file download server setup.

Music on Hold configuration.
Standards Supported
RFC 1350: The TFTP Protocol (Revision 2) (client-side only)
RFC 2616: Hypertext Transfer Protocol - HTTP/1.1 (clientside only)
MP3 File Download Server

To download a MP3 file, you may need to setup the following applications on your computer:
TFTP server with proper root path

HTTP server with proper root path
Configuring the TFTP Server

When you perform a MP3 file download by using the TFTP (Trivial File Transfer Protocol) protocol, you must
install a TFTP server running on the PC designated as the TFTP server host. It is assumed that you know how
to set the TFTP root path. If not, refer to your TFTP servers documentation.
Configuring the HTTP Server

When you to perform a MP3 file download by using the HTTP protocol, you must install a HTTP server running
on the PC designated as the server host. It is assumed that you know how to set the root path. If not, refer to
your HTTP servers documentation.
Music on Hold Configuration

The Music on Hold sub-page of the Telephony page allows you to configure the music (in the form of an MP3
file) that plays when a local user has been put on hold. Note that transfers exceeding 5 minutes are cancelled.
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Chapter 39 - Music on Hold Parameters Configuration
To set the Music on Hold parameters:

1.
In the web interface, click the Telephony link, then the Music on Hold sub-link.
Figure 214: Telephony Music on Hold Web Page
2
3
4
5
2.
In the Music On Hold Configuration section, indicate whether or not the unit should play music when
being put on hold in the Streaming drop-down menu.
When enabled, music is played toward the telephony side when being put on hold from the network
side.
3.
In the Transfer Configuration section, enter the URL to the MP3 file to use in the URL field.
This file is loaded when the Mediatrix unit starts and reloaded every time the Reload Interval value
elapses (see Step 5). It must be smaller than 1024 Kilobytes unless otherwise specified in a
customer profile.
The MP3 file downloaded must be encoded with a sampling rate of 8000 Hz (only available through
MPEG version 2.5) and in mono channel mode. All other types of file will be rejected. The decoding
output will be in mono channel mode, with a sample rate of 8000 Hz and with 8 bits per sample.
You can use the following supported protocols to transfer the file:
HTTP: HyperText Transfer Protocol.
TFTP: Trivial File Transfer Protocol.
URLs using any other transfer protocol are invalid.
Note: The HTTP protocol does not support spaces between characters in the URL.
Examples of valid URLS:
http://www.myserver.com/myfile.mp3
tftp://myserver.com:69/myfolder/myfile.mp3
When the port is not included in the URL, the default port for the chosen protocol is used.
HTTP supports basic or digest authentication mode as described in RFC 2617.
If you have selected HTTP, please note that your server may activate some caching mechanism for
the MP3 download. This mechanism caches the initial MP3 download for later processing, thus
preventing changes of the original MP3.
4.
If your server requires authentication when downloading the MP3, set the following:
The user name in the User Name field.
The password in the Password field.
452
5.
Set the time, in hours, between attempts to load the MP3 file in the Reload Interval field.
If you enter the value 0, this means that the unit loads the file only once at unit startup. Any other
value between 1 and 6000 is the number of hours between automatic reloads of the file. When a
manual file download is triggered, the counter is not reset so the next reload will happen at the same
time.
6.
To save your settings without transferring the MP3 file, click Submit.
To save your settings and transfer the MP3 file now, click Submit & Transfer Now.
To save your settings and stop a file transfer in progress, click Submit & Cancel
Transfer.
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Chapter 39 - Music on Hold Parameters Configuration
454
H A P T E R
40
Country Parameters
Configuration
This chapter describes how to configure the country information:
Select a specific country.

Additional country settings.
Country Configuration
The Misc sub-page of the Telephony page allows you to configure the country in which the unit is located.
To set the miscellaneous parameters:

1.
In the web interface, click the Telephony link, then the Misc sub-link.
Figure 215: Telephony Misc Web Page
2.
In the Country section, select the country in which the Mediatrix unit is located in the Country
Selection drop-down menu.
It is very important to set the country in which the unit is used because a number of parameter
values are set according to this choice, such as tones, rings, impedances, and line attenuations.
See Appendix A - Country-Specific Parameters on page 621 for more information on these
country-specific settings.
3.
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Additional Country Settings


by using the CLI

Input/Output User Gain

The user gain allows you to modify the input and output sound level of the Mediatrix unit.
Caution: Use these settings with great care. Media5 recommends not to modify the user gain variables
unless absolutely necessary because default calibrations may no longer be valid.
Modifying user gains may cause problems with DTMF detection and voice quality using a high user gain
may cause sound saturation (the sound is distorted). Furthermore, some fax or modem tones may no longer
be recognized. The user gains directly affect the fax communication quality and may even prevent a fax to
be sent.
You can compensate with the user gain if there is no available configuration for the country in which the
Mediatrix unit is located. Because the user gain is in dB, you can easily adjust the loss plan, e.g., if you need
an additional 1 dB for analog to digital, put 1 for user gain output.
You can use two types of configuration as described in Default vs. Specific Configurations on page 456.
To set user gain variables:

1.
In the telIfMIB, locate the countryCustomizationUserGainGroup folder.
2.
On a call involving a SIP terminal and an FXS terminal, raising the output offset will raise the volume
perceived on the FXS terminal. Define the default user output gain offset in dB in the
defaultCountryCustomizationUserGainOutputOffset variable.
telIf.defaultCountryCustomizationUserGainOutputOffset="Value"
Values range from -12 dB to +12 dB. However, going above +6 dB may introduce clipping/distortion
depending on the country selected.
3.
If you want to set a different output gain offset for one or more interfaces, set the following variables:
specificCountryCustomizationUserGainEnableConfig variable for the specific
interface you want to configure to enable.
specificCountryCustomizationUserGainOutputOffset variable for the specific line
you want to configure.

telIf.specificCountryCustomizationUserGain.EnableConfig[InterfaceId="Interface"]
="1"
456
telIf.specificCountryCustomizationUserGain.OutputOffset[InterfaceId="Interface"]
="Value"
where:
4.
Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).
Value is the output gain offset.
On a call involving a SIP terminal and an FXS terminal, raising the input offset will raise the volume
perceived on the SIP terminal. Define the default user input gain offset in dB in the
defaultCountryCustomizationUserGainInputOffset variable.
telIf.defaultCountryCustomizationUserGainInputOffset="Value"
Values range from -12 dB to +12 dB. However, going above +6 dB may introduce clipping/distortion
depending on the country selected.
5.
If you want to set a different input gain offset for one or more interfaces, set the following variables:
specificCountryCustomizationUserGainEnableConfig variable for the specific
specificCountryCustomizationUserGainInputOffset variable for the specific line
you want to configure.

telIf.specificCountryCustomizationUserGain.EnableConfig[InterfaceId="Interface"]
="1"
telIf.specificCountryCustomizationUserGain.InputOffset[InterfaceId="Interface"]=
"Value"
where:
6.
Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).
Value is the input gain offset.
Restart the TelIf service by accessing the scmMIB and setting the serviceCommandsRestart
variable for the TelIf service to restart.
scm.serviceCommands.Restart[Name=TelIf]="10"
Dialing Settings
Dialing settings allow you to configure how the Mediatrix unit dials numbers.
When selecting a country (see Country Configuration on page 455 for more details), each country has default
dialing settings. However, you can override these values and define your own dialing settings.
To set the dialing settings:

1.
In the telIfMIB, locate the countryCustomizationDialingGroup folder.
2.
Set the defaultCountryCustomizationDialingOverride variable to enable.

telIf.specificCountryCustomizationDialing.EnableConfig[InterfaceId="Interface"]=
"1"
where Interface is the name of the interface you want to configure (for instance, Slot2/Pri1).
This allows overriding the default country settings.
3.
If you want to change the override status for one or more interfaces, set the following variables:
specificCountryCustomizationDialingEnableConfig variable for the specific
specificCountryCustomizationDialingOverride variable for the specific interface
you want to configure to enable.

457
"1"
telIf.specificCountryCustomizationDialing.Override[InterfaceId="Interface"]="1"
4.
Set an inter-digit dial delay in the defaultCountryCustomizationDialingInterDtmfDialDelay

variable.
telIf.defaultCountryCustomizationDialing.InterDtmfDialDelay="Value"
This is the delay, in milliseconds (ms), between two DTMFs when dialing the destination phone
number. Values range from 50 ms to 600 ms.
5.
If you want to set a different inter-digit dial delay for one or more interfaces, set the following
variables:
specificCountryCustomizationDialingInterDtmfDialDelay variable for the
specific interface you want to configure.

"1"
telIf.specificCountryCustomizationDialing.InterDtmfDialDelay[InterfaceId="Slot3/
Bri3"]="Value"
6.
Set the DTMF duration value in the defaultCountryCustomizationDialingDtmfDuration

variable.
telIf.defaultCountryCustomizationDialing.DtmfDuration="Value"
This is the duration, in milliseconds (ms), a DTMF is played when dialing the destination phone
number. Values range from 50 ms to 600 ms.
7.
If you want to set a different DTMF duration value for one or more interfaces, set the following
variables:
specificCountryCustomizationDialingDtmfDuration variable for the specific
interface you want to configure.

"1"
telIf.specificCountryCustomizationDialing.DtmfDuration[InterfaceId="Interface"]=
"Value"
8.
Set the delay, in milliseconds, between two MFR1s when dialing on the interface in the
DefaultCountryCustomizationDialingInterMfR1DialDelay variable.
See Chapter 23 - E&M CAS Configuration on page 255 for more details on MFR1 signalling.
9.
Set the delay, in milliseconds, between two MFR1s when dialing on the interface by putting the
following line in the configuration script:
telIf.defaultCountryCustomizationDialing.InterMfR1DialDelay="Value"
Values range from 50 ms to 600 ms.

10.
458
If you want to set a different delay value for one or more interfaces, set the following variables:
"1"
telIf.specificCountryCustomizationDialing.InterMfR1DialDelay[InterfaceId="Interf
ace"]="Value"
11.
Set the duration, in milliseconds, of a MFR1 when dialling on the interface in the
DefaultCountryCustomizationDialingMfR1Duration variable.
See Chapter 23 - E&M CAS Configuration on page 255 for more details on MFR1 signalling.
12.
Set the duration, in milliseconds, of a MFR1 when dialing on the interface by putting the following
line in the configuration script:
telIf.DefaultCountryCustomizationDialing.MfR1Duration="Value"
Values range from 50 ms to 600 ms.

13.
If you want to set a different duration value for one or more interfaces, set the following variables:
"1"
telIf.specificCountryCustomizationDialing.MfR1Duration[InterfaceId="Interface"]=
"Value"
14.
Restart the TelIf service by accessing the scmMIB and setting the serviceCommandsRestart
variable for the TelIf service to restart.
scm.serviceCommands.Restart[Name=TelIf]="10"
Fax Calling Tone Detection

You can enable the fax calling tone (CNG tone) detection.
To enable fax calling tone detection:

1.
In the telIfMIB, locate the machineDetectionGroup folder.
2.
Set the defaultMachineDetectionCngToneDetection variable to enable.

telIf.defaultMachineDetection.CngToneDetection="1"
Upon recognition of the CNG tone, the Mediatrix unit switches the communication from voice mode
to fax mode and the CNG is transferred by using the preferred fax codec. This option allows for
quicker fax detection, but it also increases the risk of false detection.
If you do not want the Mediatrix unit to detect the fax calling tone, set the variable to disable(0). In
this case, the CNG tone does not trigger a transition from voice to data and the CNG is transferred
in the voice channel. With this option, faxes are detected later, but the risk of false detection is
reduced.
3.
If you want to set a different calling tone detection setting for one or more interfaces, set the
following variables:
specificMachineDetectionEnableConfig variable for the specific interface you want
to configure to enable.
specificMachineDetectionCngToneDetection variable for the specific interface you
want to configure.
telIf.specificMachineDetection.EnableConfig[InterfaceId="Interface"]="1"
telIf.specificMachineDetection.CngToneDetection[InterfaceId="Interface"]="Value"
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460
H A P T E R
41
Call Detail Record

This chapter describes how to configure call detail record:
CDR (Call Detail Record)

Call detail record (CDR) in VoIP contains information about recent system usage such as the identities of
sources (points of origin), the identities of destinations (endpoints), the duration of each call, the total usage
time in the billing period and many others.
The Misc sub-page of the Telephony page allows you to configure the CDR parameters.
To set the CDR parameters:

1.
In the Call Detail Record section of the Misc page, set the host name and port number of the device
that archives CDR log entries in the Syslog Remote Host field.
Specifying no port (or port 0) sends notifications to port 514.
Figure 216: CDR Call Detail Record Section
1
3
2.
Specify the format of the syslog Call Detail Record in the Syslog Format field.
The formal syntax description of the protocol is as follows:
Precision=DIGIT
Width=DIGIT
MacroId=(ALPHA / "_")
Macro=%[Width]|[.Precision]|[Width.Precision]MacroId
The Width field is the minimum width of the converted argument. If the converted argument has
fewer characters than the specified field width, then it is padded with spaces. If the converted
argument has more characters than the specified field width, the field width is extended to whatever
is required.
The Precision field specifies the maximum number of characters to be printed from a string.
Examples :
sipid=SipUser001
CDR Log: %sipid
CDR Log: %15sipid
CDR Log: %15.5sipid
CDR Log: %.5sipid
-->
-->
-->
-->
CDR
CDR
CDR
CDR
Log
Log
Log
Log
:
:
:
:
SipUser001
SipUser001
SipUs
SipUs
Call Detail Record predefined macros.

Control characters:
Table 398: Control Character
Character
Value
%%
\n
Split message
461
Chapter 41 - Call Detail Record
Call detail record macros:

Table 399: Call Detail Record Macros
Macro
Value
%id
CDR ID. The CDR ID is unique. The ID is incremented by one each time it is
represented in a CDR record
%sipid
SIP call ID. Blank if no SIP interface was used during the call.
%ocgnum Original calling number. Calling number as received by the unit.

%cgnum
Calling number. Calling number after manipulation by the call router.
%ocdnum Original called number. Called number as received by the unit.
462
%cdnum
Called number. Called number after manipulation by the call router.
%oiname
Original Interface name. Interface on which the call was received. Ex. isdn-Slot2/
Pri1.
%diname
Destination interface name. Interface on which the call was relayed. Ex. SIPDefault
%chan
Channel number. Blank if no PRI/BRI interface was used during the call. If 2 PRI/
BRI interface were involved, display the originating interface.
%sipla
SIP local IP address.
%sipra
SIP remote IP address or FQDN (next hop).
%siprp
SIP remote port (next hop).
%mra
Media remote IP address. Source IP address of incoming media stream. If the

stream was modified during the call, display the last stream.
%mrsp
Media remote port. Source port of incoming media stream. If the stream was
modified during the call, display the last stream.
%mdrp
Media remote port. Destination port of outgoing media stream. If the stream was
modified during the call, display the last stream.
%tz
Local time zone
%cd
Call duration (in seconds) (connect/disconnect).
%sd
Call duration (in seconds) (setup/connect).
%pdd
Post dial delay (in seconds) (setup/progress).
%css
Call setup second (local time)
%csm
Call setup minute (local time)
%csh
Call setup hour (local time)
%csd
Call setup day (local time)
%csmm
Call setup month (local time)
%csy
Call setup year (local time)
%ccs
Call connect second (local time)
%ccm
Call connect minute (local time)
%cch
Call connect hour (local time)
%ccd
Call connect day (local time)
%ccmm
Call connect month (local time)
%ccy
Call connect year (local time)
Table 399: Call Detail Record Macros (Continued)

Macro
3.
Value
%cds
Call disconnect second (local time)
%cdm
Call disconnect minute (local time)
%cdh
Call disconnect hour (local time)
%cdd
Call disconnect day (local time)
%cdmm
Call disconnect month (local time)
%cdy
Call disconnect year (local time)
%miptxc
IP Media last transmitted codec
%miptxp
IP Media last transmitted p-time
%dr
Disconnect reason (ISDN reason codes with ISUP SIP mapping)
%rxp
Received media packets. Excluding T.38.
%txp
Transmitted media packets. Excluding T.38.
%rxpl
Received media packets lost. Excluding T.38.
%rxmd
Received packets mean playout delay (ms, 2 decimals). Excluding T.38.
%rxaj
Received packets average jitter (ms, 2 decimals). Excluding T.38.
%sipdr
SIP disconnect or rejection reason.
Set the Syslog facility used by the unit to route the Call Detail Record messages in the Syslog
Facility field.
The application can use Local0 through Local7.
4.
463
Chapter 41 - Call Detail Record
464
H A P T E R
42
Call Router Configuration

This chapter describes the call router service.
Introduction to the call routers parts and types supported.

Routes parameters.
Mappings parameters.
Call signalling parameters.
SIP headers translation parameters.
Call properties translation parameters.
Hunt table parameters.
SIP Redirects parameters.
Standards Supported
ITU-T Recommendation E.164: The international public

telecommunication numbering plan.
ITU-T Recommendation F.69: List of Telex Destination

Codes.
ITU-T Recommendation X.121: International numbering plan

for public data networks.
Introduction
The Mediatrix units call router allows you to route calls between interfaces. Based on a set of routing criteria,
the call router determines the destination (interface) for every incoming call. The forwarding decisions are
based on the following tables:
Table 400: Call Router Table Types
Table
Description
Routing
The routing table contains one or more routes. Each route associates a destination to a call
that matches a set of criteria. See Routes on page 485 for more details.
Mapping
The mapping table contains one or more mapping types and expressions. A mapping
modifies call properties such as the calling and called party numbers according to the network
requirements. These mappings are specifically called within a route. See Mappings on
Call
Signalling
Call signalling specifies how to set up a call to the destination Mediatrix unit or 3rd party
equipment. Call signalling properties are assigned to a route and used to modify the
behaviour of the call at the SIP protocol level. See Signalling Properties on page 501 for
more details.
SIP
A SIP headers translation overrides the default value of SIP headers in an outgoing SIP
Headers
message. See SIP Headers Translations on page 505 for more details.
Translation
Call
A call properties translation overrides the default value of call properties in an incoming SIP
Properties message. See Call Properties Translations on page 508 for more details.
Translation
467
Chapter 42 - Call Router Configuration
Introduction
Table 400: Call Router Table Types (Continued)

Table
Description
Hunt
The hunt table contains one or more hunt entries, each with a set of possible destinations. A
hunt tries the destinations until one of the configured destinations accepts the call. See Hunt
Service on page 511 for more details.
SIP
Redirects
The SIP Redirects table allows configuring of SIP redirections that can be used as Route
destinations. When the Route source is a SIP interface, incoming SIP Invites are replied with
a 302 Moved Temporarily SIP response. See SIP Redirects on page 519 for more details.
When a new call comes from one of the Mediatrix unit interfaces, it is redirected to the routing table. The
following figure illustrates the Mediatrix unit call router:
Figure 217: Call Routing
Interfaces
Call Router
Route 1
Signalling
Properties 1
Route 2
Route 3
Mapping 1
Hunt 1
Translation 1
(SIP Headers or
call properties)
Limitations
The call routing service has the following limitations:
468
A call coming from a SIP interface cannot be routed to another SIP interface. When that occurs,
the call automatically fails.
A call automatically fails if it is redirected to a route or hunt more than 10 times.
Maximum number of Routes: 40
The call properties Called Bearer Channel and Calling Bearer Channel are limited to ISDN
interfaces only.
Maximum number of Mapping Types: 40
Maximum number of Mapping Expressions: 100
Maximum number of Hunts: 40
Maximum number of Signaling Properties: 40
Maximum number of SIP Header Translations: 100
Maxium number of Call Properties Translations: 100
Introduction
Regular Expressions
Standards Supported
IEEE Std 1003.1-2001: IEEE Standard for Information

Technology---Portable Operating System Interface (POSIX)
Some of the routing types described in Routing Type on page 470 require that you enter them following the
regular expression syntax. A regular expression is a string used to find and replace strings in other large
strings. The Mediatrix unit uses regular expressions to enter a value in several routing types, often by using
wildcard characters. These characters provide additional flexibility in designing call routing and decrease the
need for multiple entries in configuring number ranges.
The expression cannot begin by ^, it is implicit in the expression. The following table shows some of the
wildcard characters that are supported:
Table 401: Regular Expressions Wildcards
Character
Description
Single-digit place holder. For instance, 555 .... matches any dialed number beginning with
555, plus at least four additional digits. Note that the number may be longer and still match.
Repeats the previous digit 0, 1, or more times. For instance, in the pattern:
1888*1
the pattern matches:

1881, 18881, 188881, 1888881
Note: If you are trying to handle the asterisk (*) as part of a dialed number, you must use \*.
[]
Range of digits.
A consecutive range is indicated with a hyphen (-), for instance, [5-7].
A nonconsecutive range is indicated without a delimiter, for instance, [58].
Both can be used in combination, for instance [5-79], which is the same as
[5679].
You may place a (^) symbol right after the opening bracket to indicate that the specified range
is an exclude list. For instance, [^01] specifies the same range as [2-9].
Note: The call router only supports single-digit ranges. You cannot specify the range of
numbers between 99 and 102 by using [99-102].
()
Indicates a pattern (also called group), for instance, 555(2525). It is used when replacing a
number in a mapping. See Groups on page 469 for more details.
Matches 0 or 1 occurrence of the previous item. For instance, 123?4 matches both 124 and
1234.
Repeats the previous digit one or more time. For instance 12+345 matches 12345, 122345,
etc. (but not 1345). If you use the + at the end of a number, it repeats the last number one or
more times. For instance: 12345+ matches, 12345, 123455, 1234555, etc.
Indicates a choice of matching expressions (OR).
The matching criterion implicitly matches from the beginning of the string, but not necessarily up to the end.
For instance, 123 will match the criterion 1, but it will not match the criterion 2.
If you want to match the whole string, you must end the criterion with $. For instance, 123 will not match the
criterion 1$ and will match the criterion 123$.
Note: You can use the <undefined> string if you want to match a property that is not defined.
You can also use the macro local_ip_port to replace the properties by the local IP address and port of the
listening network of the SIP gateway used to send the INVITE.
Groups
469
Introduction
A group is placed within parenthesis. It is used when replacing a string in a mapping. You can use up to nine
groups (defined by \1 to \9) and matching is not case sensitive. \0 represents the whole string. Lets say
for instance you have the following string:
9(123(45)6)
The following describes how the groups are replaced in a properties manipulation:
Table 402: Groups Replacement Example
Replacement
Result
\0
9123456
\1
123456
\2
45
\3
Groups can only be used with the following routing types:
Calling/Called E.164
Calling/Called Name
Calling/Called Host
Calling/Called URI
Routing Type
Standards Supported

The following sub-sections list the available routing types of the call router and their supported values. The
routing types that offer choices use the choices as defined in the Q.931 standard. Q.931 is ISDNs connection
control protocol, roughly comparable to TCP in the Internet protocol stack. The values may also be a special
tag, as described in Special Tags on page 476.
Table 403: Routing Types Locations
Routing Type
Location
E164
Called / Calling E164 on page 471
Type of Number (TON)
Called / Calling TON on page 471
Numbering Plan Indicator (NPI)
Called / Calling NPI on page 471
Name
Called / Calling Name on page 472
Host
Called / Calling Host on page 472
URI
Called / Calling URI on page 472
Presentation Indicator (PI)
Calling PI on page 472
Screening Indicator (SI)
Calling SI on page 472
Information Transfer Capability (ITC) Calling ITC on page 473
470
Date and Time
Date/Time on page 473
Phone Context
Called / Calling Phone Context on page 474
SIP Username
Called / Calling SIP Username on page 474
Bearer Channel
Called / Calling SIP Username on page 474
Diverting Reason
Last / Original Diverting Reason on page 474
Diverting E.164
Last / Original Diverting E.164 on page 475
Introduction
Table 403: Routing Types Locations (Continued)

Routing Type
Location
Diverting Party Number Type
Last / Original Diverting Party Number Type on page 475
Diverting Public Type Of Number
Last / Original Diverting Public Type Of Number on page 475
Diverting Pivate Type Of Number
Last / Original Diverting Private Type Of Number on page 475
Diverting Number Presentation
Last / Original Diverting Number Presentation on page 476
SIP Privacy Type
SIP Privacy Type on page 476
Media5 recommends to carefully define the routing requirements and restrictions that apply to your installation
before starting the routing configuration. This will help you determine the types of routing you need. When this
is done, define the routes and mappings, as well as the hunts that you need to fulfil these requirements. You
may need several entries of the same type to achieve your goals.
See also Call Properties Parameters on page 476 for a description of the parameters used by the various
routing types and interfaces of the call router.
Called / Calling E164

This is the Called/Calling Party Number. You can enter a regular expression (called/calling party E.164
number in the call setup message) as per Regular Expressions on page 469. Note that:
A PBX may insert or modify the calling party number. Sometimes there is no calling party
number at all. This all depends on the equipment you connect to the device.
The Mediatrix unit cannot filter the redirecting number information element of the SETUP
message because it does not support the calling-Redir-E164 and Calling-Redir-Reason
routing properties criteria.
Called / Calling TON

Called or calling party type of number field in the ISDN setup message. The following values are available:
Table 404: Type of Number Values
Value
unknown
Description
Unknown number type.
international International number.

national
National number.
network
Network specific number used to indicate an administration or service number specific to the
serving network.
subscriber
Subscriber number.
abbreviated
Abbreviated number.
Note: The called type of number is set to international if the To username is an E.164 with the prefix +.
The calling type of number is set to international if the From username is an E.164 with the prefix +.
Called / Calling NPI

Called or calling party numbering plan indicator field in the ISDN setup message. The following values are
471
Introduction
available:
Table 405: Numbering Plan Indicator Values
Value
Description
unknown
Unknown numbering plan.
isdn (E.164)
ISDN/Telephony numbering plan according to ITU-T Recommendation E.164.
data (X.121) Data numbering plan according to ITU-T Recommendation X.121.

telex (F.69)
Telex numbering plan according to ITU-T Recommendation F.69.
national
Numbering plan according to a national standard.
private
A private numbering plan.
Called / Calling Name

Calling and called party name (display name). This is the human-readable name of the calling or called party.
See Regular Expressions on page 469 for more details on how to enter a proper expression.
The Mediatrix unit does not support the sending of the calling name in the user-to-user information element.
Called / Calling Host

IP address or domain name of the called or calling host in the following format:
Fqdn[:port]
If [:port] is missing, the call router uses the well-known port of the signalling protocol. Note that:
Incoming SIP calls use the calling party IP address property to store the IP address of the
remote SIP user agent. Other interfaces such as ISDN set the IP address to 0.0.0.0.
You can use a regular expression to enter an IP address or a range of IP addresses.
Called / Calling URI

Uniform Resource Identifier (URI) of:
the called party, e.g., the To-URI.

the originating VoIP peer, e.g., the From-URI of an incoming SIP call.
The URI follows the format described in RFC 3261.
Calling PI
Presentation indicator of the calling party number. The following values are available:
Table 406: Presentation Indicator Values
Value
Description
allowed
Presentation of the calling party number is allowed.
restricted
Presentation of the calling party number is restricted.
interworking The calling party number is not available due to interworking.

You may want to remove the calling party number when the user sets the presentation indicator to restricted.
To achieve this, route restricted calls to a mapping that sets the Calling E164 to an empty string.
Calling SI
472
Introduction
Screening indicator of the calling party number. The following values are available:
Table 407: Screening Indicator Values
Value
Description
notscreened
The user provides the calling party number but the number is not screened by the network.
Thus the calling party possibly sends a number that it does not own.
passed
The calling party number is provided by the user and it passes screening.
failed
The calling party number is set by the user and verification of the number failed.
network
The originating network provides the number in the calling party number parameter.
You may want to remove the calling party number when it is not screened or screening failed. To do so, route
these calls to a mapping that sets the Calling E164 to an empty string. If you want to drop calls when the calling
party number is not screened or screening failed, use the Calling Si as criteria for the route.
Calling ITC
The information transfer capability field of the bearer capability information element in the ISDN setup
message. The following values are available:
Table 408: Information Transfer Capability Values
Value
Description
speech
Voice terminals (telephones).
unrestricted
Unrestricted digital information (64 kbps).
restricted
Restricted digital information (64 kbps).
3.1Khz
Transparent 3.1 kHz audio channel.
udi-ta
Unrestricted digital information with tones/announcements.
Note: This was formerly transparent 7.1 kHz audio channel.

video
Video conference terminals.
The Mediatrix unit currently supports the following Information Transfer Capabilities when receiving calls to
and from the ISDN (named as in Q.931, 05/98):
Speech
Unrestricted Digital Information
3.1 kHz Audio
Those are respectively referenced as Speech, Unrestricted and 3.1 kHz in the call routing configuration.
When initiating calls towards the ISDN, the Mediatrix unit uses the calling ITC value if it is one of the three
listed above. If none is set, it uses 3.1 kHz Audio. If the calling ITC set by the call router is different from the
three listed above, the call is rejected.
Note: Terminals connected to analog extensions (e.g. of a PBX) do not supply information transfer
capability values in their call setup. The configuration of the analog port on the Terminal Adapter, NT or PBX
is thus responsible to insert this value. The configuration of this value is however often omitted or wrong.
The ITC value may therefore not be a reliable indication to differentiate between analogue speech, audio or
Fax Group 3 connections. Furthermore, calls from SIP interfaces do not differentiate between bearer
capabilities. They always set the information transfer capability property to 3.1Khz.
Date/Time
473
Introduction
Day of week and time period and/or date and time period. The following are the accepted formats:
Table 409: Date/Time Accepted Formats
Format
Description
Date/Time Period format
Week Day/Time Period format
'DD.MM.YYYY/HH:MM:SS-DD.MM.YYYY/
HH:MM:SS'
'DD.MM.YYYY/HH:MM:SS-HH:MM:SS'
'DD.MM.YYYY-DD.MM.YYYY'
'DD.MM.YYYY'
'HH:MM:SS-HH:MM:SS'
'DDD'
'DDD,DDD...'
'DDD/HH:MM:SS-HH:MM:SS'
'DDD,DDD.../HH:MM:SS-HH:MM:SS'
DDD must be one of: SUN, MON, TUE, WED, THU, FRI,
SAT.
Many of the formats above can be concatenated to form one expression. They must be separated by |. For
instance: 25.12.2006 | SUN.
Called / Calling Phone Context

This is a user parameter in a URI. For instance:
sip:1234;phone-context=1234@domain.com;user=phone
You can enter a regular expression (called/calling party phone context in the call setup message) as per
Regular Expressions on page 469.
Called / Calling SIP Username

Calling and called party SIP username. See Regular Expressions on page 469 for more details on how to
enter a proper expression.
Called / Calling Bearer Channel

Calling and called party bearer channel. See Regular Expressions on page 469 for more details on how to
enter a proper expression.
Last / Original Diverting Reason

Standards Supported
RFC 5806: Diversion Indication in SIP
This is the last or original diverting reason in ISDN setup and SIP INVITE messages. The following values are
available:
Table 410: Diverting Reason Values
Value
474
Description
cfb
Call Forward on Busy Allowed.
cfu
Call Forward on Unavailable Restricted
cfnr
Call Forward on No Answer Interworking
unknown
unknown
Introduction
Refer to You can set the SIP transfer method when an endpoint is acting as the transferor in a blind transfer
scenario. on page 350 to select the SIP method used to receive/send call diversion information in an INVITE.
Last / Original Diverting E.164

Last or original party number to which the call was being routed when the first diversion occurred. You can
enter a regular expression (called/calling party E.164 number in the call setup message) as per Regular
Expressions on page 469. Note that:
A PBX may insert or modify the calling party number. Sometimes there is no calling party
number at all. This all depends on the equipment you connect to the device.
The Mediatrix unit cannot filter the redirecting number information element of the SETUP
message because it does not support the calling-Redir-E164 and Calling-Redir-Reason
routing properties criteria.
Last / Original Diverting Party Number Type

Table 411: Diverting Party Number Type Values
Value
Description
unknown
public
Public number.
private
Private number.
Last / Original Diverting Public Type Of Number

Diverting or original called number public type of number field in the ISDN Setup message. Used only when
the diverting or original called number type of number is 'public'. The following values are available:
Table 412: Diverting Public Type of Number Values
Value
unknown
Description
international International number.

national
National number.
networkspecific
Network specific number used to indicate an administration or service number specific to the
serving network.
subscriber
Subscriber number.
abbreviated
Abbreviated number.
Last / Original Diverting Private Type Of Number

Diverting or original called number private type of number field in the ISDN Setup message. Used when the
diverting or original called party number type is 'private'. The following values are available:
Table 413: Diverting Private Type of Number Values
Value
Description
unknown
Unknown.
leg2-reg
Leg2 reg.
leg1-reg
Leg1 reg.
475
Introduction
Table 413: Diverting Private Type of Number Values (Continued)

Value
Description
pisnspecific
PISN Specific.
subscriber
Subscriber number.
abbreviated
Abbreviated number.
Last / Original Diverting Number Presentation

Diverting or original called number presentation. The following values are available:
Table 414: Diverting Presentation Values
Value
Description
allowed
Presentation of the party number is allowed.
restricted
Presentation of the party number is restricted.
interworking The party number is not available due to interworking.

restrictedaddress
Restricted address.
SIP Privacy Type

Calling SIP privacy level of the call. The following values are available:
Table 415: SIP Privacy Values
Value
Description
disabled
No privacy is used.
none
Use P-Asserted Identity privacy.
id
Use P-Preferred Identity privacy.
Special Tags
You can use the following special tags as routing types values.
Table 416: Special Tags
Tag
Description
undefined
Matches if the property is not defined for the call.
default
Always matches. Generally used to set a default route if the previous criteria do not match.
Call Properties Parameters

The following sections describe the parameters used by the various call properties (routing types) and
interfaces of the call router.
Call Properties to SIP
476
Introduction
This section describes the information the call router uses for the various SIP fields.
Table 417: Call Properties to SIP
SIP Field
To
From
Description
The Mediatrix unit uses the calling URI to populate the To field if not undefined.
Otherwise, the unit does the following:
Uses the called Name for the friendly name if not undefined.
Uses the called SipUsername for the user name if not empty or
undefined; otherwise, uses the called E164 for the username. If it is
empty or undefined, the Mediatrix unit rather uses the value defined
in the Default Username Value field of the SIP > Interop > SIP
Interop parameters as username (see SIP Interop on page 316 for
more details). The unit uses the called Phone Context for the user's
'phone-context' parameter if not empty. If a 'phone-context'
parameter is added, the URI parameter 'user' is also automatically
added. Its value is defined in the SIP URI User Parameter Value
field of the SIP > Interop > SIP Interop parameters. If empty, then
the value 'phone' is used
Uses the called Host for the host if not undefined, otherwise uses the
configured home domain proxy host.
Prefixes the user name with + and adds the URI parameter user
with the value phone if the called TON is international.
If there is no URI parameter user yet and the SIP URI User
Parameter Value field of the SIP > Interop > SIP Interop parameters
is not empty, then the parameter is added with the value defined by
the field.
The Mediatrix unit uses the called URI to populate the From field if not
undefined. Otherwise, the unit does the following:
Uses the calling Name for the friendly name if not undefined.
Uses the calling SipUsername for the user name if not empty or
undefined; otherwise, uses the calling E164 for the username. If it is
empty or undefined, the Mediatrix unit rather uses the value defined
in the Default Username Value field of the SIP > Interop > SIP
Interop parameters as username (see SIP Interop on page 316 for
more details).The unit uses the calling Phone Context for the user's
'phone-context' parameter if not empty. If a 'phone-context'
parameter is added, the URI parameter 'user' is also automatically
added. Its value is defined in the SIP URI User Parameter Value
field of the SIP > Interop > SIP Interop parameters. If empty, then
the value 'phone' is used.
Uses the calling Host for the host if not undefined, otherwise uses
the configured home domain proxy host.
Prefixes the user name with + and adds the URI parameter user
with the value phone if the calling TON is international.
If there is no URI parameter user yet and the SIP URI User
Parameter Value field of the SIP > Interop > SIP Interop parameters
is not empty, then the parameter is added with the value defined by
the field.
Request URI
The Mediatrix unit uses the same information as the To field.
Contact
The Mediatrix unit uses the same information as the From field, but with the
current IP address/port for the host.
477
Introduction
Table 417: Call Properties to SIP (Continued)

SIP Field
Diversion
Description
A Diversion header is added if the Last Diverting E.164 property is present and
not empty. This Diversion header is constructed as follows:
The username of the URI is set to the value of the Last Diverting
E.164 property.
The host of the URI is set to the configured home domain proxy host.
The reason field is set according to value of the Last Diverting

Reason property:
cfu: "unconditional"
cfb: "user-busy"
cfnr: "no-answer"
All other values or when undefined: "unknown'.
The field counter is set to the value of DivertingCounter if the

Original Diverting E.164 property is set to empty or undefined,
otherwise it is set to DivertingCounter -1.
A second Diversion header is added if the Last Diverting E.164 and Original
Diverting E.164 properties are present and not empty. This Diversion header is
constructed as follows:
The username of the URI is set to the value of the Original Diverting
E.164 property.
The host of the URI is set to the configured home domain proxy host.
The reason field is set according to the value of the Original

Diverting Reason property:
cfu: "unconditional"
cfb: "user-busy"
cfnr: "no-answer"
All other values or when undefined: "unknown'.
The field counter is set to 1.
SIP to Call Properties

This section describes the SIP information the call router uses for the various call properties.
Table 418: SIP to Call Properties
Property
478
SIP Information
Called URI
The URL of the To field.
Calling URI
The URL of the From field.
Called Name
The friendly name in the To field. The property is undefined if there is no

friendly name.
Calling Name
The friendly name in the From field. The property is undefined if there is no
friendly name.
Called E164
The user name of the Request-Uri field if the user name is a compatible E.164.
The prefix + and separator - are removed. The property is undefined if there
is no user name or if it is not compatible.
Calling E164
The user name of the From field if the user name is a compatible E.164. The
prefix + and separator - are removed. The property is undefined if there is no
user name or if it is not compatible.
Called Host
The host of the To field.

Introduction
Table 418: SIP to Call Properties (Continued)

Property
SIP Information
Calling Host
The host of the Contact field.
Called TON
Set to international if the To user name is an E.164 with the prefix +;

otherwise, the property is undefined.
Calling TON
Set to international if the From user name is an E.164 with the prefix +;
otherwise the property is undefined.
Called Phone Context
Set to the parameter phone-context of the user name of the To if the user
name is an E.164, otherwise the property is undefined.
Calling Phone Context
Set to the parameter phone-context of the user name of the From if the user
name is an E.164, otherwise the property is undefined.
Called SIP Username
Set to the username of the Request-Uri. Note that this does not include the
username parameter like the phone-context.
Calling SIP Username
Set to the username of the From. Note that this does not include the username
parameter like the phone-context.
Last Diverting Reason
If the INVITE contains at least one Diversion header, this value is set according
to the reason field value of the first Diversion header:
"user-busy": cfb
"unconditional":cfu
"no-answer": cfna
All other values: unknown
Otherwise, the property is undefined.

The reason field comparison is not case sensitive.
Original Diverting
Reason
If the INVITE contains more than one Diversion header, this value is set
according to the reason field value of the last Diversion header:
"user-busy": cfb
"unconditional":cfu
"no-answer": cfna
All other values: unknown

The reason field comparison is not case sensitive.
Last Diverting E.164
If the INVITE contains at least one Diversion header, this value is set to the
username of the URI (can be a SIP URI, SIPS URI or TEL URI) of the first
Diversion header converted into an E.164. It can be set to empty if there is no
username or if the username is not an E.164.
Original Diverting E.164
If the INVITE contains more than one Diversion header, this value is set to the
username of the URI (can be a SIP URI, SIPS URI or TEL URI) of the last
Diversion header converted into an E.164. It can be set to empty if there is no
username or if the username is not an E.164.
Diverting Counter
If the INVITE contains at least one Diversion header, this value is set to the
sum of the counter field of all Diversion headers. If a diversion header does not
contain the counter field, the value 1 is assumed for the header.
All others
The property is undefined.
Call Properties to ISDN

479
Introduction
This section describes the information the call router uses for the various ISDN information elements.
Table 419: Call Properties to ISDN
Information Element
Bearer Capabilities
Calling Party Number
Description
If valid, the calling ITC is used to fill the information transfer capability (octet 3
[5:1]). Otherwise, the ITC is set to 3.1 kHz audio. If more than one bearer
capability information elements is provided in a prioritized list, they all receive
the same ITC. This information element is included in the SETUP message
only for outgoing calls.
Uses the calling E164 to fill the field number digits (octet 4).
Uses the calling TON to fill the field type of number (octet 3 [7:5]).
Uses the calling PI to fill the field presentation indicator (octet 3a [7:6).
Uses the calling SI to fill the field screening indicator (octet 3a [2:1]).
Uses the calling NPI to fill the field numbering plan identification (octet 3
[4:1]).
Called Party Number
Uses the called E164 to fill the field number digits (octet 4).
Uses the called TON to fill the field type of number (octet 3 [7:5]).
Uses the called NPI to fill the field numbering plan identification (octet 3 [4:1]).
Display
Uses the calling E164 to fill the field display information (octet 3).
Called Bearer Channel
The called bearer channel is used to select a specific ISDN bearer channel for
an outgoing ISDN call.
ISDN to Call Properties

This section describes the ISDN information the call router uses for the various call properties.
Table 420: ISDN to Call Properties
Property
480
ISDN Information
Calling Name
Field display information (octet 3) of the Display information element, if

included in the SETUP Q.931 message.
Called E164
Field number digits (octet 4) of the called party information element included
in the SETUP Q.931 message.
Calling E164
Field number digits (octet 4) of the calling party information element included
in the SETUP Q.931 message.
Called TON
Field type of number (octet 3 [7:5]) of the called party information element
Calling TON
Field type of number (octet 3 [7:5]) of the calling party information element
Calling PI
Field presentation indicator (octet 3a [7:6) of the calling party information

element included in the SETUP Q.931 message.
Calling SI
Field screening indicator (octet 3a [2:1]) of the calling party information

element included in the SETUP Q.931 message.
Calling ITC
Field information transfer capability (octet 3 [5:1]) of the bearer capability

information element included in the SETUP Q.931 message.
Called NPI
Field numbering plan identification (octet 3 [4:1]) of the called party

Calling NPI
Field numbering plan identification (octet 3 [4:1]) of the calling party

Introduction
Table 420: ISDN to Call Properties (Continued)

Property
ISDN Information
Calling Bearer Channel
Represents the ISDN bearer channel on which the ISDN call is received.
All others
The property is undefined.
Call Properties to FXS

This section describes the information the call router uses for the various call properties to FXS.
Table 421: Call Properties to FXS
Caller ID
Description
Number
If the PI property is present and not set to "allowed", the number is "P".
Otherwise, the number is set to the value of the E164 property (truncated to the
first 20 characters). See Auto-Routing on page 523 for details.
Name
If the PI property is present and not set to "allowed", the name is "Anonymous".
Otherwise, the name is set to the value of the Name property (truncated to the
first 50 characters). See Auto-Routing on page 523 for details.
FXS to Call Properties

This section describes the information the call router uses for the various FXS to call properties.
Table 422: FXS to Call Properties
Caller ID
Description
Calling E164
If the auto routing is enabled and the E164 field of the Call Router > Autorouting page is not empty (see Auto-Routing on page 523 for details), the
value of the E164 field. Otherwise, the property is not present.
Calling Name
If the auto routing is enabled and the Name field of the Call Router > Autorouting page is not empty (see Auto-Routing on page 523 for details), the
value of the Name field. Otherwise, the property is not present.
If the auto routing is enabled and the SIP Username field of the Call Router >
Auto-routing page is not empty (see Auto-Routing on page 523 for details),
the value of the SIP Username field. Otherwise, the property is not present.
Called E164
For automatic calls, the E.164 defined in the Automatic Call Target field of the
Telephony > Services page (see Automatic Call on page 423 for more
details).
For other calls, the dialed digit after the transformation defined in the
Transformation field of the Allowed DTMF Map section (Telephony > DTMF
Maps page see Allowed DTMF Maps on page 409 for more details).
Called Name
For automatic calls, the name specified in the Automatic Call Target field of the
details). The property is not present if the target address does not contain a
name.
For other calls, the property is not present.
481
Introduction
Table 422: FXS to Call Properties (Continued)

Caller ID
Called Host
Description
For automatic calls, the host specified in the the Automatic Call Target field of
the Telephony > Services page (see Automatic Call on page 423 for more
host.
For other calls, the host defined in the Target field of the Allowed DTMF Map
section (Telephony > DTMF Maps page see Allowed DTMF Maps on
page 409 for more details). The property is not present if the target host is not
configured for the matching DTMF map.
Call Properties to FXO

This section describes the information the call router uses for the various call properties to FXO.
Table 423: Call Properties to FXO
Caller ID
Dialled number
Description
The Called E164 property.
FXO to Call Properties

This section describes the information the call router uses for the various FXO to call properties.
Table 424: FXO to Call Properties
Caller ID
Calling E164
Description
If the caller ID is detected, the numbers provided by the caller ID.
If the auto routing is enabled and the E164 field of the Call Router > Autorouting page is not empty (see Auto-Routing on page 523 for details), the
value of the E164 field. Otherwise, the property is not present.
Calling Name
If the caller ID is detected, the name provided by the caller ID.

If the auto routing is enabled and the Name field of the Call Router > Autorouting page is not empty (see Auto-Routing on page 523 for details), the
value of the Name field. Otherwise, the property is not present.
If the caller ID is detected, the property is not present.

If the auto routing is enabled and the SIP Username field of the Call Router >
Auto-routing page is not empty (see Auto-Routing on page 523 for details),
the value of the SIP Username field. Otherwise, the property is not present.
Called E164
For automatic calls, the E.164 defined in the Automatic Call Target field of the
details).
For other calls, the dialed digit after the transformation defined in the
Transformation field of the Allowed DTMF Map section (Telephony > DTMF
Maps page see Allowed DTMF Maps on page 409 for more details).
Called Name
For automatic calls, the name specified in the Automatic Call Target field of the
name.
For other calls, the property is not present.
482
Introduction
Table 424: FXO to Call Properties (Continued)

Caller ID
Called Host
Description
For automatic calls, the host specified in the the Automatic Call Target field of
the Telephony > Services page (see Automatic Call on page 423 for more
host.
For other calls, the host defined in the Target field of the Allowed DTMF Map
section (Telephony > DTMF Maps page see Allowed DTMF Maps on
page 409 for more details). The property is not present if the target host is not
configured for the matching DTMF map.
SIP/ISDN Call Default Values

When performing a call from SIP to ISDN or ISDN to SIP, some ISDN informations are missing from the SIP
packet. The Dgw v2.0 Application sets the following default values when the information is missing. You
cannot filter on these default values, but you can filter with the <undefined> or <default> values.
Table 425: SIP/ISDN Calls Default Values
Parameter
Default Value
SIP to ISDN Calls
TON (calling)
unknown
TON (called)
unknown
NPI (calling and called) unknown

SI (calling)
User-side: not-screened
Network-side: network
ITC (calling)
PI (calling)
3.1 kHz audio

1. When the Calling Party Number E.164 is missing: interworking. In this
case, this value overrides any value set by the call router.
2. When CLIR is enabled (user-side only): restricted. In this case, this value
overrides any value set by the call router.
3. All other cases: allowed. This is the default value if the two cases above
do not apply and no value has been set by the call router.
ISDN to SIP Calls
SI (calling)
Network-side: The SI in the incoming Calling Party information element is ignored

and replaced by one of the following:
1. No calling IA5 digits received: network.
2. NPI is not unknown nor ISDN telephony: network.
3. TON is not international nor national: network, called IA5 digits are
discarded.
4. PI is set to interworking: network.
5. Otherwise: passed.
User-side: not-screened.
483
Introduction
Table 425: SIP/ISDN Calls Default Values (Continued)

Parameter
PI (calling)
Default Value
Network-side:
1. CLIR enabled: restricted. The PI is set to restricted no matter if a PI is
present in the incoming Calling Party IE.
2. CLIR disabled, no IA5 digits provided: interworking.
3. CLIR disabled, IA5 digits provided: allowed.
User-side:
1. CLIR disabled, no IA5 digits provided: interworking.
2. CLIR disabled, IA5 digits provided: allowed.
ITC (calling)
Must be provided in the incoming Bearer Capabilities information element

provided by the ISDN peer that initiated the call. There is no default value, the call
should be rejected if missing.
TON (called)
The Called TON must be provided by the ISDN peer that initiated the call.
TON (calling)
unknown
NPI (called
The Called NPI must be provided by the ISDN peer that initiated the call.
NPI (calling)
unknown
Note that the calling PI, SI, TON and NPI are present in Calling Party information elements in SETUP
messages sent by the network-side only when CLIP is enabled. They should always be present in messages
sent by the user-side. See Chapter 21 - ISDN Configuration on page 179 for more details on CLIP.
Call Routing Status

The routes, mappings, and hunts currently in use, as well as the available interfaces, are displayed in the Call
Router > Status page.
484
Routes
Figure 218: Call Router Status Web Page
Routes
The routing table contains one or more routes. These routes forward an incoming or outgoing call to another
route, interface, or hunt based on a specific call property such as the called party number. It may also use a
mapping to modify the call setup message of a call and a signalling property to modify the behaviour of the
call at the SIP protocol level.
Once the call router finds a route that matches, it does not check the other routes, even if some of them may
still match. The routes sequence is thus very important. The call router follows the routing table rows (routes)
as they are entered in the web interface. If you want the call router to try to match one row before another one,
you must put that row first.
When a call arrives, the call router proceeds as follows:
1.
It examines the call property as specified with the routes.

To select a route, the call must match all three of the Source, Properties Criteria, and Expression
Criteria parameters.
2.
It selects the first matching route in the list of routes.
485
3.
Routes
It routes the call to the specified destination interface, hunt, or route.
Note: You can revert back to the configuration displayed in the Call Router > Status web page at any time
by clicking the Rollback button at the bottom of the page. All modified settings in the Call Router > Route
Config page will be lost.
You can add up to 40 routes.
Creating/Editing a Route
The web interface allows you to create a route or modify the parameters of an existing one.
To create or edit a route:

1.
In the web interface, click the Call Router link, then the Route Config sub-link.
Figure 219: Call Router Route Config Web Page
2
3
2.
Locate the Route section.
3.
If you want to add a route before an existing entry, locate the proper row in the table
and click the
button of this row.
If you want to add a route at the end of the existing rows, click the
bottom right of the Route section.
button at the
If you want to edit an existing route, locate the proper row in the table and click the
button.
This brings you to the Configure Route panel.

Figure 220: Configure Route Panel
4
5
6
7
8
486
Routes
4.
Enter one or more sources to compare with the call and match in order to select the route in the
Source field.
A source may be:
route-name: The call uses the route name.
sip-name: The call comes from the SIP interface name.
isdn-name: The call comes from the ISDN interface name.
r2-name: The call destination is set to the R2 interface name.
e&m-name: The call comes from the E&M interface name.
fxs-name: The call destination is set to the FXS interface name.
fxo-name: The call destination is set to the FXO interface name.
If you want to use multiple sources, you must separate them by commas.
For instance, if you want to route calls that come from the SIP interface default, enter the following
value:
sip-default
If you want to route calls that come from the SIP interfaces default and other, enter the following
value:
sip-default,sip-other
Keep in mind that to select a route, the call must match all three of the Source, Properties Criteria,
and Expression Criteria parameters.
Note: When using endpoint gateways, SIP interface names are composed of both the gateway name and
a username; for example, a SIP source on an endpoint gateway may be: sip-default/5551212. When using
trunk gateways, SIP interface names are based on the gateway name only.
5.
Select a call property to compare with the call and match in order to select the route in the Properties
Criteria drop-down menu.
The call router offers several different routing types. Each type specifies which call property the call
router examines.
Table 426: Routing Types
Type
Description
Called E164
Routes calls based on the called party E.164 number.
Calling E164
Routes calls based on the calling party E.164 number.
Called TON
Routes calls based on the called party type of number.
Calling TON
Routes calls based on the calling party type of number.
Called NPI
Routes calls based on the called party numbering plan indicator.
Calling NPI
Routes calls based on the calling party numbering plan indicator.
Called Name
Routes calls based on the display name of the called party.
Calling Name Routes calls based on the display name of the calling party.
Called Host
Routes calls based on the signalling IP address or domain name.
Calling Host
Routes calls based on the signalling IP address or domain name.
Called URI
Routes calls based on the To-URI.
Calling URI
Routes calls based on the From-URI.
Calling PI
Routes calls based on the presentation indicator.
Calling SI
Routes calls based on the screening indicator.
487
Routes
Table 426: Routing Types (Continued)

Type
Description
Calling ITC
Routes calls based on the information transfer capability.
Date/Time
Routes calls based on the date and/or time the call arrived at the call router. A
link called Time criteria editor appears on the right of the Expression criteria
field. Use it to easily configure the Date/Time type.
Called Phone
Context
Routes calls based on the called party phone context.
Calling
Phone
Context
Routes calls based on the calling party phone context.
Called SIP
Username
Routes calls based on the called party SIP username.
Calling SIP
Username
Routes calls based on the calling SIP username.
Called
Bearer
Channel
Routes calls based on the called bearer channel properties.
Calling
Bearer
Channel
Routes calls based on the calling bearer channel properties.
Calling SIP
Privacy
Routes calls based on the calling SIP privacy properties.
6.
Enter the expression (related to the call properties selected in the previous step) to compare with
the call and match in order to select the route in the Expression Criteria field.
See Routing Type on page 470 for a list of available values for each call property.
For instance, if the property is Calling TON, you could instruct the call router to look for the following
expression:
international
If you have selected the Date/Time property in the above step, you can click the Time criteria
editor link and use the editor to easily configure the Date/Time parameters.
488
Routes
Figure 221: Date/Time Criteria Editor (Day Time)
Select between the Day-Time or Time-Period settings in the Select Criteria Type dropdown menu. If you select Time-Period, the editor changes as follows:
Figure 222: Date/Time Criteria Editor (Time Period)
Select or enter the parameters you want, then click the Add to List button. If a
parameter is invalid (for instance, the end date is inferior to the start date), it is
displayed in red in the Time Criteria List field.
To remove an existing parameter, select it in the Time Criteria List field, then click the
Remove Selected button.
To update an existing parameter, select it in the Time Criteria List field, then click the
Update Selected button.
To remove all parameters, click the Clear Parameters button.
When done, click the Submit button.
7.
If applicable, enter the name of mappings to apply to the call in the Mappings field.
You can enter more than one mapping by separating them with commas. These mappings are
executed in sequential order.
You can use the Suggestion columns drop-down menu to select an existing mapping, if any.
The manipulations are executed before sending the call to the new destination. See Mappings on
If you leave this field empty, no mapping is required.
8.
Select the call signalling property of the route used to modify the behaviour of the call at the SIP
protocol level in the Call Signaling drop-down menu.
489
Routes
You must set call signaling properties as defined in Signalling Properties on page 501. You can
use the Suggestion columns drop-down menu to select between existing properties, if any.
9.
Select the destination of the call when it matches in the Destination field.
The destination can be:
route-name: The call destination is set to the route name.
hunt-name: The call destination is set to the hunt name.
sip-name: The call destination is set to the SIP interface name.
isdn-name: The call destination is set to the ISDN interface name.
r2-name: The call destination is set to the R2 interface name.
e&m-name: The call destination is set to the E&M interface name.
fxs-name: The call destination is set to the FXS interface name.
fxo-name: The call destination is set to the FXO interface name.
SipRedirect-name: When the Route source is a SIP interface, incoming SIP Invites are
replied with a 302 'Moved Temporarily' SIP response. See SIP Redirects on page 519
or more details.
For instance, if you want to route calls to the hunt CallCenter, enter the following:
hunt-CallCenter
Note: When using endpoint gateways, SIP interface names are composed of both the gateway name and
a username; for example, a SIP source on an endpoint gateway may be: sip-default/5551212. When using
trunk gateways, SIP interface names are based on the gateway name only.
10.

This brings you back to the main Call Router > Route Config web page.
You can see a yellow Yes in the Config Modified section at the top of the window. It warns you that
the configuration has been modified but not applied (i.e., the Call Router > Status differs from the
Call Router > Route Config). The Route Config sub-menu is a working area where you build up a
Call Router configuration. While you work in this area, the configured parameters are saved but not
applied (i.e., they are not used to process incoming calls). The yellow Yes flag warns you that the
configuration has been modified but is not applied.
11.
Click the Apply button to enable the route.

The current routes applied are displayed in the Call Router > Status web page. You can also see
Examples
The following are some examples of routes:
Figure 223: Routes Examples
Moving a Route
Once the call router finds a routing entry that matches, it does not check the other entries, even if some of
them may still match. The routes sequence is thus very important. The call router follows the routing table rows
as they are entered in the web interface. If you want the call router to try to match one row before another one,
you must put that row first.
490
Mappings
To move a routing entry up or down:

1.
Either click the
or
2.
Click the Apply button to update the Call Router > Status web page.
Deleting a Route
You can delete a routing row from the table in the web interface.
To delete a routing entry:

1.
Click the
2.
Mappings
Mapping entries modify the call setup message of a call. They thus influence the routing decision and/or the
setup message leaving the call router. They are specifically called within a route.
Like the routing table, the mapping table finds the first matching entry. It then executes it by manipulating a
call property. A mapping always examines one call property and changes another property.
The call router executes all mapping entries that match by following the mapping table rows as they are
entered in the web interface. If you want the call router to try to match one row before another one, you must
put that row first.
The mapping may work with three types of call properties:
calling party properties

called party properties
generic properties
Generic properties are used for call properties that apply to both calling and called parties.
The web interface mapping configuration is separated in two parts: Mapping Type and Mapping Expression.
You must properly configure both parts for the mapping to work as required.
When a call arrives at the mapping table, the call router proceeds as follows:
1.
It examines the call property as specified in the Criteria (input) value of the Mapping Type part.
2.
It selects the first matching entry.
3.
It replaces the property specified in the Transformation (output) value of the Mapping Expression
part with the value of the selected entry.
Creating/Editing a Mapping Type

The Mapping Type part allows you to define the input call property to match and to define which call property
to change. The mapping type then uses one or more corresponding mapping expressions that you can define
in Creating/Editing a Mapping Expression on page 493.
You can add up to 40 Mapping Types.
491
Mappings
To create or edit a mapping type:

1.
2
3
2.
Locate the Mapping Type section.
3.
If you want to add a mapping type entry before an existing entry, locate the proper row
in the table and click the
button of this row.
If you want to add a mapping type entry at the end of the existing rows, click the
button at the bottom right of the Mapping Type section.
If you want to edit an existing entry, locate the proper row in the table and click the
button.
This brings you to the Configure Mapping Type panel.

Figure 225: Configure Mapping Type Panel
4
5
6
4.
Enter the name of the mapping in the Name field.

This is the name used in a route when calling a mapping. It must be unique. Media5 suggests to
use the type as part of the name for ease of identification.
There must be at least one corresponding mapping expression in the Mapping Expression table
with the exact same name. See Creating/Editing a Mapping Expression on page 493 for more
details.
5.
Select the input call property to compare with the call and match in order to select the mapping in
the Criteria drop-down menu.
6.
Select the call property to transform in the Transformation drop-down menu.
7.
492
Click the Submit button to go back to the main Call Router > Route Config web page.
You can now define a corresponding mapping expression.
Mappings
Click the Submit and Insert Expression button to directly access the proper mapping
expression dialog.
Creating/Editing a Mapping Expression

The Mapping Expression part defines the actual transformation to apply to the corresponding mapping type.
Each mapping expression must match a mapping type as defined in Creating/Editing a Mapping Type on
page 491.
You can add up to 100 Mapping Expressions.
To create or edit a mapping expression:

1.
2.
Locate the Mapping Expression section.
3.
If you want to add a mapping expression entry before an existing entry, locate the
If you want to add a mapping expression entry at the end of the existing rows, click the
proper row in the table and click the
button of this row.
button at the bottom right of the Mapping Expression section.
button.
This brings you to the Configure Mapping Expression panel.

Figure 227: Configure Mapping Expression Panel
4
5
6
7
4.
Enter the name of the mapping expression in the Name field.
493
Mappings
This name must match a mapping type as defined in Creating/Editing a Mapping Type on
page 491. You can use the Suggestion columns drop-down menu to select an existing mapping
type. When a name matches a mapping type, its type is displayed in the Type row as follows:
input type to output type
You can define several mapping expressions with the same name. In that case, the first row
matching the call is used. The rows are used in ascending order.
5.
Enter the expression (related to this specific input type) to compare with the call and match in order
to select the mapping in the Criteria field.
This string differs depending on the input type selected in the Mapping Type part (Criteria dropdown menu). For instance, if your input type is Calling TON, you could instruct the call router to look
for the following expression:
international
See Routing Type on page 470 for a list of available transformation values.
Table 427: Input Type Criteria
Input Type
Criteria
None
No criteria, always matches.
E164
If the Transformation value of the Mapping Type part is also a generic property,
this is applied to both this Calling E164 and Called E164 property.
Called E164
Selects an entry based on the called party E.164 number. You can use wildcards
to summarize entries as per Called / Calling E164 on page 471.
Calling E164
Selects an entry based on the calling party E.164 number. You can use
wildcards to summarize entries as per Called / Calling E164 on page 471.
Name
this is applied to both this Calling Name and Called Name property.
Called Name
Selects an entry based on the display name of the called party. You can use
wildcards to summarize entries as per Called / Calling Name on page 472.
Calling Name Selects an entry based on the display name of the calling party. You can use
wildcards to summarize entries as per Called / Calling Name on page 472.
494
TON
this is applied to both this Calling TON and Called TON property.
Called TON
Selects an entry based on the called party type of number as per Called /
Calling TON on page 471.
Calling TON
Selects an entry based on the calling party type of number as per Called /
Calling TON on page 471.
NPI
this is applied to both this Calling NPI and Called NPI property.
Called NPI
Selects an entry based on the called party numbering plan indicator as per
Called / Calling NPI on page 471.
Calling NPI
Selects an entry based on the calling party numbering plan indicator as per
Called / Calling NPI on page 471.
Host
this is applied to both this Calling Host and Called Host property.
Called Host
Selects an entry based on the remote signalling IP address or domain name of

the destination VoIP peer. You can use wildcards to summarize entries as per
Called / Calling Host on page 472.
Mappings
Table 427: Input Type Criteria (Continued)

Input Type
Criteria
Calling Host
Selects an entry based on the remote signalling IP address or domain name of

the originating VoIP peer. You can use wildcards to summarize entries as per
Calling PI
Selects an entry based on the presentation indicator as per Calling PI on

page 472.
Calling SI
Selects an entry based on the screening indicator as per Calling SI on

page 472.
Calling ITC
Selects an entry based on the information transfer capability as per Calling ITC
on page 473.
URI
this is applied to both this Calling URI and Called URI property.
Called URI
Selects an entry based on the called SIP URI properties. You can use wildcards
to summarize entries as per Called / Calling URI on page 472.
Calling URI
Selects an entry based on the calling SIP URI properties. You can use wildcards
to summarize entries as per Called / Calling URI on page 472.
Date/Time
Selects an entry based on the date and/or time the call arrived at the call router
as per Date/Time on page 473.
Phone
Context
Selects an entry based on the called or calling phone context properties as per
Called / Calling Phone Context on page 474.
Called Phone
Context
Selects an entry based on the called phone context properties as per Called /
Calling Phone Context on page 474.
Calling
Phone
Context
Selects an entry based on the calling phone context properties as per Called /
Calling Phone Context on page 474.
SIP
Username
Selects an entry based on the called or calling SIP username properties as per
Called / Calling SIP Username on page 474.
Called SIP
Username
Selects an entry based on the called SIP username properties as per Called /
Calling SIP Username on page 474.
Calling SIP
Username
Selects an entry based on the calling SIP username properties as per Called /
Last
Diverting
Reason
Selects an entry based on the last diverting reason properties as per Last /
Original Diverting Reason on page 474.
Last
Diverting
E164
Selects an entry based on the last diverting E.164 properties as per Last /
Original Diverting E.164 on page 475.
Last
Selects an entry based on the party number type of the last diverting number
Diverting
properties as per Last / Original Diverting Party Number Type on page 475.
Party
Number Type
Last
Diverting
Public Type
Of Number
Selects an entry based on the public type of number of the last diverting number
properties as per Last / Original Diverting Public Type Of Number on page 475.
495
Mappings
Table 427: Input Type Criteria (Continued)

Input Type
Criteria
Last
Diverting
Private Type
Of Number
Selects an entry based on the private type of numbekr of the last diverting
number properties as per Last / Original Diverting Private Type Of Number on
page 475.
Last
Diverting
Number
Presentation
Selects an entry based on the presentation of the last diverting number

properties as per Last / Original Diverting Number Presentation on page 476.
OriginalDiver
tingReason
Selects an entry based on the original diverting reason properties as per Last /
Original Diverting Reason on page 474.
OriginalDiver
tingE164
Selects an entry based on the original diverting E.164 properties as per Last /
Original Diverting E.164 on page 475.
Original
Selects an entry based on the party number type of the original diverting number
Diverting
properties as per Last / Original Diverting Party Number Type on page 475.
Party
Number Type
Original
Diverting
Public Type
Of Number
Selects an entry based on the public type of number of the original diverting
number properties as per Last / Original Diverting Public Type Of Number on
page 475.
Called
Bearer
Channel
Selects an entry based on the called bearer channel properties as per Called /
Calling
Bearer
Channel
Selects an entry based on the calling bearer channel properties as per Called /
Calling SIP
Privacyl
Selects an entry based on the calling SIP privacy properties as per SIP Privacy
Type on page 476.
If you are editing a Date/Time property, you can click the Time criteria editor link and use the editor
to easily configure the Date/Time parameters.
Figure 228: Date/Time Criteria Editor (Day Time)
496
Select between the Day-Time or Time-Period settings in the Select Criteria Type dropdown menu. If you select Time-Period, the editor changes as follows:
Mappings
Figure 229: Date/Time Criteria Editor (Time Period)
6.
Select or enter the parameters you want, then click the Add to List button. If a
parameter is invalid (for instance, the end date is inferior to the start date), it is
displayed in red in the Time Criteria List field.
To remove an existing parameter, select it in the Time Criteria List field, then click the
Remove Selected button.
To update an existing parameter, select it in the Time Criteria List field, then click the
Update Selected button.
To remove all parameters, click the Clear Parameters button.
When done, click the Submit button.
Enter the transformation (related to this specific output type) to apply in the Transformation field.
If the transformation is to replace part of an expression, it can use the matched group of the criteria.
\0 will be replaced by the whole criteria capability and \1 to \9 by the matched group. See
Groups on page 469 for more details.
See Routing Type on page 470 for a list of available transformation values.
Table 428: Output Type Transformation
Output Type
Transformation
None
No transformation is applied.
E164
If the Criteria value of the Mapping Type part is also a generic property, this is
applied to both the Calling E164 and Called E164 properties.
Called E164
Modifies the called party E.164 number as per Called / Calling E164 on
page 471.
Calling E164
Modifies the calling party E.164 number as per Called / Calling E164 on
page 471.
Name
applied to both the Calling Name and Called Name properties.
Called Name
Sets the display name of the called party as per Called / Calling Name on
page 472.
Calling Name Sets the display name of the calling party as per Called / Calling Name on
page 472.
TON
applied to both the Calling TON and Called TON properties.
Called TON
Sets the called party type of number as per Called / Calling TON on page 471.
Calling TON
Sets the calling party type of number as per Called / Calling TON on page 471.
497
Mappings
Table 428: Output Type Transformation (Continued)

Output Type
Transformation
NPI
applied to both the Calling NPI and Called NPI properties.
Called NPI
Sets the called party numbering plan indicator as per Called / Calling NPI on
page 471.
Calling NPI
Sets the calling party numbering plan indicator as per Called / Calling NPI on
page 471.
Host
applied to both the Calling Host and Called Host properties.
Called Host
Sets the remote IP address or domain name of the destination VoIP peer as per
Calling Host
Sets the remote IP address or domain name of the originating VoIP peer as per
Calling PI
Sets the presentation indicator as per Calling PI on page 472.
Calling SI
Sets the screening indicator as per Calling SI on page 472.
Calling ITC
Sets the information transfer capability as per Calling ITC on page 473.
URI
applied to both the Calling URI and Called URI properties.
Called URI
Sets the called URI as per Called / Calling URI on page 472.
Calling URI
Sets the calling URI as per Called / Calling URI on page 472.
Phone
Context
applied to both the Calling Phone Context and Called Phone Context properties.
Called Phone
Context
Sets the called Phone Context as per Called / Calling Phone Context on
page 474.
Calling
Phone
Context
Sets the calling Phone Context as per Called / Calling Phone Context on
page 474.
SIP
Username
applied to both the Calling SIP Username and Called SIP Username properties.
Called SIP
Username
Sets the called SIP Username as per Called / Calling SIP Username on
page 474.
Calling SIP
Username
Sets the calling SIP Username as per Called / Calling SIP Username on
page 474.
Last
Diverting
Reason
Sets the last diverting reason properties as per Last / Original Diverting
Reason on page 474.
Last
Diverting
E164
Sets the last diverting E.164 properties as per Last / Original Diverting E.164
on page 475.
Last
Sets the party number type of the last diverting number properties as per Last /
Diverting
Original Diverting Party Number Type on page 475.
Party
Number Type
498
Mappings
Table 428: Output Type Transformation (Continued)

Output Type
Transformation
Last
Diverting
Public Type
Of Number
Sets the public type of number of the last diverting number properties as per
Last / Original Diverting Public Type Of Number on page 475.
Last
Diverting
Private Type
Of Number
Sets the private type of number of the last diverting number properties as per
Last / Original Diverting Private Type Of Number on page 475.
Last
Diverting
Number
Presentation
Sets the presentation of the last diverting number properties as per Last /
Original Diverting Number Presentation on page 476.
Original
Diverting
Reason
Sets the original diverting reason properties as per Last / Original Diverting
Reason on page 474.
Original
Diverting
E164
Sets the original diverting E.164 properties as per Last / Original Diverting
E.164 on page 475.
Original
Sets the party number type of the original diverting number properties as per
Diverting
Last / Original Diverting Party Number Type on page 475.
Party
Number Type
Original
Diverting
Public Type
Of Number
Sets the public type of number of the original diverting number properties as per
Last / Original Diverting Public Type Of Number on page 475.
Original
Diverting
Private Type
Of Number
Sets the private type of number of the original diverting number properties as per
Last / Original Diverting Private Type Of Number on page 475.
Original
Diverting
Number
Presentation
Sets the Presentation of the original diverting number properties as per Last /
Original Diverting Number Presentation on page 476.
Called
Bearer
Channel
Sets the called bearer channel properties as per Called / Calling SIP
Username on page 474.
Calling
Bearer
Channel
Sets the calling bearer channel properties as per Called / Calling SIP
Username on page 474.
Debug
Reserved for debug configuration.
You cannot use Date/Time as an output type transformation.

7.
If applicable, enter the name of one or more subsequent mappings to execute in the Sub Mappings
field.
You can enter more than one mapping by separating them with commas. The mappings are
executed in sequential order.
You can use the Suggestion columns drop-down menu to select between existing values, if any.
499
Mappings
You may want to send the result of the first mapping to another one. Once the subsequent mapping
is finished, the call router continues to check the mapping entries for matching entries. For instance,
if the call router is checking the fourth mapping entry and that entry uses subsequent mapping, the
call router executes the subsequent mapping, then resumes checking the fifth mapping entry, and
so on.
The maximal number of subsequent interleaved mapping is 3.
8.
Click the Submit button to go back to the main Call Router > Route Config web page.
You can see a yellow Yes in the Config Modified section at the top of the window. It
warns you that the configuration has been modified but not applied (i.e., the Call Router
> Status differs from the Call Router > Route Config). The Route Config sub-menu is a
working area where you build up a Call Router configuration. While you work in this area,
the configured parameters are saved but not applied (i.e., they are not used to process
incoming calls). The yellow Yes flag warns you that the configuration has been modified
but is not applied.
9.
Click the Submit and Insert Expression button to create another expression for the
same type.
Click the Apply button to enable the mapping entry.

The current mappings applied are displayed in the Call Router > Status web page. You can also
Examples
The following are some examples of mappings:
Figure 230: Mappings Examples
Moving a Mapping Type or Expression Row

The mapping entries sequence is very important. The call router follows the mapping table rows as they are
entered in the web interface. If you want the call router to try to match one row before another one, you must
put that row first.
To move a mapping entry up or down:

1.
In the Mapping Type or Mapping Expression table, either click the

want to move until the entry is properly located.
2.
or
arrow of the row you
Deleting a Mapping Type or Expression Row

You can delete a mapping row from the Mapping Type or Mapping Expression table in the web interface.
To delete a mapping entry:
500
1.
Click the
2.
Signalling Properties
Standards Supported
RFC 3323: A Privacy Mechanism for the Session Initiation

Protocol (SIP) (only supports 'none' as Privacy level)
RFC 3325: Private Extensions to the Session Initiation

Protocol (SIP) for Asserted Identity within Trusted Networks
(supports 'id' as Privacy level. Accept/send P-AssertedIdentity and P-Preferred-Identity.)
Call signalling specifies how to set up a call to the destination Mediatrix unit or 3rd party equipment. Call
signalling properties are assigned to a route and used to modify the behaviour of the call at the SIP protocol
level.
Signaling Properties are applied after mappings rules.
Like the routing table, the signalling properties table finds the first matching entry. It then executes it by
modifying the behaviour of the call.
You can add up to 40 Signalling Properties.
Creating/Editing a Signalling Property

The web interface allows you to create a signalling property or modify the parameters of an existing one. The
signalling properties are called from a route as described in Routes on page 485.
To create or edit a signalling property:

1.
2
3
2.
Locate the Signaling Properties section.
3.
If you want to add a signalling property entry before an existing entry, locate the proper
row in the table and click the
button of this row.
If you want to add a signalling property entry at the end of existing rows, click the
button at the bottom right of the Signaling Properties section.
button.
This brings you to the Configure Signaling Properties panel.
501
Figure 232: Configure Signaling Properties Panel
4
6
5
7
8
10
11
12
4.
Enter the name of the signalling property in the Name field.

The name must be unique. It will be used in routes to call a specific signalling property as described
in Routes on page 485.
5.
Select whether or not the early connect feature is enabled in the Early Connect drop-down menu.
When early connect is enabled, the SIP call is connected by sending a 200 OK message instead of
a 183 Session Progress message with early media, if the called party answers the call. It allows
interoperability with units that do not support the 183 Session Progress with SDP message.
When early connect is disabled, call progress tones or announcements are transmitted in the early
SIP dialog.
6.
Select whether or not the early disconnect feature is enabled in the Early Disconnect drop-down
menu.
This feature is useful to avoid hearing the end of call tone when the far end party terminates the call
during a conference.
When early disconnect is:
enabled, the SIP BYE message is sent upon receiving the ISDN Disconnect signal.
disabled, the SIP BYE message is sent upon receiving the ISDN Call release signal.
If early disconnect is enabled but no ISDN Disconnect message is received, the SIP BYE
message is sent upon receiving an ISDN Call release signal as if the early disconnect was
disabled.
7.
Define the SIP messages destination (where an INVITE is sent) in the Destination Host field.
It can override the Called Host property set by a mapping rule because signalling properties are
applied after mappings.
You can also use the macro local_ip_port to replace the properties by the local IP address and
port of the listening network of the SIP gateway used to send the INVITE.
8.
Define whether or not to enable the 180 with SDP allowed feature in the Allow 180 SDP drop-down
menu.
Table 429: 180 with SDP Parameters
Parameter
Enable
502
Description
The unit can send a SDP in the provisional response 180. Thus when the ISDN
peer sends an alerting with indication to open the voice (or if the voice is already
opened), the unit sends a 180 with SDP. This is the default value.
Table 429: 180 with SDP Parameters (Continued)

Parameter
Disable
Description
A SIP 183 with SDP is sent instead of a 180 with SDP. This does not affect the 180
without SDP. This is useful if your proxy has issues receiving 180 with SDP
messages.
The SIP 183 with SDP replacing the SIP 180 with SDP is not sent if a 183 with
SDP has already been sent.
9.
Define whether or not to enable the 183 without SDP allowed feature in the Allow 183 No SDP dropdown menu.
Table 430: 183 without SDP Parameters
Parameter
10.
Description
Enable
When enabled, the unit sends a 183 without SDP upon receiving an ISDN
progress indicator without any indication to open a voice stream. This is the default
value.
Disable
When disabled, nothing is sent instead of a 183 without SDP. This does not affect
the 183 with SDP. This is useful if your proxy has issues receiving 183 without
SDP messages.
Set the privacy level of the call in the Privacy drop-down menu.
Table 431: Privacy Levels
Level
Description
Effects on incoming SIP call
Effects on outgoing SIP call
Disable No privacy
is used.
None
None
None
None
Adds two headers:
Use PAsserted
Identity
privacy.
Privacy: none
P-Asserted-Identity:
p_asserted_identity_v
alue
p_asserted_identity_value is the
call's From URI unless a SIP
header translation has been added
to the Signaling Properties for the
Identity-header.
Id
Use PPreferred
Identity
privacy.
The calling-name is empty and the

PI is set to restricted.
Always adds one header:
P-Preferred-Identity:
p_preferred_identity_v
alue
p_ preferred _identity_value is the

call's From URI unless a SIP
header translation has been added
to the Signaling Properties for the
Identity-header.
If the incoming call's PI property is
restricted, another header is
added:
Privacy: id
503
Table 431: Privacy Levels (Continued)

Level
Rpid
Description
Use
RemoteParty-ID
privacy.
Effects on incoming SIP call

None
Effects on outgoing SIP call

One header always added :
Remote-Party-ID:
remote_party_id_value
"Optional Friendly
Name"<sip:410202@10.4.125.12>
;party=calling
Where remote_party_id_value
should be set by the SIP Headers
Translation. It consists of an
optional friendly name followed by
the SIP URI and the party
direction.
Example:
Remote-Party-ID: "John
Doe"<sip:410202@10.4.125.
12>;party=calling
11.
Enter the name of one or more SIP headers translation to apply to the call in the SIP Headers
Translations field.
You must define SIP headers translations as defined in SIP Headers Translations on page 505.
You can use the Suggestion columns drop-down menu to select between existing translations, if
any.
You can enter more than one translation. In that case, the translations are separated with , and
are executed in sequential order.
12.
Enter the name of one or more call properties translation to apply to the call in the Call Properties
Translations field.
You must set call properties translations as defined in Call Properties Translations on page 508.
You can use the Suggestion columns drop-down menu to select between existing translations, if
any.
You can enter more than one translation. In that case, the translations are separated with , and
are executed in sequential order.
13.

14.
Click the Apply button to enable the signalling property entry.

The current properties applied are displayed in the Call Router > Status web page. You can also
Examples
The following are some examples of signalling properties:
504
SIP Headers Translations
Figure 233: Signalling Properties Examples
Moving a Signalling Property Row

The signalling properties entries sequence is very important. The call router follows the signalling properties
table rows as they are entered in the web interface. If you want the call router to try to match one row before
another one, you must put that row first.
To move a signalling property entry up or down:

1.
Either click the
or
2.
Deleting a Signalling Property Row

You can delete a signalling property row from the table in the web interface.
To delete a signalling property entry:

1.
Click the
2.

A SIP Headers Translation overrides the default value of SIP headers in an outgoing SIP message. It modifies
the SIP headers before the call is sent to its destination.
Like the routing table, the SIP headers translation table finds the first matching entry. It then executes it by
You can add up to 100 SIP Headers Translations.
Creating/Editing a SIP Headers Translation

The web interface allows you to create a SIP header translation or modify the parameters of an existing one.
The SIP headers translations are called from a signalling property as described in Signalling Properties on
page 501.
To create or edit a SIP headers translation:

1.
2
3
505
2.
Locate the SIP Headers Translations section.
3.
If you want to add a SIP headers translation before an existing entry, locate the proper
row in the table and click the
button of this row.
If you want to add a SIP headers translation at the end of existing rows, click the
button at the bottom right of the SIP Headers Translations section.
button.
This brings you to the Configure SIP Headers Translation panel.

Figure 235: Configure SIP Headers Translation Panel
4
6
5
7
4.
Enter the name of the SIP headers translation in the Name field.
5.
Set which SIP header is modified by this translation in the SIP Header drop-down menu.
Table 432: SIP Headers
SIP Header
6.
Description
From Header (Host Part)
Host part of the From header's URI.
From Header (User Part)
User part of the From header's URI.
Identity Header (Host Part)
Host part of the Identity header's URI.
Identity Header (User Part)
User part of the Identity header's URI.
Identity Header (Phone Number)
Phone number in the Identity header's tel URL.
Request Line (Host Part)
Host part of the Request line's URI.
Request Line (User Part)
User part of the Request line's URI.
To Header (Host Part)
Host part of the To header's URI.
To Header (User Part)
User part of the To header's URI.
Set what information is used to build the selected SIP header in the Built From drop-down menu.
Table 433: Built From Information
Built From
506
Description
Called E164
Use the called party E.164 property.
Destination Host
Use the destination host configured in the signalling

properties of which this translation is part.
Domain
Use the domain name configured in the unit.
Fix Value
Use a fix value as defined in the Fix Value field (see Step 7).
Host Name
Use the host name configured in the unit.
Table 433: Built From Information (Continued)

Built From
7.
Description
Local Ip
Use the local IP address.
Calling Bearer Channel
Use the calling bearer channel.
SIP Endpoint Username
Use the SIP username associated with the endpoint.
Calling Name
Use the calling party name property.
Calling E164
Use the calling party E.164 property.
If you have selected Fix Value in the Built From drop-down menu, enter a fix value to be inserted
in the SIP header in the Fix Value field.
For instance, you could hide the callers name in a SIP message by using the From Header (User
Part) SIP header and entering anonymous in the Fix Value field.
8.

9.
Click the Apply button to enable the SIP headers translation entry.
Example
The following is an example of SIP headers translations:
Figure 236: SIP Headers Translations Example
Moving a SIP Headers Translation Row

The SIP headers translation entries sequence is very important. The signalling properties table follows the SIP
headers translation table rows as they are entered in the web interface. If you want the signalling properties
table to try to match one row before another one, you must put that row first.
To move a SIP headers translation entry up or down:

1.
Either click the
or
2.
Deleting a SIP Headers Translation Row

You can delete a SIP headers translation row from the table in the web interface.
To delete a SIP headers translation entry:
1.
Click the
2.
507
Call Properties Translations

A Call Properties Translation overrides the default value of call properties in an incoming SIP message. It
modifies the call properties before the call is sent to its destination.
Like the routing table, the call properties translation table finds the first matching entry. It then executes it by
You can add up to 100 Call Properties Translations.
Creating/Editing a Call Properties Translation

The web interface allows you to create a call properties translation or modify the parameters of an existing
one. The call properties translations are called from a signalling property as described in Signalling
Properties on page 501.
To create or edit a call properties translation:

1.
2
3
2.
Locate the Call Properties Translations section.
3.
If you want to add a call properties translation before an existing entry, locate the
proper row in the table and click the
button of this row.
If you want to add a call properties translation at the end of existing rows, click the
button at the bottom right of the Call Properties Translations section.
button.
This brings you to the Configure Call Properties Translation panel.

Figure 238: Configure Call Properties Translation Panel
4
6
5
7
4.
508
Enter the name of the call properties translation in the Name field.
5.
Set which call property is modified by this translation in the Call Property drop-down menu.
Table 434: Call Properties
Call Property
Description
Called E164
Called party E.164 property.
Calling E164
Calling party E.164 property.
Called Name
Called party name property.
Calling Name
Calling party name property.
Called Uri
Called URI name property.
Calling Uri
Calling URI name property.
Called Bearer Channel Called bearer channel property.

6.
Set what information is used to build the selected call property in the Built From drop-down menu.
Table 435: Built From Information
Built From
Description
Domain
Use the domain name configured in the unit.
Fix Value
Use a fix value as defined in the Fix Value field (see Step 7).
From Header (Uri)
Use the From header's URI.
From Header (Friendly Name)
Use the friendly name part of the From header.
From Header (User Part)
Use the user part of the From header's URI.
Identity Header (Uri)
Use the Identity header's URI.
Identity Header (User Part)
Use the user part of the Identity header's URI.
Identity Header (Phone Number)
Use the phone number in the Identity header's tel URL. The
phone number is not retrieved if the received tel URL is
invalid. Only the phone number part is retrieved. Examples:
Received header: P-Preferred-Identity:

<tel:8298749;phone-context=819>
Retrieved phone number: 8298749

<tel:+8298749>
Retrieved phone number: 8298749

<tel:8298749>
Retrieved phone number: None, the received header

is invalid.
Identity Header (Friendly Name)
Use the friendly name in the Identity header's URI.
Local Ip
Use the local IP address.
Request Line (Uri)
Use the Request line's URI.
Request Line (User Part)
Use the user part of the Request line's URI.
To Header (Uri)
Use the To header's URI.
To Header (Friendly Name)
Use the friendly name part of the To header.
To Header (User Part)
Use the user part of the To header's URI.
509
7.
If you have selected Fix Value in the Built From drop-down menu, enter a fix value to be inserted
in the call property in the Fix Value field.
For instance, you could hide the callees name in a SIP message by using the From Header (User
Part) SIP header and entering anonymous in the Fix Value field.
8.

9.
Click the Apply button to enable the call properties translation entry.
Example
The following is an example of call properties translations:
Figure 239: Call Properties Translations Example
Moving a Call Properties Translation Row

The call properties translation entries sequence is very important. The signalling properties table follows the
call properties translation table rows as they are entered in the web interface. If you want the signalling
properties table to try to match one row before another one, you must put that row first.
To move a call properties translation entry up or down:

1.
Either click the
or
2.
Deleting a Call Properties Translation Row

You can delete a call properties translation row from the table in the web interface.
To delete a SIP headers translation entry:
510
1.
Click the
2.
Hunt Service
Hunt Service
Routes and mappings only manipulate address properties of a call. The hunt service hunts an incoming call
to multiple interfaces. It accepts a call routed to it by a route or directly from an interface and creates another
call that is offered to one of the configured destination interfaces. If this destination cannot be reached, the
hunt tries another destination until one of the configured destinations accepts the call. When an interface
accepts a call, the interface hunting is complete and the hunt service merges the original call with the new call
to the interface that accepted the call.
The hunt sequence is very important. The call router follows the hunt rows as they are entered in the web
interface. If you want the call router to try to match one row before another one, you must put that row first.
You can add up to 40 Hunts.
Creating/Editing a Hunt
The web interface allows you to create a hunt or modify the parameters of an existing one.
To create or edit a hunt:

1.
2
3
2.
Locate the Hunt section.
3.
If you want to add a hunt entry before an existing entry, locate the proper row in the
table and click the
button of this row.
If you want to add a hunt entry at the end of existing rows, click the
bottom right of the Hunt section.
button at the
button.
This brings you to the Configure Hunt panel.
511
Hunt Service
Figure 241: Configure Hunt Panel
4
5
6
7
8
4.
Enter the name of the hunt in the Name field.

The name must be unique. If more than one hunt have the same name, only the first hunt is used.
5.
Define a list of hunt destinations separated by commas in the Destinations field.

This is the interface, route, or hunt that is tried during the hunts interface hunting. The destination
can either be:
route-name: The call destination is the route name.
hunt-name: The call destination is the hunt name.
sip-name: The call destination is the SIP interface name.
isdn-name: The call destination is the ISDN interface name.
r2-name: The call destination is the R2 interface name.
e&m-name: The call destination is the E&M interface name.
fxs-name: The call destination is the FXS interface name.
fxo-name: The call destination is the FXO interface name.
Only FXS interfaces are supported if the selection algorithm Simultaneous is used (see Step 6).
For instance:
isdn-Slot2/Pri1, route-test (Mediatrix 3000 Series)
isdn-Bri1, route-test (Mediatrix 4400 Series)
6.
Select the algorithm used to select the order of the destination in the Selection Algorithm drop-down
menu.
The algorithm can be:
Sequential: The hunt tries the destination in the same order as listed. The first
destination hunted is the first listed.
Cyclic: The Mediatrix unit starts from the destination that follows the destination used
for the last hunt. Subsequent calls try another first destination in a round-robin method.
For instance, if the destination is set to 'x, y, z', the destination the hunt tries is in the
following order:
1. x,y,z
2. y,z,x
3. z,x,y
4. x,y,z
7.
512
Simultaneous: The hunt tries every available destination at the same time. The first
destination to pick up has the call. Other destinations stop ringing. This method can
only have FXS endpoints as destinations.
Set the maximal time, in seconds (s), allowed to an interface to handle the call in the Timeout field.
Hunt Service
After this timeout has elapsed, the next destination is tried when the current destination does not
answer. This feature is useful to ensure a minimal time of response and fallback to other
destinations. Some interfaces (e.g. SIP, which has a default timeout of 32 seconds) may wait an
arbitrary long time until an answer is returned.
Note: This parameter is not applicable if the selection algorithm Simultaneous is used (see Step 6).
Setting the field to 0 disables the timeout, which means that the call router waits indefinitely for the
interface to respond. This does not affect the internal interface timeouts (the ISDN timeout as
defined in ITU norms or the SIP transmission timeout) that will eventually stop the call and the call
router will try another destination.
Example:
You want a call from ISDN to SIP to fallback to another ISDN interface when the SIP destination
cannot be contacted within 5 seconds.
You thus create a hunt with the following destinations in order:
sip-[gateway name], isdn-[fallback interface]
and set the timeout to 5. The Selection Algorithm drop-down menu must be set to Sequential to
always try the SIP destination first.
Figure 242: Hunt Timeout Example
The Mediatrix unit has the following behaviour if the SIP transmission timeout has the default value
(32 seconds):
a.
A new call comes from an ISDN interface and the call router sets the destination of the call to
the isdn-to-sip hunt.
b.
The call router starts the hunt timeout (5 s) and tries the first destination sip-default.
c.
The SIP interface performs a DNS query to resolve the server name. The DNS result returns
server A and server B.
d.
The SIP interface sends an INVITE to the server A.
e.
The hunt timeout elapses, so the call router cancels the call to the SIP interface and tries the
second destination isdn-Slot3/Bri2. The hunt timeout is restarted.
f.
The SIP interface continues to send the INVITE retransmission until the SIP transmission
timeout elapses. RFC 3261 states that an INVITE request cannot be cancelled until the
destination sends a response. If the destination responds before the SIP transmission timeout
elapses, a CANCEL or BYE request is sent. The SIP interface will not try to use the server B
location.
The Mediatrix unit has the following behaviour if the SIP transmission timeout is set to 3 seconds:
a.
A new call comes from an ISDN interface and the call router sets the destination of the call to
the isdn-to-sip hunt.
b.
The call router starts the hunt timeout (5 s) and tries the first destination sip-default.
c.
The SIP interface performs a DNS query to resolve the server name. The DNS result returns
server A and server B.
d.
The SIP interface sends an INVITE to the server A.
e.
A SIP transmission timeout occurs after 4 seconds and the SIP interface sends an INVITE to
the server B.
f.
The hunt timeout elapses, so the call router cancels the call to the SIP interface and tries the
second destination isdn-Slot3/Bri2. The hunt timeout is restarted.
g.
The SIP interface continues to send the INVITE retransmission until the SIP transmission
timeout elapses. RFC 3261 states that an INVITE request cannot be cancelled until the
513
Hunt Service
destination sends a response. If the destination responds before the SIP transmission timeout
elapses, a CANCEL or BYE request is sent.
Note: The maximal response time of a SIP interface is the transmission timeout total of all SIP destination
locations + the DNS query time.
The SIP transmission timeout can be set in the Transmission Timeout field of the SIP Interop
section, SIP > Interop page (SIP Interop on page 316).
8.
Select call rejection causes to continue the hunt in the Causes field.
When an interface has a problem placing a call to the final destination, it drops the call by specifying
a drop cause based on Q.850 ISUP drop causes. Separate the causes with commas.
See Call Rejection (Drop) Causes on page 514 for a list of drop causes.
Note: This parameter is not applicable if the selection algorithm Simultaneous is used (see Step 6).
9.

10.
In the main Call Routing Config web page, click the Apply button to enable the hunt.
The current hunts applied are displayed in the Call Router > Status web page. You can also see
Examples
The following are some examples of hunts:
Figure 243: Hunt Example
Call Rejection (Drop) Causes

When a destination interface drops the call, the hunt service must supply a call rejection cause based on Q.850
ISUP drop causes. The Mediatrix unit offers the following drop causes categories:
Normal Event
Service or Option Not Available
Service or Option Not Implemented
Invalid Message
Protocol Error
Interworking
Note: You can use any custom code between 1 and 127.
514
Hunt Service
Normal Event
The following table lists all normal events drop causes. These causes are used to drop the original call.
Table 436: Normal Event Drop Causes
#
Cause
Description
Unassigned
(unallocated) number
The calling user requested a destination that cannot be reached because

the number is unassigned.
No route to specified
transit network
The destination is asked to route the call through an unrecognized network.

This may mean that:
The wrong transit network code was dialed.
The transit network does not serve this equipment.
The transit network does not exist.
No route to destination The called party cannot be reached because the network through which the
call has been routed does not serve the destination address.
Channel unacceptable
The sending entity cannot accept the channel most recently identified for
use in this call.
Call awarded and

being delivered in an
established channel
The user has been awarded the incoming call, which is being connected to a
channel already established to that user for similar calls.
16
Normal call clearing
The call is being cleared because one of the users involved with the call has
requested that the call be cleared (usually, a call participant hung up).
17
User busy
The called party is unable to accept another call because all channels are in
use. It is noted that the user equipment is compatible with the call.
18
No user responding
The called party does not respond to a call establishment message with
either an alerting or connect indication within the time allotted. The number
that is being dialed has an active D-channel, but the far end chooses not to
answer.
19
User alerting, no
answer
The called party has been alerted but does not respond with a connect
indication within the time allotted.
21
Call rejected
The remote equipment can accept the call but rejects it for an unknown
reason, although it could have accepted it because the equipment sending
this cause is neither busy nor incompatible.
22
Number changed
The called number indicated by the calling party is no longer assigned.
26
Non-selected user
clearing
The user has not been awarded the incoming call.
27
Destination out of
order
The destination indicated by the user cannot be reached because the

destination's interface is not functioning correctly. This can be a temporary
condition, but it could last for an extended period.
28
Invalid number format

(incomplete number)
The called party cannot be reached because the called party number is not
in a valid format or is not complete.
29
Facility rejected
The network cannot provide the facility requested by the user.
30
Response to STATUS
ENQUIRY
The STATUS message is generated in direct response to receiving a

STATUS ENQUIRY message.
31
Normal, unspecified
Reports a normal event only when no other cause in the normal class
applies.
515
Hunt Service
The following table lists all resource unavailable drop causes. These causes are used to hunt the next
destination.
Table 437: Resource Unavailable Drop Causes
#
Cause
Description
34
No circuit/channel
available
There is no appropriate circuit or channel presently available to handle the

call (usually, no B-channels are available to make the selected call).
38
Network out of order
The network is not functioning properly and the condition is likely to last for
an extended period.
41
Temporary failure
The network is not functioning properly and the condition should be resolved
quickly.
42
Switching equipment
congestion
Cannot reach the destination because the network switching equipment is

temporary experiencing high traffic.
43
Access information
discarded
The network could not deliver access information to the remote user as
requested.
44
Requested circuit/
channel not available
The other side of the interface cannot provide the circuit or channel
indicated by the requested entity.
47
Resource unavailable,
unspecified
The requested channel or service is unavailable for an unknown reason.
Service or Option Not Available

The following table lists all service or option not available drop causes. These causes are used to drop the
original call.
Table 438: Service or Option Not Available Drop Causes
#
Cause
Description
57
Bearer capability not

authorized
The user has requested a bearer capability that is implemented on the

equipment but the user is not authorized to use it.
58

presently available
The user has requested a bearer capability that is implemented by the

equipment and is currently unavailable.
63
Service or option not

available, unspecified
The network or remote equipment cannot provide the requested service

option for an unspecified reason.
Service or Option Not Implemented

The following table lists all service or option not implemented drop causes. These causes are used to drop the
original call.
Table 439: Service or Option Not Implemented Drop Causes
#
516
Cause
Description
65

implemented
The remote equipment does not support the requested bearer capability.
66
Channel type not

implemented
The remote equipment does not support the requested channel type.
69
Requested facility not

implemented
The remote equipment does not support the requested supplementary

service.
Hunt Service
Table 439: Service or Option Not Implemented Drop Causes (Continued)

#
Cause
Description
70
Only restricted digital

information bearer
capability is available
The calling party has requested an unrestricted bearer service but the
remote equipment only supports the restricted version of the requested
bearer capacity.
79
Service or option not

implemented,
unspecified
The network or remote equipment cannot provide the requested service

option for an unspecified reason. This can be a subscription problem.
Invalid Message
The following table lists all invalid message drop causes. These causes are used to drop the original call.
Table 440: Invalid Message Drop Causes
#
Cause
Description
81
Invalid call reference

value
The remote equipment has received a message with a call reference that is
not currently in use on the user-network interface.
82
Identified channel
does not exist
Indicates a call attempt on a channel that is not configured.
83
A suspended call
exists, but this call
identity does not
Attempted to resume a call with a call identity that differs from the one in use
for any presently suspended calls.
84
Call identity in use
The network has received a call suspended request containing a call identity
that is already in use for a suspended call.
85
No call suspended
The network has received a call resume request containing a call identity
information element that does not indicate any suspended call.
86
Call having the

requested call identity
has been cleared
The network has received a call identity information element indicating a

suspended call that has in the meantime been cleared while suspended.
88
Incompatible
destination
The remote equipment has received a request to establish a call with

compatibility attributes that cannot be accommodated.
91
Invalid transit network

selection
Received a transit network identification of an incorrect format was received.
95
Invalid message,
unspecified
Received an invalid message event.
Protocol Error
The following table lists all protocol error drop causes. These causes are used to drop the original call.
Table 441: Protocol Error Drop Causes
Cause
Description
96
Mandatory information
element is missing
The remote equipment has received a message that is missing an

information element (IE). This IE must be present in the message before the
message can be processed.
97
Message type nonexistent or not

implemented
The remote equipment has received a message with a missing information

element that must be present in the message before the message can be
processed.
517
Hunt Service
Table 441: Protocol Error Drop Causes (Continued)

#
Cause
Description
98
Message not
compatible with call
state or message type
non-existent or not
implemented
The remote equipment has received a message that is not allowed while in
the current call state.
99
Information element
non-existent or not
implemented
The remote equipment has received a message that includes information

elements or parameters that are not recognized.
100 Invalid information

element contents
The remote equipment has received a message that includes invalid

information in the information element or call property.
101 Message not

compatible with call
state
Received an unexpected message that is incompatible with the call state.
102 Recovery on time

expiry
A procedure has been initiated by the expiration of a timer in association

with error handling procedures.
111
An unspecified protocol error with no other standard cause occurred.
Protocol error,
unspecified
Interworking
The following table lists all interworking drop causes. These causes are used to drop the original call.
Table 442: Interworking Drop Causes
#
Cause
Description
127 Interworking,
unspecified
An event occurs, but the network does not provide causes for the action it
takes. The precise problem is unknown.
Moving a Hunt
The hunt sequence is very important. The call router follows the hunt rows as they are entered in the web
interface. If you want the call router to try to match one row before another one, you must put that row first.
To move a hunt entry up or down:

1.
Either click the
or
2.
Deleting a Hunt
You can delete a hunt row from the table in the web interface.
To delete a hunt entry:
518
1.
Click the
button of the row you want to move.
2.
SIP Redirects
SIP Redirects
The SIP Redirect allows SIP redirections to be configured. These SIP Redirect entries can be used as
destinations in route rules. This type of destination is valid only when the Source of the route rule is a SIP
interface.
When a route rule is configured with a SIP Redirect destination, incoming SIP Invites are replied with a 302
"Moved Temporarily" SIP response.
Creating/Editing a SIP Redirect

The web interface allows you to create a SIP Redirect or modify the parameters of an existing one.
To create or edit a SIP Redirect:

1.
2
3
2.
Locate the SIP Redirects section.
3.
If you want to add a SIP Redirect entry before an existing entry, locate the proper row in
the table and click the
button of this row.
If you want to add a SIP Redirect entry at the end of existing rows, click the
at the bottom right of the SIP Redirects section.
button
button.
This brings you to the Configure SIP Redirect panel.

Figure 245: Configure SIP Redirect Panel
4.
Enter the name of the SIP Redirect in the Name field.

The name must be unique. If more than one SIP Redirect have the same name, only the first SIP
Redirect is used.
5.
Set the Destination Host field with the host address inserted in the Moved Temporarily response.
6.
519
SIP Redirects
7.
In the main Call Routing Config web page, click the Apply button to enable the SIP Redirect.
The current SIP Redirects applied are displayed in the Call Router > Status web page. You can also
Examples
The following are some examples of SIP Redirects:
Figure 246: SIP Redirects Example
Moving a SIP Redirect

The SIP Redirect sequence is very important. The call router follows the SIP Redirect rows as they are entered
in the web interface. If you want the call router to try to match one row before another one, you must put that
row first.
To move a SIP Redirect entry up or down:

1.
Either click the
or
2.
Deleting a SIP Redirect

You can delete a SIP Redirect row from the table in the web interface.
To delete a SIP Redirect entry:
520
1.
Click the
button of the row you want to move.
2.
Hairpinning
Hairpinning
Hairpinning is defined as a call between two telephony endpoints without using SIP.
Hairpinning is only supported between ISDN and R2 endpoints.
The Call Router does not produce an error when configuring a route between telephony interfaces that do not
support the hairpinning, but the call will fail if it uses the configured route. This is not limited to direct routes in
the route table configuration, but also for calls that use multiple routes and hunts where the source and the
final destination are telephony interfaces that do not support hairpinning.
Calls between the following telephony interfaces are allowed:
ISDN -> ISDN

ISDN -> R2
ISDN -> SIP
R2 -> ISDN
R2 -> R2
R2 -> SIP
FXS -> SIP
FXO -> SIP
SIP -> ISDN
SIP -> R2
SIP -> FXS
SIP-> FXO
Hairpinning is thus possible with the Mediatrix 440x, Mediatrix 34xx, Mediatrix 35xx, and Mediatrix 36xx
models.
Calls between the following telephony interfaces are not supported:
ISDN -> FXS

R2 -> FXS
FXO -> FXS
FXS -> FXS
FXS -> ISDN
FXS -> R2
FXS ->FXO
SIP -> SIP
Hairpinning is not possible with the Mediatrix 41xx, C7 and Mediatrix 33xx models.
The Mediatrix 37xx models partially support hairpinning with their ISDN card.
You can still make a loopback call on the same unit between two interfaces that do not support hairpinning by
performing a SIP loopback. To do this, you need to:
make a route from the source telephony interface to a SIP interface (Routes on page 485)
make a route from the SIP interface to the destination telephony interface (Routes on
page 485)
associate a Signalling Properties to override the SIP destination (Signalling Properties on

page 501)
Configuration Examples
The following are examples of configuration you could do with the call router.
521
Configuration Examples
Figure 247: Configuration Examples
522
H A P T E R
43
Auto-Routing Configuration
This chapter describes the auto-routing feature.
Auto-Routing
The auto-routing feature is an aid to call routing configuration. When this feature is enabled, routing rules are
automatically generated for all endpoints marked as "Auto-routable". For each auto-routable endpoint, two
rules are generated and added to the Call Router: one directing incoming calls from the associated autorouting SIP gateway to the endpoint, and one sending outgoing calls from the endpoint to the associated autorouting SIP gateway.
The auto-routing routes are not displayed in the Route Configuration page because you cannot edit them.
They are however listed in the Status page and are attributed a type:
User: the route has been manually entered by the user.

Auto: this is an auto-routing route.
Note: Auto-routing can only be used if the username of the endpoint is an E.164 string and the username
part of the request-URI of the received INVITE can be converted into an E.164. See Manual Routing on
To activate auto-routing:
1.
In the web interface, click the Call Router link, then the Auto-routing sub-link.
Figure 248: Call Router Auto-Routing Web Page
2
3
4
5
6
7
2.
In the top section, set the Auto-routing drop-down menu with the proper behaviour.
If you select Enable, routes are automatically added to the Route Table in order to connect the
endpoints marked as eligible for auto-routing (see Step 3) and the designated SIP gateway (see
Step 4). These automatic routes are displayed in the Call Router > Status page, but do not show up
in the Call Router > Route Configuration page.
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Chapter 43 - Auto-Routing Configuration
3.
Auto-Routing
Select the type of criteria to use to create automatic rules from SIP to the telephony endpoints in the
Criteria Type drop-down menu.
Table 443: Criteria Types
Parameter
4.
Description
E164
The E.164 associated with the endpoint is used as criterion.
Sip Username
The SIP username associated with the endpoint is used as

criterion.
Set the Incoming Mappings field with the name of the properties manipulations associated with the
route from the SIP gateway to the endpoint.
You can specify more than one mapping by separating them with ','. They are executed in sequential
order. See Mappings on page 491 for more details.
5.
Set the Outgoing Mappings field with the name of the properties manipulations associated with the
route from the endpoint to the SIP gateway.
You can specify more than one mapping by separating them with ','. They are executed in sequential
order. See Mappings on page 491 for more details.
6.
Set the Incoming Signaling Properties field with the name of the signaling properties associated with
the route from the SIP gateway to the endpoint.
See Signalling Properties on page 501 for more details.
7.
Set the Outgoing Signaling Properties field with the name of the signaling properties associated with
the route from the endpoint to the SIP gateway.
See Signalling Properties on page 501 for more details.
8.
Click the Submit button to enable auto-routing.

The current routes applied are displayed in the Call Router > Status web page. They are added at
the end of the routes that are already present, if any. This ensures that the user-defined routes
always have precedence over the automatic routes when both types of routes apply to the same
endpoint.
Endpoints Auto-Routing
This section allows you to link an endpoint to several SIP gateways.
To set Endpoints auto-routing parameters:

1.
In the Endpoints auto-routing section of the Auto-routing page, locate the proper endpoint in the
table and click the
button.
The Configure Auto-Routing page displays:
524
Auto-Routing
Figure 249: Configure Auto-Routing Section
2
3
4
2.
Select whether or not automatic routes are generated for the endpoint when auto-routing is enabled
in the Auto-routable drop-down menu.
Table 444: Auto-routable Parameters
Parameter
3.
Description
Enable
Automatic routes allowing incoming and outgoing calls to and from

the endpoint are added to the Route Table when auto-routing is
enabled.
Disable
Automatic route generation is turned off for this endpoint.
HardwareDependent
Automatic routes are generated if the endpoint belongs to an FXS

interface.
Select the SIP gateways to use as the destination of outgoing calls and the source of incoming calls
when generating auto-routing rules in the Auto-routing Gateway drop-down menu.
If you leave the field blank, it is the same as disabling the auto-routing feature.
More than one SIP gateway can be defined. The SIP gateways names are separated by comas.
Example:
gw1,gw2,gw3
When one SIP gateway is defined:
A route is automatically created from the SIP gateway to the telephony interface.
A route is automatically created from the telephony interface to the SIP gateway if the
Auto-routing Destination field is empty. Otherwise, the destination of the route uses the
destination defined in the Auto-routing Destination field.
When several SIP gateways are defined:
Routes are automatically created from each defined SIP gateway to the telephony
interface.
A route is automatically created from the telephony interface to the destination defined
in the Auto-routing Destination field. No route is created if the destination is left empty.
If available, two additional parameters are displayed:
If an endpoint has a telephone number that is associated with it, it is displayed in the
corresponding E164 column. This is the User Name field as configured in the SIP >
Registration page as long as the name follows the E.164 syntax.
If an endpoint has a friendly name that is associated with it, it is displayed in the
corresponding Name column. This is the Friendly Name field as configured in the SIP >
Registration page.
Pleas note that routes are created only if a user name is associated with the telephony endpoint in
the registration table. See Endpoints Registration on page 291 for more details.
525
4.
Auto-Routing
Set the destination to use for the routes from the telephony interface in the Auto-routing Destination
field.
The destination can be:
route-name: The route destination is set to the route name.
hunt-name: The route destination is set to the hunt name.
sip-name: The route destination is set to the SIP interface name.
isdn-name: The route destination is set to the ISDN interface name.
r2-name: The route destination is set to the R2 interface name.
e&m-name: The route destination is set to the E&M interface name.
fxs-name: The route destination is set to the FXS interface name.
fxo-name: The route destination is set to the FXO interface name.
5.
You can copy the configuration of the selected endpoint to one or more endpoints of the Mediatrix
unit in the Apply to the Following Endpoints section at the bottom of the page. You can select
specific endpoints by checking them, as well as use the Check All or Uncheck All buttons.
6.
When you are finished, you have the choice to:
Click the Submit button to enable auto-routing.

The current routes applied are displayed in the Call Router > Status web page. They are
added at the end of the routes that are already present, if any. This ensures that the
user-defined routes always have precedence over the automatic routes when both types
of routes apply to the same endpoint.
Click the Submit & Create Hunt button to perform a submit action and go to the hunt
creation page. This option is available only if the destination is set to an unexisting
hunt.
Click the Submit & Edit Hunt button to perform a submit action and go to the hunt
edition page. This option is available only if the destination is set to an existing hunt.
Click the Submit & Create Route button to perform a submit action and go to the route
creation page. This option is available only if the destination is set to an unexisting
route.
Click the Submit & Edit Route button to perform a submit action and go to the route
edition page. This option is available only if the destination is set to an existing route.
Manual Routing
Auto-routing can only be used if the username of the endpoint is an E.164 string and the username part of the
request-URI of the received INVITE can be converted into an E.164.
The conversion of a username into an E.164 follows these rules:
The prefix + is removed. Note that if the Map Plus To TON International drop-down menu is
set to Enable, the call property type of number is set to international. See Misc Interop on
The visual separator - is removed.

The username parameter is removed. The username parameter is a suffix beginning with ;.
All remaining characters need to be 0123456789*#abcdABCD.
Examples of conversion:
5551234 --> 5551234
#20 --> #20
555-1234 --> 5551234
+1-819-555-1234 --> 18195551234
5551234;parameter --> 5551234
5551234_parameter --> cannot convert
To use a username not compatible with E.164, you must disable the auto-routing and use manual routes.
Figure 250 gives an example of manual routes for an endpoint using 5550001_paramter as user.
526
Auto-Routing
Figure 250: Manual Routes Example
527
528
Auto-Routing
H A P T E R
44
Configuration Script
This chapter describes the configuration script download feature, which allows updating the Mediatrix unit
configuration by transferring a configuration script from a remote server or from the local file system. The
Mediatrix unit is the session initiator, which allows NAT traversal. You can also configure the Mediatrix unit to
automatically update its configuration.
You can also generate a configuration script from the running configuration of the Mediatrix unit.
Configuration scripts are files containing textual commands that are sent over the network to a Mediatrix unit.
Upon receiving the file, the unit executes each command line in sequence. Script commands can assign
values to configuration variables, or execute configuration commands. See Creating a Configuration Script
on page 549 for more details on how to create a configuration script.
Scripts are written by the system administrator and can be used to accomplish various tasks, such as
automating recurrent configuration tasks or batch-applying configuration settings to multiple devices. Scripts
can be executed once or periodically at a specified interval. They can also be scheduled to execute when the
Mediatrix unit restarts.
Configuration script server setup.

Configuration script server parameters.
Configuration download procedure.
Generating a configuration script from the running configuration.
Automatic configuration update parameters.
How to create a configuration script from scratch.
Standards Supported
RFC 959: File Transfer Protocol (client-side only)
RFC 2617: HTTP Authentication: Basic and Digest Access

Authentication
RFC 3617: Uniform Resource Identifier (URI) Scheme for the

Trivial File Transfer Protocol
draft-ietf-http-authentication-03
Configuration Script Server

To download a configuration script, you may need to setup the following applications on your computer:

SNTP server properly configured
HTTPS server with proper root path

When you perform a configuration script download by using the TFTP (Trivial File Transfer Protocol) protocol,
you must install a TFTP server running on the PC designated as the TFTP server host. It is assumed that you
know how to set the TFTP root path. If not, refer to your TFTP servers documentation.
531
Chapter 44 - Configuration Script
Configuring the SNTP Server

When you use the automatic configuration script update feature (see Automatic Configuration Update on
page 544 for more details) or the HTTPS protocol, you need to have a time server SNTP that is accessible
and properly configured. It is assumed that you know how to configure your SNTP server. If not, refer to your
SNTP servers documentation. You can also refer to SNTP Configuration on page 95 for more details on how
to configure the Mediatrix unit for a SNTP server.
at boot time.

When you to perform a configuration script download by using the HTTP protocol, you must install a HTTP
server running on the PC designated as the server host. It is assumed that you know how to set the root path.
If not, refer to your HTTP servers documentation.
Configuring the HTTPS Server

Standards Supported
RFC 2459: X.509 Digital Certificates
RFC 2818: HTTP Over TLS (client side only)
RFC 3268: Advanced Encryption Standard (AES)

Ciphersuites for Transport Layer Security (TLS)
RFC 3280: Internet X.509 Public Key Infrastructure Certificate

and Certificate Revocation List (CRL) Profile
When you perform a configuration script download that requires authentication or privacy by using the HTTP
over the Transport Layer Security (TLS) protocol (HTTPS), you must install a HTTPS server running on the
PC designated as the server host. It is assumed that you know how to set the root path and SSL/TLS security
configuration. If not, refer to your HTTPS servers documentation.
Caution: You must have a time server SNTP that is accessible and properly configured, or the automatic
configuration update feature may not work properly. It is assumed that you know how to configure your
SNTP server. If not, refer to your SNTP servers documentation. You can also refer to SNTP Configuration
on page 95 for more details on how to configure the Mediatrix unit for a SNTP server.
When two peers establish a HTTPS connection, they negotiate and decide on a cipher suite to use for data
encryption. The client suggests a list of cipher suites and the server selects one that it supports. Some cipher
suites are more secured than others. The Mediatrix unit acts as a client.
The Mediatrix unit suggests a wide range of cypher suites, which includes cipher suites that are not very
secure. The final choice rests with the server and it is thus possible that the transfer uses a SSL/TLS link that
is not very secure.
Media5 recommends to use cipher suites based on the RSA key exchange mechanism, because the DiffieHellman key exchange mechanism introduces a noticeable delay in the HTTPS session establishment.
Furthermore, Media5 recommends using cipher suites based on the following SSL/TLS algorithms:
Table 445: Suggested Secure Parameters
Suggested Parameter
Key Exchange Mechanism
Ciphers
532
Description
RSA
Diffie-Hellman
AES (128 and 256 bits)
3DES (168 bits)
Table 445: Suggested Secure Parameters (Continued)

Suggested Parameter
Message Digests
Description
SHA-1
The following six recommended cipher suites are based on the algorithms of Table 445:
Table 446: Recommended Cipher Suites
ID
Name
0x0035
0x0039
0x000a
0x0016
0x002f
0x0033
Scripts Transfer Cipher Suite Settings


by using the CLI
533
You can define the allowed cipher suites for the network security settings when using the HTTPS transfer
protocol. When the device initiates an HTTPS connection to a server it will negotiate the cipher suite according
to its configuration.
Parameter
CS1
CS2
534
Description
TLS_DHE_DSS_WITH_DES_CBC_SHA
This represents a secure configuration using SHA-1.This value includes the following cipher
suites:

Parameter
Description
this represents the most secure configuration using SHA-2. Only the most secure cipher
CS3
To set the script transfer cipher suite configuration parameter:

1.
In the confMIB, locate the scripttransferGroup folder in the scriptGroup folder.
2.
Set the script transfer cipher suite configuration in the ScriptsTransferCipherSuite variable.
conf.ScriptsTransferCipherSuite="Value"

Value
Meaning
100
CS1
200
CS2
300
CS3
Certificates
The Mediatrix unit contains embedded security certificates formatted as per ITU x.509 and RFC 3280. The
certificates are factory-installed. You can also add new certificates as described in Chapter 47 - Certificates
Management on page 569.
When contacting a HTTPS server, the Mediatrix unit establishes a TLS connection by (among others):
negotiating cipher suites

checking the server certificates validity (dates)
The Mediatrix unit then checks the servers identity by validating the host name used to contact it against the
information found in the servers certificate, as described in RFC 2818, section 3.1.
If any of the above does not succeed, the Mediatrix unit refuses the secure connection. To help detect such
errors, you can increase the syslog messages level.
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Generating a Configuration Script from the Running Configuration

You can generate a configuration script from the running configuration of the Mediatrix unit and export it.
You can export the configuration in two ways:
To a URL you specify with one of the supported transfer protocols.

By directly downloading the exported script via your web browser. This option uses the
protection provided by your web browser (it is protected if you log on to the unit via HTTPS).
Exporting a Configuration to a URL

The Export Script section allows you to generate a configuration script from the running configuration of the
Mediatrix unit and export it.
To export a configuration to a URL:

1.
In the web interface, click the Management link, then the Configuration Scripts sub-link.
Figure 251: Management Configuration Scripts Web Page
2
3
4
2.
Select the content to export in the generated configuration script in the Content drop-down menu.
Table 449: Exported Configuration Script Content
Parameter
All Config
Description
Exports everything.
Modified Config Export only the configuration that has been modified (differs from the default
values).
3.
Set the Service Name field with the name of the service from which to export configuration.
You can use the Suggestion drop-down menu to select one of the available services. You can use
the special value All to export the configuration of all services.
4.
Set the Send To URL field with the URL where to send the exported configuration script.
The URL should follow this format:
protocol://[user[:password]@]hostname[:port]/[path/]filename
The brackets [ ] denote an optional parameter.

The filename may contain a %mac% macro that is substituted by the MAC address of the unit at the
moment of sending the configuration script. For instance, the %mac%.cfg value for a Mediatrix unit
with MAC address 0090f12345ab will be 0090f12345ab.xml.
The filename may contain macros that are substituted at the moment of sending the configuration
script. The supported macros are:
536
%mac% - the MAC address of the unit
%version% - the MFP version of the unit
For instance, the %mac%.cfg value for a Mediatrix unit with MAC address 0090f12345ab will be
0090f12345ab.xml.
The transfer protocols supported are:
TFTP
FTP
FILE
Examples of valid URLs:
tftp://tftpserver.com:69/folder/script.cfg
ftp://guest@ftpserver.com/script.cfg
ftp://username:password@ftpserver.com/script.cfg
file://script.cfg
The protocols default port is used if none is specified.

5.
Set the Privacy Key field with the key used to encrypt the configuration script to export.
Caution: The Privacy Key field is not accessible if you have the User or Observer access right. See Users
The key is encoded in hexadecimal notation. You can thus use characters in the range 0-9, A-F,
and a-f. All other characters are not supported.
The maximum key length is 64 characters, which gives a binary key of 32 bytes (256 bits). It is the
maximum key size supported by the MxCryptFile application.
For instance, a 32-bit key could look like the following: A36CB299.
If the field is empty, the configuration script is not encrypted.
To decrypt the exported configuration script, you must use the MxCryptFile application. MxCryptFile
is a command line tool that encrypts or decrypts files to be exchanged with the Mediatrix unit.
Contact your sales representative for more details.
6.
Initiate the configuration scripts exportation by clicking the Submit & Export Now button at the
bottom of the page.
The Mediatrix unit immediately generates and transfers a configuration script based on the export
settings set in the previous steps.
Exporting a Configuration Script to your PC

This section describes how to export the configuration of a Mediatrix unit to the PC.
If you are currently using an unsecure HTTP access, script transfers through web browser are disabled. This
is to avoid transferring the configuration in clear text. To enable the section, you can:
Access the secure site (recommended) by clicking the corresponding link at the top of the
window. This is the recommended way to proceed.
Activate unsecure certificate transfer by clicking the corresponding link at the top of the window.
This is not recommended.
537
To export a configuration script to the PC:

1.
In the Transfer Scripts Through Web Browser section of the Configuration Scripts page, select the
content to export in the generated configuration script in the Content drop-down menu.
Table 450: Exported Configuration Script Content
Parameter
All Config
Description
Exports everything.
Modified Config Export only the configuration that has been modified (differs from the default
values).
Figure 252: Transfer Scripts Through Web Browser Section
3
1
2
2.
If required, set the Privacy Key field with the key used to encrypt the configuration script to export.
The maximum key length is 64 characters, which gives a binary key of 32 bytes (256 bits). It is the
maximum key size supported by the MxCryptFile application.
If the field is empty, the configuration script is not encrypted.
To decrypt the exported configuration script, you must use the MxCryptFile application. MxCryptFile
is a command line tool that encrypts or decrypts files to be exchanged with the Mediatrix unit.
Contact your sales representative for more details.
3.
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Click the Export & Download button.
Executing Configuration Scripts Settings

You can execute the configuration in two ways:
From the configuration script server URL that you specify with one of the supported transfer
protocols.
By directly executing the script via your web browser. This option uses the protection provided
by your web browser (it is protected if you log on to the unit via HTTPS).
Executing a Script from the Configuration Script Server

This section describes how to configure the IP address and port number of the configuration script server. This
server contains the configuration scripts the Mediatrix unit will download.
When performing a configuration script download, you can download two different scripts:
A generic configuration script that should be used to update a large number of units with the
same configuration.
A specific configuration script that contains the configuration for a single unit, for instance the
telephone numbers of its endpoints.
You can use a specific configuration script but no generic configuration script or vice-versa. You can also use
both generic and specific configuration scripts. When both the generic and specific configuration scripts are
downloaded, settings from the specific configuration script always override the settings from the generic
configuration script. These scripts must be located in the same directory.
Each script is executed independently from the other one. A script that is empty, cannot be found or has an
invalid syntax does not prevent the execution of the other script. If one or both scripts fail, error messages are
sent.
To set the configuration scripts server parameters:

1.
In the Configuration Scripts section of the Configuration Scripts page, set the name of the generic
configuration script to download in the Generic File Name field.
This script should be used to update a large number of units with the same configuration. The script
name is case sensitive hence it must be entered properly.
If you select File in the Transfer Protocol drop-down menu (Step 4), this means that you can select
a script located in the units persistent file system. You can use the Suggestion drop-down menu to
select one of the available scripts in the file system.
To see the content of the units file system persistent memory, go to the File Manager (Chapter 51
- File Manager on page 613). All installed configuration scripts/images are listed.
This field may contain some macros that are substituted by the actual value at the moment of
fetching the configuration script. The supported macros are:
%product% - the Product name of the unit.
%productseries% - the Product series name of the unit.
For instance, the %mac%.xml value for a Mediatrix unit with MAC address 0090f12345ab will be
0090f12345ab.xml.
If you leave the field empty, the Mediatrix unit does not download the generic configuration script.
539
Figure 253: Execute Scripts Section

1
2
3
5
6
8
2.
Set the name of the specific configuration script to download in the Specific File Name field.
This script should be used to update the configuration of a single unit. The script name is case
sensitive hence it must be entered properly.
a script located in the units persistent file system. You can use the Suggestion drop-down menu to
select one of the available scripts in the file system.
This field may contain a macro that is substituted by the actual value when downloading the
configuration script. The Mediatrix unit supports the %mac% macro, which will be substituted by the
MAC address of the unit. For instance, the %mac%.xml value for a Mediatrix unit with MAC
address 0090f12345ab will be 0090f12345ab.xml.
This field may contain some macros that are substituted by the actual value when downloading the
configuration script. The supported macros are:
0090f12345ab.xml.
If the variable is empty (after macro substitution), the Mediatrix unit does not download the specific
configuration script.
3.
Set the path of the directory where the configuration scripts are located in the Location field.
The path is case sensitive hence it must be entered properly. It is relative to the root of the
configuration scripts server. Use the / character when defining the path to indicate sub-directories.
0090f12345ab.xml.
The path differs depending on the transfer protocol selected (see Step 5).
Example: All Transfer Protocols Except File
Lets consider the following example for all protocols except File:
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The directory that contains the configuration script is called: Config_Script.
This directory is under C:/Root/Download.

Table 451: Path Configurations Example
Root Path
Corresponding Path Name
c:/root/download
Config_Script
c:/
root/download/Config_Script
c:/root
download/Config_Script
The following are some tips to help your download process:
Use the / character when defining the path to indicate sub-directories. For instance,
root/download.
If you are using the TFTP protocol to download the software, note that some TFTP
servers on Windows do not recognize the / character and produce an error. In this
case, use the \ character.
Use basic directory names, without spaces or special characters such as ~, @, etc.,
which may cause problems.
Cut and paste the path and/or name of the directory that contains the extracted scripts
into the configuration download path of the Mediatrix unit (you may have to convert \
into /) to eliminate typographical errors.
Note that you can define the C:/Root/Download part as you want. The script names may also differ
from the example shown above.
When the Transfer Protocol is set to File, you may prefix the path by one of the following to indicate
storage media:
Persistent: for onboard persistent storage. The configuration script is saved into the
persistent file system of the Mediatrix unit (in flash memory). This is the default value.
Volatile: for onboard non-persistent storage. The configuration script is saved into the
non-persistent RAM memory of the Mediatrix unit. All information is lost the next time
the unit restarts.
Table 452: Path Configurations Example (File)
Location
Onboard persistent storage of the
Mediatrix unit under the directory
Script-1
4.

Persistent:Script-1
or
Script-1
Set the transfer protocol to transfer the configuration scripts in the Transfer Protocol field.
You can select from five different transfer protocols:
HTTPS: HyperText Transfer Protocol over Transport Layer Security.
FTP: File Transfer Protocol. Note that the Mediatrix unit FTP client does not support the
EPSV command.
Note: The configuration script download via TFTP can only traverse NATs of types Full Cone or
Restricted Cone. If the NAT you are using is of type Port Restricted Cone or Symmetric, the script
transfer will not work.
File: Complete path to a configuration image in a storage device. You can view and
manage all files created with the File transfer protocol by using the File Manager. See
File Manager on page 613 for more details.
HTTP and HTTPS support basic or digest authentication mode as described in RFC 2617. HTTPS
requires a valid certificate.
541
If you have selected HTTP or HTTPS, please note that your server may activate some caching
mechanism for the script download. This mechanism caches the initial script download for later
processing, thus preventing changes or update of the original script. This can cause strange
problems if you want to edit a configuration script to modify values and upload it immediately. The
result will still return the original script and not the new one.
5.
If your server requires authentication when downloading the configuration script, set the following:
6.
Set the static configuration scripts server IP address or domain name and port number in the Host
Name field.
This is the current address of the PC that hosts the configuration scripts.
Use the special port value zero to indicate the protocol default. For instance, the TFTP default port
is 69, the HTTP default port is 80, and the HTTPS default port is 443.
The default value is 0.0.0.0:0.
7.
Set the key used to decrypt configuration scripts when they are encrypted in the Privacy key field.
You can secure the exchange of configuration scripts between the server and the Mediatrix unit. A
privacy key allows the unit to decrypt a previously encrypted configuration script.
To encrypt a configuration script, you must use the MxCryptFile application. MxCryptFile is a
command line tool that encrypts files before sending them to the Mediatrix unit. Contact your sales
representative for more details.
Each character encodes 4 bits and the maximum key length is 112 characters, which gives a binary
key of 56 bytes. It is the maximum supported by the MxCryptFile application.
This key must match the key used for the encryption of the relevant configuration script.
If the field is empty, the configuration script is not decrypted by the unit and the configuration update
fails.
Encryption is auto-detected.
8.
Define whether or not to allow the execution of a script even if it is identical to the last executed script
in the Allow Repeated Execution drop-down menu.
Table 453: Allow Repeated Execution Parameters
Parameter
Description
Auto
Uses the configured value of ScriptsAllowRepeatedExecution.
Enable
Allows repeated execution of the same script.
Disable
Does not allow repeated execution of the same script.
The script retry mechanism is not enabled for the DHCP triggered scripts (see DHCPv4 AutoProvisioning on page 546 for more details).
542
9.
Click Submit & Execute Now to execute the script.
Executing a Script from your PC

This section describes how to execute a script located on the PC.
If you are currently using an unsecure HTTP access, script transfers through web browser are disabled. This
is to avoid transferring the configuration in clear text. To enable the section, you can:
Access the secure site (recommended) by clicking the corresponding link at the top of the
window. This is the recommended way to proceed.
Activate unsecure certificate transfer by clicking the corresponding link at the top of the window.
This is not recommended.
To execute a script from the PC:

1.
In the Transfer Scripts Through Web Browser section of the Configuration Scripts page, type the
name of a script in the Upload Parameters field or select an existing one on the PC with the Browse
button.
When a script is executed, it is not installed in the units file system persistent memory. You can click
the Clear Selection link to empty the field and enter another name.
Figure 254: Transfer Scripts Through Web Browser Section
3
1
2.
If required, set the key used to decrypt configuration scripts when they are encrypted in the Privacy
key field.
You can secure the exchange of configuration scripts between the server and the Mediatrix unit. A
privacy key allows the unit to decrypt a previously encrypted configuration script.
To encrypt a configuration script, you must use the MxCryptFile application. MxCryptFile is a
command line tool that encrypts files before sending them to the Mediatrix unit. Contact your sales
representative for more details.
Each character encodes 4 bits and the maximum key length is 112 characters, which gives a binary
key of 56 bytes. It is the maximum supported by the MxCryptFile application.
This key must match the key used for the encryption of the relevant configuration script.
If the field is empty, the configuration script is not decrypted by the unit and the configuration update
fails.
Encryption is auto-detected.
3.
Click the Upload & Execute button.
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Configuration Download Procedure
Configuration Download Procedure

The following steps explain how to download configuration scripts from the web interface.
Note: The configuration download via TFTP can only traverse NATs of types Full Cone or Restricted
Cone. If the NAT you are using is of type Port Restricted Cone or Symmetric, the file transfer will not
work.
To download configuration scripts:

1.
Place the configuration scripts to download on the computer hosting the configuration scripts server.
These scripts must be in a directory under the servers root path.
2.
Initiate the configuration scripts download by clicking the Submit & Execute Now button at the
bottom of the page.
The Mediatrix unit immediately downloads the configuration scripts.
Automatic Configuration Update

This section describes how to configure the Mediatrix unit to automatically update its configuration. This
update can be done:
Every time the Mediatrix unit restarts.

At a specific time interval you can define.
Automatic Update on Restart

The Mediatrix unit may download new configuration scripts each time it restarts.
NAT Variations
NAT treatment of UDP varies among implementations. The four treatments are:
Full Cone: All requests from the same internal IP address and port are mapped to the same
external IP address and port. Furthermore, any external host can send a packet to the internal
host by sending a packet to the mapped external address.
Restricted Cone: All requests from the same internal IP address and port are mapped to the
same external IP address and port. Unlike a full cone NAT, an external host (with IP address
X) can send a packet to the internal host only if the internal host had previously sent a packet
to IP address X.
Port Restricted Cone: Similar to a restricted cone NAT, but the restriction includes port
numbers. Specifically, an external host can send a packet, with source IP address X and
source port P, to the internal host only if the internal host had previously sent a packet to IP
address X and port P.
Symmetric: All requests from the same internal IP address and port, to a specific destination
IP address and port, are mapped to the same external IP address and port. If the same host
sends a packet with the same source address and port, but to a different destination, a
different mapping is used. Furthermore, only the external host that receives a packet can send
a UDP packet back to the internal host.
For more details on NAT treatments, refer to RFC 3489.
544
To set the automatic update every time the Mediatrix unit restarts:
1.
Set the configuration scripts parameters as defined in Executing Configuration Scripts Settings on
page 539.
2.
These scripts must be in a directory under the root path.
3.
In the Automatic Script Execution section of the Configuration Scripts page, select Enable in the
Execute on Startup drop-down menu.
Figure 255: Automatically Update Scripts Section
3
The automatic configuration update will be performed each time the Mediatrix unit restarts.
The unit configuration is only updated if at least one parameter value defined in the downloaded
configuration scripts is different from the actual unit configuration.
4.
Automatic Update at Periodic Intervals

You can configure the Mediatrix unit to download new configuration scripts periodically.
To set the automatic update at periodic intervals:

1.
Set the configuration scripts parameters as defined in Executing Configuration Scripts Settings on
page 539.
2.
These scripts must be in a directory under the root path.
3.
In the Automatic Script Execution section of the Configuration Scripts page, select Enable in the
Execute Periodically drop-down menu.
Figure 256: Automatic Script Execution Section
3
5
4
6
4.
Select the time base for configuration updates in the Time Unit drop-down menu.
Table 454: Time Unit Parameters
Parameter
Description
Minutes
Updates the units configuration every x minutes.
Hours
Updates the units configuration every x hours.
Days
Updates the units configuration every x days. You can define the time
range when to perform the update in the Time Range field (see Step 6).
You can specify the x value in the Period field (see Step 5).
5.
Set the waiting period between each configuration update in the Period field.
545
Available values are from 1 to 60. The time unit for the period is specified in the Time Unit field (see
Step 4).
6.
If you have selected Days in Step 4, set the time range when to initiate a configuration update in
the Time Range field.
The format should be one of the following:
hh[:mm[:ss]]
hh[:mm[:ss]] - hh[:mm[:ss]]
hh: Hours.
mm: Minutes
ss: Seconds
where:
The time range is based on the Static Time Zone field of the Network - Host page (see Time
Configuration on page 96 for more details).
You must have a time server SNTP that is accessible and properly configured or the automatic
configuration update feature may not work properly. It is assumed that you know how to configure
your SNTP server. If not, refer to your SNTP servers documentation.
at boot time.
The configuration scripts are downloaded at the specified time or at a random time within the interval
specified and thereafter at the period defined by the Period field. Lets say for instance the automatic
unit configuration update is set with the time range at 11:30 - 15:30 and the update period at every
2 days.
7.
If the automatic update is enabled before 11:30, the first update will take place the
same day between 11:30 - 15:30, the second update two days later within the same
time range, and so on.
If the automatic update is enabled between 11:30 - 15:30, the first update will take
place the day after between 11:30 - 15:30, the second download two days later within
the same time range, and so on.
If the automatic update is enabled after 15:30, the first update will take place the day
after between 11:30 - 15:30, the second download two days later within the same time
range, and so on.
DHCPv4 Auto-Provisioning
Note: This feature does not support IPv6. See IPv4 vs. IPv6 Availability on page 87 for more details.
You can configure the Mediatrix unit to automatically download new configuration scripts upon receiving
options 66 (tftp-server) or 67 (bootfile), or vendor-specific option 43 using sub-options 66 and 67 in a DHCPv4
answer. A DHCP answer includes both Bound and Renew.
The contents of option 66, 67 or 43 defines which script to download. The units configuration is not used to
download the script. This allows the unit, for instance, to download a script from a server after a factory reset
and to reconfigure itself without a specific profile.
The syntax of options 66, 67 and 43 is as follows:
[FileType] = [protocol]://[username] :[password]@[fqdn server]/[path]
For instance:
Script=https://admin :adminpw@script-server.media5.com/Mx3000config/%mac%.cfg
The Mediatrix unit supports only the Script file type for now.
546
The following is an example of a valid option 67 (Bootfile):

Option: (t=67, l=53) Bootfile name = "Script=http://192.168.50.1/digest/
%mac%__2.0.6.84.cfg"
Option: (67) Bootfile name
Length: 53
Value: 5363726970743D687474703A2F2F3139322E31136382E3530...
To set DHCPv4 auto-provisioning:

1.
In the Automatic Script Execution section of the Configuration Scripts page, set the Allow DHCP to
Trigger Scripts Execution drop-down menu with the proper behaviour.
Figure 257: Automatically Update Scripts Section
When enabled, the DHCPv4 options tftp-server (option 66), bootfile (option 67) and vendor-specific
option 43 are used to download a configuration script. If this configuration script is identical to the
last executed script, it will not be run again. The script retry mechanism is not enabled for the
DHCPv4 triggered scripts (see Executing Configuration Scripts Settings on page 539 for more
details).
If options 66, 67 and 43 are received, all scripts are executed independently. The script defined by
the tftp-server (option 66) option is executed first.
If you are using HTTPS to transfer scripts, you must have a time server SNTP that is accessible and
properly configured. It is assumed that you know how to configure your SNTP server. If not, refer to
your SNTP servers documentation.
at boot time.
When a DHCPv4 download script is configured in HTTPS, the script execution is deferred and a 30
seconds timer is started to let enough time for the NTP synchronization.
This timer is independent for each HTTPS script launched. If, for instance, a DHCPv4 answer has
option 66, 67 and 43 configured in HTTPS and if the Update on Restart feature is used, up to 3
minutes can pass before any other operation such as backup, restore, script execution can be
processed.
Once the NTP synchronization is established, the deferred scripts are started immediately one after
the other, ending the timer.
When synchronization is already established, there is no timer, even in HTTPS.
2.
DHCP Option Format

Defines the file server address format of DHCP options 66 and 67. The ScriptsDhcpOptionsFormat variable
can have four values.
FullyQualified: Allows up to 2 DHCP options (66 and/or 67) to specify a string in the format
Script=[protocol]://[username] :[password]@[server]/[path]/[file].
Url: Allows one DHCP option to specify a file or folder URL in the format [protocol]://[username]
:[password]@[server]/[path]/[file]. If the URL ends with a '/', it is further completed with the path
and filename specified in variables ScriptLocation and ScriptGenericFileName.
The following macros can be inserted in the URL and will be replaced by their actual values:
%mac% - the MAC address of the unit.
547
%version% - the MFP version of the unit.
ServerHost: Allow one DHCP option to specify the IP address or FQDN of a file server. Uses
the path and filename specified in variables ScriptLocation and ScriptGenericFileName,
use the transfer protocol, username and password specified in ScriptTransferProtocol,
ScriptTransferUsername and ScriptTransferPassword.
AutoDetect: Allows one DHCP option to specify a script file or folder by automatically detecting
the format of the dhcp Option. A value beginning with "Script=" is considered as
"FullyQualified", A value beginning with "[protocol]://" is considered as a URL. A value that
looks like an IPv4/IPv6 address or domain name is considered as a "ServerHost".
You can configure the parameter by:

using a MIB browser
using the CLI

To set the DHCP Option Format

1.
In the Conf MIB, set the ScriptsDhcpOptionsFormat variable, or
2.

Conf.ScriptsDhcpOptionsFormat = "Value"
Number of Retries

by using the CLI
When using the automatic configuration update (on restart or at a specific interval), the Mediatrix unit may
encounter a problem upon restarting the unit (such as a DHCP server problem) that prevents the update to
succeed. You can define a maximum number of attempts to retry a script transfer until it succeeds when it fails
upon an automatic transfer on restart or automatic periodic transfer. The retries are only attempted if the server
is unreachable. Unreachable port or file not found errors don't trigger the retry mechanism.The time interval
between each retry is 30 seconds.
To set the number of retries:

1.
In the confMIB, set the number of retries in the scriptsTransferRetriesNumber variable.

conf.scriptsTransferRetriesNumber="Value"
where Value may be as follows
-1 means a retry to infinity.
0 means no retry.
The maximum number of retries is 100.
548
Creating a Configuration Script

Configuration scripts are text files that contain command lines interpreted by the Mediatrix unit. Most
commands contained in a script assign values to configuration variables. Script commands can also execute
configuration commands. This configuration script can then be downloaded into the Mediatrix unit as
described in the current chapter.
Writing configuration scripts requires a bit of knowledge about the Mediatrix units configuration variables tree
structure. Each parameter that is accessed via the unit's web interface maps to a variable in the configuration
tree. For detailed information on these mappings, please refer to Appendix D - Web Interface SNMP
Variables Mapping on page 663.
Configuration scripts use the Media5 proprietary scripting language, as described in Appendix B - Scripting
Language on page 649.
Refer to Appendix B - Scripting Language on page 649 for samples of configurations you can use in a
configuration script. The samples include the configuration required to perform a basic call between an ISDN
telephone and an analog telephone. These samples may also be used in the Mediatrix unit Command Line
Interface.
549
550
H A P T E R
45
Configuration Backup/Restore
This chapter describes the configuration backup/restore feature, which allows you to backup (upload) all the
SNMP (MIB) and Web configuration of the Mediatrix unit into a configuration image file located on a remote
server or to the local file system.
Configuration backup download server setup.

Backup/restore configuration parameters.
Standards Supported

Authentication

Configuration Backup Download Server

To backup/restore a configuration image, you may need to setup the following applications on your computer:


When you perform a configuration backup/restore by using the TFTP (Trivial File Transfer Protocol) protocol,
you must install a TFTP server running on the PC designated as the TFTP server host. It is assumed that you
know how to set the TFTP root path. If not, refer to your TFTP servers documentation.

When you use the HTTPS protocol, you need to have a time server SNTP that is accessible and properly
configured. It is assumed that you know how to configure your SNTP server. If not, refer to your SNTP servers
documentation. You can also refer to SNTP Configuration on page 95 for more details on how to configure
the Mediatrix unit for a SNTP server.
at boot time.
549
Chapter 45 - Configuration Backup/Restore

When you to perform a configuration backup/restore by using the HTTP protocol, you must install a HTTP
server running on the PC designated as the server host. It is assumed that you know how to set the root path.
If not, refer to your HTTP servers documentation.

Standards Supported


When you perform a configuration backup/restore that requires authentication or privacy by using the HTTP
over the Transport Layer Security (TLS) protocol (HTTPS), you must install a HTTPS server running on the
PC designated as the server host. It is assumed that you know how to set the root path and SSL/TLS security
configuration. If not, refer to your HTTPS servers documentation.
is not very secure.
Suggested Parameter
Ciphers
Message Digests
Description
RSA
Diffie-Hellman
3DES (168 bits)
SHA-1
ID
550
Name
0x0035
0x0039
0x000a
0x0016
0x002f
0x0033
Image Transfer Cipher Suite Settings


by using the CLI
Parameter
CS1
CS2
Description
suites:
551

Parameter
Description
CS3
To set the image transfer cipher suite configuration parameter:

1.
In the confMIB, locate the imagetransferGroup folder in the imageGroup folder.
2.
Set the image transfer cipher suite configuration in the ImageTransferCipherSuite variable.
conf.ImageTransferCipherSuite="Value"

Value
Meaning
100
CS1
200
CS2
300
CS3
Certificates

552
Backup/Restore Configuration
This section describes how to set the backup/restore configuration parameters and some related files (e.g.
certificates). You can restore this configuration in case the Mediatrix unit loses it for any reason or to clone a
unit with the configuration of another unit. The configuration backup images are in XML format and may be
encrypted or in clear text.
Note: The files under the File service are not included in the backup process. In the same way, the restore
process will not remove any file under the File service.
Please note that you can use a backup file from an older firmware version and use it in a unit with a more
recent firmware version. However, a backup file from a newer firmware version than the one actually in the
unit cannot be used for a restore operation on the unit. For instance, lets say you perform the following
backups:
In firmware v1.1r4.1, you make the backup Backup_X1.

In firmware v1.1r4.2 you make the backup Backup_X2
Application v1.1r4.2 is more recent than v1.1r4.1. The following table describes the various scenarios
possible.
Table 459: Backup Matrix
Scenario
Supported
Not
Supported
Apply the backup Backup_X1 in a unit with firmware v1.1r4.1.

You can backup or restore the configuration to/from two sources:
To/from an image located on an image server

To/from an image located on your PC (transfer images through the web browser)
This section explains both methods.
553
Performing a Backup/Restore to/from on an Image Server or File System

To set the configuration backup/restore parameters:
1.
In the web interface, click the Management link, then the Backup / Restore sub-link.
Figure 258: Management Backup / Restore Web Page
2
3
4
5
6
7
8
9
2.
Set the name of the configuration image in which you want to backup or from which you want to
restore the Mediatrix unit configuration in the File Name field.
The file name is case sensitive hence it must be entered properly. Make sure to write the file
extension.
an image located in the units persistent file system. You can use the Suggestion drop-down menu
to select one of the available images in the file system.
This field may contain a macro that is substituted by the actual value when backing up or restoring
the unit's configuration. The Mediatrix unit supports the %mac% macro, which will be substituted by
the MAC address of the unit. For instance, the %mac%.bkp value for a Mediatrix unit with MAC
address 0090F12345AB will be 0090F12345AB.bkp.
This field may contain macros that are substituted by the actual value when backing up or restoring
the unit's configuration. The supported macros are:
%version% - the MFP version of the unit.
For instance, the %mac%.bkp value for a Mediatrix unit with MAC address 0090F12345AB will
be 0090F12345AB.bkp.
554
3.
Select a transfer protocol to transfer a configuration image in the Transfer Protocol drop-down
menu.
You can select from five different transfer protocols:
EPSV command.
File: Complete path to a configuration image in the Mediatrix units onboard storage
space. You can view and manage all files created with the File transfer protocol by
using the File Manager. See File Manager on page 613 for more details.
Note: The configuration image backup via TFTP can only traverse NATs of types Full Cone or Restricted
Cone. If the NAT you are using is of type Port Restricted Cone or Symmetric, the transfer will not work.
The backup operation currently supports the following protocols:
TFTP
FTP
File
The restore operation supports all the transfer protocols.

mechanism for the configuration image transfer. This mechanism caches the initial configuration
image transfer for later processing, thus preventing changes or update of the original image. This
can cause strange problems if you want to edit a configuration image to modify values and upload
it immediately. The result will still return the original image and not the new one.
4.
Set the configuration backup/restore server hostname or FQDN and IP port in the Host Name field.
This is the current address and port number of the PC that hosts the configuration image file. Use
the special port value 0 to indicate the protocol default. For instance, the TFTP default port is 69
and the HTTP default port is 80.
NAT Variations
NAT treatment of UDP varies among implementations. The four treatments are:
Full Cone: All requests from the same internal IP address and port are mapped to the same
external IP address and port. Furthermore, any external host can send a packet to the internal
host by sending a packet to the mapped external address.
Restricted Cone: All requests from the same internal IP address and port are mapped to the
same external IP address and port. Unlike a full cone NAT, an external host (with IP address
X) can send a packet to the internal host only if the internal host had previously sent a packet
to IP address X.
Port Restricted Cone: Similar to a restricted cone NAT, but the restriction includes port
numbers. Specifically, an external host can send a packet, with source IP address X and
source port P, to the internal host only if the internal host had previously sent a packet to IP
address X and port P.
Symmetric: All requests from the same internal IP address and port, to a specific destination
IP address and port, are mapped to the same external IP address and port. If the same host
sends a packet with the same source address and port, but to a different destination, a
different mapping is used. Furthermore, only the external host that receives a packet can send
a UDP packet back to the internal host.
For more details on NAT treatments, refer to RFC 3489.
555
5.
Set the path of the directory where the configuration image is located in the Location field.
The path is case sensitive hence it must be entered properly. It is relative to the root of the
configuration transfer server. Use the / character when defining the path to indicate subdirectories.
0090f12345ab.xml.
The path differs depending on the transfer protocol selected (see Step 4).
Example: All Transfer Protocols Except File
Lets consider the following example for all protocols except File:
The directory that contains the configuration image is called: Config_Image.

Root Path
c:/root/download
Config_Image
c:/
root/download/Config_Image
c:/root
download/Config_Image
The following are some tips to help your process:
root/download.
If you are using the TFTP protocol to download the software, note that some TFTP
servers on Windows do not recognize the / character and produce an error. In this
case, use the \ character.
Cut and paste the path and/or name of the directory that contains the file into the
configuration download path of the Mediatrix unit (you may have to convert \ into /)
to eliminate typographical errors.
Note that you can define the C:/Root/Download part as you want. The file name may also differ
Example: Transfer Protocol is File
When the Transfer Protocol is set to File, you may prefix the path by one of the following to indicate
storage media:
556
Persistent: for onboard persistent storage. The configuration image is saved into the
persistent file system of the Mediatrix unit (in flash memory). This is the default value.
Volatile: for onboard non-persistent storage. The configuration image is saved into the
non-persistent RAM memory of the Mediatrix unit. All information is lost the next time
the unit restarts.
Table 461: Path Configurations Example (File)
Location
Onboard persistent storage of the

Mediatrix unit under the directory
Backup-1
6.
Persistent:Backup-1
or
Backup-1
If your server requires authentication, set the following:
7.
Define the Backup Content drop-down menu with the information to include in the backup.
Table 462: Backup Content Parameters
Parameter
Description
Config
Only the unit's configuration is included in the backup image.
Config And Certificates The unit's configuration and certificates are included in the backup
image. Media5 strongly recommends to activate encryption when
including certificates in the backup image because host certificates
include the private key (see Steps 9-10).
8.
Set the privacy algorithm in the Privacy Algorithm field.

This defines the encryption method to use for backup operations. This parameter is not used for
restore operations.
You can secure the exchange of configuration image between the server and the Mediatrix unit. A
privacy key allows the unit to decrypt a previously encrypted configuration image. During a restore
of the backup image, the encryption is auto-detected.
The configuration image must have been encrypted before use.
Table 463: Privacy Algorithm
Parameter
None
Description
Backup images are not encrypted.
DefaultAlgo Backup images are encrypted with the default algorithm.

9.
Set the decryption key in the Privacy key field.
Caution: The Privacy key field is not accessible if you have the User or Observer access right. See Users
This is the key used for:
backup operations to encrypt backup images
restore operations to decrypt backup images when encrypted (encryption is autodetected).
557
The key is encoded in hexadecimal notation. You can thus use characters in the range 0-9, A-F. All
other characters are not supported.
Each character encodes 4 bits of the key. For instance, a 32-bit key requires 8 characters.
If you enter too many bits, the key is truncated to the first 448 bits.
If you do not enter enough bits, the key is padded with zeros.
This key must match the key used for the encryption of the configuration image. If the variable is
empty, the configuration image is not decrypted.
10.
To save your settings without performing a backup/restore, click Submit.
To save your settings and perform a backup now, click Submit & Backup Now.
To save your settings and perform a restore now, click Submit & Restore Now.
Transferring Images Through the Web Browser

This section describes how to perform a backup to or restore from an image located on the PC.
To see the content of the units file system persistent memory, go to the File Manager (Chapter 51 - File
Manager on page 613). All installed configuration scripts/images are listed.
To perform a backup to or restore from an image on the PC:

1.
In the Transfer Images Through Web Browser section of the Backup / Restore page, type the name
of an image in the Upload Parameters field or select an existing one on the PC with the Browse
button.
If you are currently using an unsecure HTTP access, the Transfer Images Through Web Browser
section is disabled. This is to avoid transferring an image in clear text. To enable the section, access
the secure site by clicking the Activate unsecure image transfer through web browser link at the top
of the window.
When an image is run, it is not installed in the units file system persistent memory. You can click
the Clear Selection link to empty the field and enter another name.
Figure 259: Transfer Images Through Web Browser Section
3
1
2
2.
Set the decryption key in the Privacy key field.

This is the key used for:
backup operations to encrypt backup images
restore operations to decrypt backup images when encrypted (encryption is autodetected).
The key is encoded in hexadecimal notation. You can thus use characters in the range 0-9, A-F. All
other characters are not supported.
Each character encodes 4 bits of the key. For instance, a 32-bit key requires 8 characters.
If you enter too many bits, the key is truncated to the first 448 bits.
If you do not enter enough bits, the key is padded with zeros.
This key must match the key used for the encryption of the configuration image. If the variable is
empty, the configuration image is not decrypted.
3.
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Click the Upload Now button.
H A P T E R
46
Firmware Download
This chapter describes how to install, uninstall and update software components on the Mediatrix unit by using
the web interface, according to a supplied Firmware Pack selection.
Note: If you have backed up the configuration of your unit, Media5 recommends that you perform a new
backup every time you upgrade the firmware pack of the unit to avoid restore issues.
What is a firmware pack?

Firmware pack server setup.
Firmware pack version and name to download.
Transfer parameters.
Firmware pack update procedure.
Standards Supported

Authentication

What is a Firmware Pack?

A firmware pack file is a regular zip file that contains the modules and features to install on the Mediatrix unit.
When unzipping a firmware pack, the contents is extracted according to a pre-defined tree architecture. This
creates a directory that contains the files required for the Mediatrix unit to properly update its firmware. The
firmware pack contains all the modules to install. When performing the upgrade operation, the Mediatrix unit
checks the modules versions of the firmware pack against its own modules versions and installs only the
modules that have changed.
Note: The currently installed firmware pack is only required when downgrading.
Before Performing a Firmware Upgrade or Downgrade

To download a firmware pack, you may need to setup the following applications on your computer:

MIB browser (with the current Mediatrix unit MIB tree)
Firmware pack zip file
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Chapter 46 - Firmware Download

Syslog daemon (optional)

When you perform a firmware pack update by using the TFTP Trivial File Transfer Protocol) protocol, you must
install a TFTP (server running on the PC designated as the update files server. This PC must not have a
firewall running. Media5 also recommends to place the PC and the Mediatrix unit in the same subnet.
It is assumed that you know how to set the TFTP root path. If not, refer to your TFTP servers documentation.

When you use the HTTPS protocol, you need to have a time server SNTP that is accessible and properly
configured. It is assumed that you know how to configure your SNTP server. If not, refer to your SNTP servers
documentation. You can also refer to SNTP Configuration on page 95 for more details on how to configure
the Mediatrix unit for a SNTP server.
at boot time.

When you perform a firmware pack update by using the HTTP protocol, you must install a HTTP server running
on the PC designated as the update files server. This PC must not have a firewall running. Media5 also
recommends to place the PC and the Mediatrix unit in the same subnet.
It is assumed that you know how to set the root path. If not, refer to your HTTP servers documentation.

Standards Supported


When you perform a firmware pack update that requires authentication or privacy by using the HTTP over the
Transport Layer Security (TLS) protocol (HTTPS), you must install a HTTPS server running on the PC
designated as the update files server. It is assumed that you know how to set the root path and set the SSL/
TLS security configuration. If not, refer to your HTTPS servers documentation.
is not very secure.
560
Suggested Parameter
Ciphers
Message Digests
Description
RSA
Diffie-Hellman
3DES (168 bits)
SHA-1
ID
Name
0x0035
0x0039
0x000a
0x0016
0x002f
0x0033
Mediatrix Firmware Pack HTTPS Transfer Cipher Suite Settings


by using the CLI
561
Parameter
CS1
CS2
562
Description
This represents a secure configuration using SHA-1.This value includes the following cipher
suites:

Parameter
CS3
Description
To set the HTTPS transfer cipher suite configuration parameter:

1.
In the fpuMIB, locate the mfptransferGroup folder in the mfpRepositoryGroup folder.
2.
Set the Mediatrix firmware pack HTTPS transfer cipher suite configuration in the
MfpTransferCipherSuite variable.
fpu.MfpTransferCipherSuite="Value"

Value
Meaning
100
CS1
200
CS2
300
CS3
Certificates

Caution: You must have a time server SNTP that is accessible and properly configured. It is assumed that
you know how to configure your SNTP server. If not, refer to your SNTP servers documentation. You can
also refer to SNTP Configuration on page 95 for more details on how to configure the Mediatrix unit SNTP
client.
563
Firmware Packs Configuration

This section allows you to define the firmware pack version and name(s) to properly download them.
To set the firmware pack parameters:

1.
In the web interface, click the Management link, then the Firmware Download sub-link.
Figure 260: Management Firmware Download Web Page
3
4
5
6
2.
In the Firmware Packs Installed section of the Firmware Upgrade page, click one of the available
buttons if required:
Table 468: Available Buttons
Button
Description
Factory Reset You can apply a factory reset to the current unit by clicking the Factory Reset
button. See Factory Reset on page 16 for more details.
Rollback
You can revert back to the previously installed MFP found in the recovery bank
at any time by clicking the Rollback button. If the recovery bank contains a
MFP that can be used, it is displayed in the Bank column of the Firmware
Packs Installed section. When a rollback is performed, the configuration of the
MFP in the recovery bank applies. The current configuration is lost. The
Rollback button is displayed only if the current bank's application and the
recovery bank's application both support the rollback mechanism and have
been both installed from an application supporting the rollback. Otherwise, the
Rollback button is not displayed.
Note: This feature does not apply to the Mediatrix 4102S model.
3.
In the Firmware Packs Configuration section of the Firmware Upgrade page, enter the version of
the firmware pack to install in the Version field.
Currently, you cannot install two firmware packs with different versions.
564
4.
Set the Automatic Restart Enable drop-down menu with whether or not to automatically restart the
system when needed for completing a firmware update operation.
You can also set a grace delay in the next step.
5.
If automatic restart is enabled, set the Automatic Restart Grace Delay field with the grace delay, in
minutes, that the unit waits for all telephony calls to be terminated before the automatic restart can
occur.
The maximum value is set to 10080 minutes (7 days).
During that delay, it is impossible to make new calls but calls in progress are not terminated. When
all calls are completed, then the unit restarts.
You can also set a services restart grace period as described in Graceful Restart of Services on
page 58.
6.
Enter the name of up to five firmware packs to install in the Firmware Pack fields.
You can install several firmware packs at the same time. In that case, enter the firmware pack
names in different rows of the table.
When extracting the content of the ZIP file, available firmware packs are listed as directories under
the Mediatrix/FirmwarePacks directory.
Note: The Language drop-down menu currently supports only English.

7.
Proceed to Transfer Configuration on page 566.
565
Transfer Configuration
The following describes how to configure the transfer parameters required to perform a firmware update.
To setup the firmware download path:

1.
In the Transfer Configuration section of the Firmware Upgrade page, select a transfer protocol to
transfer a firmware pack in the Transfer Protocol drop-down menu.
Figure 261: Transfer Configuration Section
1
3
2
4
EPSV command.
mechanism for the firmware pack download.
2.
Set the static update files server IP address or domain name and port number to use when
downloading a firmware pack in the Host Name field.
This is the current address and port number of the PC that hosts the firmware packs. Use the special
port value 0 to indicate the protocol default. For instance, the TFTP default port is 69, the HTTP
default port is 80, and the HTTPS default port is 443.
This parameter is not required if you have selected the File transfer protocol.
3.
Set the firmware download path in the Location field.

This is the location of the folder that contains the modules to download into the Mediatrix unit. In
other words, this is where the zip file containing the firmware pack has been extracted. This path is
relative to the root of the external media and excludes.
Lets consider the following example:
The directory that contains the files required for download is called:

Root Path
c:/root/download
N/A
c:/
root/download
c:/root
download
The following are some tips to help your download process:
566
root/download.
If you are using the TFTP protocol, note that some TFTP servers on Windows do not
recognize the / character and produce an error. In this case, use the \ character.
Cut and paste the path and/or name of the directory that contains the extracted files
into the firmware download path of the Mediatrix unit (you may have to convert \ into /
) to eliminate typographical errors.
Note that you can define the C:/Root/Download part as you want. The file names may also differ
4.
If your server requires authentication when downloading a firmware pack, set the following:
5.
Proceed to Firmware Pack Update Procedure on page 568.
Certificate Validation

by using the CLI
When downloading a MFP from an HTTPS server, you can define the level of security to use when validating
the server's certificate.
Table 470: Certificate Validation Parameters
Parameter
Description
NoValidation Allow a connection to the server without validating its certificate. The only condition is to
receive a certificate from the server. This option provides partial security and should be
selected with care.
HostName
Allow a connection to the server by validating its certificate is trusted and valid. The
validations performed on the certificate include the expiration date and that the Subject
Alternate Name (SAN) or Common Name (CN) matches the FQDN or IP address of the
server.
To set the certificate validation parameter:

1.
In the fpuMIB, set the certificate validation behaviour in the MfpTransferCertificateValidation

variable.
fpu.MfpTransferCertificateValidation="Value"

Table 471: Certificate Validation Values
Value
Meaning
100
NoValidation
200
HostName
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Firmware Pack Update Procedure
Default Settings On Install


by using the CLI
You can define if the unit is to automatically execute a factory reset upon completion of a firmware installation.
Table 472: Default Settings On Install Parameters
Parameter
Description
Disable
The unit does not execute a factory reset upon completion of a firmware installation.
Enable
The unit automatically executes a factory reset upon completion of a firmware installation.
To set the default setting on install parameter:

1.
In the fpuMIB, set the default setting on install in the defaultSettingsOnInstall variable.
fpu.defaultSettingsOnInstall="Value"
where Value may be Disable (default) or Enable.
Firmware Pack Update Procedure

The following describes how to update the firmware pack of the Mediatrix unit.
Extracting the Firmware Pack Zip File

The firmware pack zip file contains the firmware information required for the update. Extract the contents of
the zip file on the PC designated as the update files server without modifying the defined folder name. This
creates a directory that contains the files required for the Mediatrix unit to properly update its firmware.
You must extract the zip file under the root path as defined in the update files server or the firmware pack
update will not proceed.
Launching a Firmware Pack Update

The following describes how to launch a firmware pack update.
To launch the firmware pack update:

1.
If not already done, set the firmware packs parameters as defined in Firmware Packs
2.
If not already done, unzip the firmware pack file as described in Extracting the Firmware Pack Zip
File on page 568.
3.
If not already done, set the transfer configuration parameters as described in Transfer
4.
To save your settings without performing a firmware update, click Submit.
To save your settings and perform a firmware update now, click Submit & Install Now.
The firmware pack update may take several minutes, depending on your Internet
568
Firmware Pack Update Status
connection, network conditions and servers conditions.
Caution: Media5 recommends to close and re-open your Web browser after a reboot that installs a
firmware update. This is because your browser may activate a caching mechanism for some files. This
mechanism caches some of the files to improve performance. This may cause problems when the cached
files change in the Mediatrix unit after a firmware update and the web pages are no longer compatible with
the cached files.
Firmware Pack Downgrade

It is possible to downgrade a Mediatrix unit from the current version to an older version. The procedure is the
same as with a firmware upgrade.
Firmware Pack Update Status

When the Mediatrix unit initiates a firmware pack update, the LEDs indicate the status of the process.
Table 473: LED States in Firmware Pack Update
Event
Firmware pack downloading
and writing
LED State
All LEDs are cycling from left to right, individually blinking 1 Hz, 33%
duty.
Warning: Do not turn the Mediatrix unit off while in this state.
Firmware pack download failed
All LEDs are blinking at 3 Hz, 50% duty. One LED out of two has a 180
degree phase. This pattern lasts for 8 seconds.
You can also view the firmware pack update status in the Status section of the Firmware Upgrade page.
Note: When the firmware pack update fails, the Mediatrix unit tries to download the firmware three times.
In some cases, the unit may also restart.
Spanning Tree Protocol (STP)

Many network switches use the Spanning Tree Protocol (STP) to manage Ethernet ports activity. When a
firmware pack update occurs, the Ethernet connector of the Mediatrix unit may switch off. This shutdown may
trigger these network switches to shutdown the matching Ethernet port for at least one minute. This shutdown
on the switch side can prevent firmware pack update.
To prevent this, the Mediatrix unit supports the STP. However, this management has a potential time cost. It
may appear from time to time that firmware pack updates take more time. This is normal.
When using the unit, Media5 recommends to disable the Spanning Tree Protocol on the network to which the
unit is connected.
569
570
Spanning Tree Protocol (STP)
H A P T E R
47
Certificates Management
This chapter describes how to transfer and manage certificates into the Mediatrix unit.
Standards Supported

Introduction
The Mediatrix unit uses digital certificates, which are a collection of data used to verify the identity of the holder
or sender of the certificate.
The certificates contain the following information:
certificate name
issuer and issued to names
Validity period (the certificate is not valid before or after this period)
Usage of the certificate (Identifies in which role or context a certificate can be used by the host
it authenticates).
TlsClient: The certificate identifies a TLS client. A host authenticated by this kind of
certificate can act as a client in a SIP over TLS connection when mutual authentication
is required by the server.
TlsServer: The certificate identifies a TLS server. A host authenticated by this kind of
certificate can serve files or web pages using the HTTPS protocol or can act as a
server in a SIP over TLS connection.
whether or not the certificate is owned by a CA (Certification Authority)
The Mediatrix unit uses two types of certificates:

Table 474: Certificates Types
Type
Description
Host
Certificates used to certify the unit (e.g.: a web server with HTTPS requires a host
certificate).
Others
Any other certificate including trusted CA certificates used to certify peers (e.g.: a SIP
server with TLS).
The transferred certificate must be in Privacy Enhanced Mail (PEM) (host or others) or Distinguished Encoding
Rules (DER) (others) format. When transferring a host certificate, the certificate must be appended to the
private key to form one PEM file. The private key must not be encrypted.
You can transfer a certificate by using the HTTP or HTTPS protocol, but Media5 recommends to use HTTPS.
To access the unit via HTTPS, your browser must support RFC 2246 (TLS 1.0). The latest version of Microsoft
Internet Explorer supports HTTPS browsing.
569
Chapter 47 - Certificates Management
Introduction
HTTPS Transfer Cipher Suite Settings


by using the CLI
Parameter
CS1
CS2
570
Description
being introduced. This should be changed if additional network security is required. Thie Web
server only accepts the following cipher suites:
This represents a secure configuration using SHA-1. The Web server only accepts requests
using cipher suites:
Managing Certificates

Parameter
CS3
Description
TLS_RSA_WITH_AES_128_GCM_SHA256 TLS_RSA_WITH_AES_128_CBC_SHA256

1.
In the cert MIB, locate the transferGroup folder.
2.
Set the HTTPS transfer cipher suite configuration in the TransferHttpsCipherSuite variable.
cert.TransferHttpsCipherSuite="Value"

Value
Meaning
100
CS1
200
CS2
300
CS3
Managing Certificates
You can view certificates information and you can delete certificates.
571
Certificate Authorities
To view and manage certificates:

1.
In the web interface, click the Management link, then the Certificates sub-link.
Figure 262: Management Certificates Information Web Page
2
3
The Host Certificates section contains the certificates used to certify the unit. The Others
Certificates section contains any other certificate used to certify peers.
2.
If applicable, delete a certificate in the Host Certificates or Others Certificates sections by clicking
the
button of the certificate you want to delete.
3.
If applicable, delete a certificate in the Other Certificates section by clicking the

certificate you want to delete.
4.
button of the
Certificate Authorities
This section contains information specific to certificate authority (CA) files.
To view and manage certificate authorities information:

1.
In the Certificate Authorities section of the Certificates page, define a specific OCSP URL to use for
certificate revocation status of certificates issued by this certificate authority (CA) in the
corresponding Override OCSP URL field.
Figure 263: Certificate Authorities Section
2
572
Certificate Upload through the Web Browser
The URL should follow one of these formats:

http://hostname[:port]
http://hostname/path/filename.xxx
Note: The default empty value means that the OCSP URL present in the certificate to verify will be used
for checking its revocation status.
2.
Certificate Upload through the Web Browser

The following steps explain how to transfer (add) a certificate from the web interface.
To upload a certificate:
1.
If you are currently using an unsecure HTTP access, the Certificate Upload Through Web Browser
section is disabled. This is to avoid transferring a certificate in clear text. To enable the section,
access the secure site by clicking the Activate unsecure certificate transfer through web browser
link at the top of the window.
2.
In the Certificate Upload Through Web Browser section of the Certificates page, select the type of
the certificate in the Type drop-down menu.
Before transferring the certificate, you must indicate whether this is a Host or Others certificate.
Figure 264: Certificate Upload Through Web Browser Section
2
3.
Use the Browse button to select the certificate to transfer.

The maximum certificate name is 50 characters.
4.
Initiate the certificate upload by clicking the Upload Now button.

The Mediatrix unit immediately transfers the certificate. Once the certificate is transferred, you must
restart the SipEp and Web services in the System > Services page (Chapter 4 - Services on
page 55) before using the newly transferred certificate. Click the link in the message that is
displayed to access the Services web page.
Transferring a Certificate via Configuration Script


by using the CLI
You can use a special command to transfer a certificate by configuration script or CLI. This command has the
following parameters: URL of the certificate to download, its Type (Host/ Others), the username and password.
See Appendix B - Scripting Language on page 649 for more details on the Media5 proprietary scripting
language
To transfer a certificate via configuration script:

1.
Use the following line in the CLI or a configuration script:

Cert.DownloadCertificate FileUrl=Value UserName=Value Password=Value Type=Value
573
Host Certificate Associations
where the different values may be as follows:

Table 477: Certificate Transfer Values
Value
FileUrl
Description
URL to a Certificate file that is loaded upon executing the execution of Download
command. The transfer protocols supported are:
HTTP
HTTPS
TFTP
FTP
Examples of valid URLS:
http://www.myserver.com/Cert_MxDefault001.der
tftp://myserver.com:69/myfolder/Cert_MxDefault001.der
When the port is not included in the URL, the default port for the chosen protocol is
used.
This field may contain some macros that are substituted by the actual value at the
moment of fetching the configuration script. The supported macros are:
UserName
When authentication is required by the remote file server, this variable is used as
the username.
Password
When authentication is required by the remote file server, this variable is used as
the password.
Type
Type of certificate to transfer.
Host: Certificate used to certify the host system.
Other: Remote systems certificates and issuers certificates.
For instance, a valid command would be:

Cert.DownloadCertificate FileUrl=http://www.myserver.com/Cert_MxDefault001.der
UserName=MyName Password=MyPassword Type=Host

The Host Certificate Associations section allows you to define which services can use the host certificates.
To set host certificate associations:

1.
In the Host Certificate Associations section of the Certificates page, check the services that can use
a given host certificate.
Figure 265: Host Certificate Associations Section
Table 478: Host Certificate Associations Parameters

Parameter
574
Description
SIP
Specifies if this certificate can be used for SIP security.
Web
Specifies if this certificate can be used for Web security.
Table 478: Host Certificate Associations Parameters (Continued)

Parameter
2.
Description
EAP
Specifies if this certificate can be used for EAP security.
Conf
Specifies if this certificate can be used for Conf security.
Fpu
Specifies if this certificate can be used for Fpu security.
File
Specifies if this certificate can be used for File security.
Cert
Specifies if this certificate can be used for Cert security.
575
576
H A P T E R
48
SNMP Configuration
This chapter describes how to set the SNMP parameters of the Mediatrix unit.
Standards Supported
RFC 1157: Simple Network Management Protocol (SNMP)
RFC 1910: User-based Security Model for SNMPv2
RFC 2104: HMAC: Keyed-Hashing for Message

Authentication
RFC 2576: Coexistence between Version 1, Version 2, and

Version 3 of the Internet-standard Network Management
Framework
RFC 2741: Agent Extensibility (AgentX) Protocol Version 1
RFC 3411: An Architecture for Describing Simple Network

Management Protocol (SNMP) Management Frameworks
RFC 3412: Message Processing and Dispatching for the

RFC 3413: Simple Network Management Protocol (SNMP)

Applications
RFC 3414: User-based Security Model (USM) for version 3 of

the Simple Network Management Protocol (SNMPv3)
RFC 3415: View-based Access Control Model (VACM) for the

Simple Network Management Protocol (SNMP)a
RFC 3416: Version 2 of the Protocol Operations for the

RFC 3417: Transport Mappings for the Simple Network

Management Protocol (SNMP)
RFC 3826: The Advanced Encryption Standard (AES) Cipher

Algorithm in the SNMP User-based Security Model
a. The Mediatrix unit complies to RFC 3415 but does not support the related MIBs.
Introduction
All parameters available in the Mediatrix unit web interface may also be configured via SNMP. The Mediatrix
unit SNMP feature offers the following options:
Remote management
The Mediatrix unit SNMP feature allows you to configure all the MIB services by using a SNMP browser to
contact the MIBs of the Mediatrix unit. It is assumed that you have basic knowledge of TCP/IP network
administration.
Note: The Mediatrix units SNMP settings do not support IPv6. See IPv4 vs. IPv6 Availability on page 87
for more details.
You can use the MIB browser built in the Media5 Unit Manager Network.
575
Chapter 48 - SNMP Configuration
SNMP Configuration Section
Youcanalsouseanythird-partySNMPbrowserornetworkmanagementapplicationrunningtheSNMPprotocoltomonitorand
configuretheMediatrixunit.However,theinformationmaynotbepresentedinthesamemannerdependingontheSNMPbrowser
used.
Locate the proper parameter to modify and change (SET) its value.

The SNMP Configuration section allows you to configure the SNMPv3 privacy information that allows securing
the Mediatrix unit, as well as defining where the Mediatrix unit must send traps.
To set SNMP parameters:

1.
In the web interface, click the Management link, then the Snmp sub-link.
Figure 266: Management Snmp Web Page
2
3
4
6
5
7
8
9
2.
Set the SNMP Listening Port field with the port number on which the SNMP service listens for
incoming SNMP requests.
The default value is 161.
3.
Specify with which SNMP version a user can connect to the system by setting one of the following
drop-down menus to enable:
Table 479: SNMP Versions
SNMP Version Drop-down menu to set to Enable
SNMPv1
Enable SNMP V1
SNMPv2
Enable SNMP V2
SNMPv3
Enable SNMP V3
By default, SNMPv3 is enabled.
Caution: It is possible to disable all three versions of SNMP on the Mediatrix unit. If you do so, you will no
longer be able to access the unit in SNMP. To recover from this situation, you must perform a factory reset
procedure.
576
Note: Please note that a public user might be granted (unsecure) access by using SNMPv1 or SNMPv2,
while an admin user should rather be granted a SNMPv3 access. Furthermore, access for users in
SNMPv3 will require authentication and could be done with or without privacy according to the units
configuration. This means that the unit does not grant an SNMPv3 access without authentication and
privacy.
4.
If SNMPv3 is enabled, set the Authentication Protocol drop-down menu with the authentication
protocol to use with SNMPv3.
Table 480: SNMP Authentication Protocol
Protocol
Description
MD5
MD5 encoding is used. This is the default value.
SHA1
SHA1 encoding is used.
Caution: The Authentication Protocol field is not accessible if you have the User or Observer access right.
See Users on page 607 for more details.
SNMPv3 will grant access to all users who are configured in the unit and have a password with 8
characters or more (in the AAA service as described in Chapter 50 - Access Control Configuration
on page 607).
5.
If SNMPv3 is enabled, set the privacy protocol to use with SNMPv3 in the Privacy Protocol dropdown menu.
Table 481: SNMP Privacy Protocol
Protocol
Description
None
No encryption is used. The Privacy Password parameter is ignored. This is the

default value.
DES
DES encryption is used.
Caution: The Privacy Protocol field is not accessible if you have the User or Observer access right. See
Users on page 607 for more details.
6.
If you are using the DES privacy, set the password to use in the Privacy Password field.
Caution: The Privacy Password field is not accessible if you have the User or Observer access right. See
7.
Set the Community field with the string to use for the community field of SNMPv1 and SNMPv2
read-write commands and traps.
This field must not be empty.
The use of a community name provides context for agents receiving requests and initiating traps.
An SNMP agent won't respond to a request from a management system outside its configured
community.
The community name field may influence the AAA user name that will be used by the Mediatrix for
non-authenticated SNMP access (SNMPv1 and SNMPv2). See Additional SNMP Parameters on
page 579 for more information.
8.
Specify that traps can be sent by setting the Enable SNMP Traps drop-down menu to enable.
There are five conditions that the Mediatrix unit checks before sending a trap:
The traps are enabled.
The destination address is valid.

577
The NetSnmp Agent is ready.
The destination address is reachable according to the routing table.
The appropriate physical link is up.
If all of those conditions are true, then the Mediatrix unit sends the traps. If any of those conditions
is false, the Mediatrix unit waits (1 second) and retries until it succeeds. Even if the traps are
delayed, they will be sent with the appropriate timestamp when all the conditions are met.
Furthermore, the SNMP version(s) currently enabled (see Step 2 for more details) define which type
of trap may be sent.
Table 482: Trap Type Sent vs SNMP Version Enabled
SNMP Version Enabled
SNMPv1 SNMPv2 SNMPv3
Trap Sent
Trap V1
Trap V2c
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Enabled
Note: You can also enable the traps via the CLI. See Chapter 2 - Command Line Interface (CLI) on
page 19 for details on how to work with the CLI.
The Mediatrix unit handles five different types of trap:
Table 483: Trap Types
Trap
coldStart
Description
A coldStart(0) trap means that the sending protocol entity is reinitializing
itself so that the agent's configuration or the protocol entity
implementation may be altered.
This trap is sent prior to a reboot that follows a firmware update, a
backup restoration or a default settings application. Note that if the unit is
shut down unexpectedly (power failure, power switch), this trap is not
emitted.
When the unit reboots because of a firmware upgrade, no coldStart traps
are sent before this reboot. In that specific case, a coldStart trap is sent
after the reboot if the installation scripts succeeded.
warmStart
A warmStart(1) trap means that the sending protocol entity is reinitializing

itself so that neither the agent configuration nor the protocol entity
implementation is altered.
This trap is sent prior to all other reboots. Note that if the unit is shut
down unexpectedly (power failure, power switch), this trap is not emitted.
When the unit reboots because of a firmware upgrade, no warmStart
traps are sent before this reboot. In that specific case, a warmStart trap is
sent after the reboot if the installation scripts failed.
578
Table 483: Trap Types (Continued)

Trap
linkDown
Description
A linkDown(2) trap means that the SNMPv2 entity acting in an agent role
has detected that the ifOperStatus object for one of its communication
links is about to enter the down state from some other state. This other
state is indicated by the included value of ifOperStatus.
The Trap-PDU of type linkDown includes ifIndex, ifAdminStatus,
ifOperStatus (as of RFC 2233) of the interface that generated the trap.
linkUp
A linkUp(3) trap means that the SNMPv2 entity, acting in an agent role,
has detected that the ifOperStatus object for one of its communication
links left the down state and transitioned into some other state (but not
into the notPresent state). This other state is indicated by the included
value of ifOperStatus.
The Trap-PDU of type linkUp includes ifIndex, ifAdminStatus,
ifOperStatus (as of RFC 2233) of the interface that generated the trap.
authenticationFailure An authenticationFailure(4) trap means that the sending protocol entity is

the addressee of a protocol message that is not properly authenticated.
This trap is sent when an authentication failure occurs from the Web, CLI
or SNMP interface.
9.
If the traps are enabled, set the Trap Destination (s) field with the addresses/FQDNs and ports
where to send traps.
You can specify up to 5 destinations by using a comma between them (comma is not authorized
within a FQDN). The port numbers are optional. Note that the traps are sent simultaneously to all
destinations.
Example:
trapdest.com:2345, 123.45.67.89

10.
Additional SNMP Parameters


by using the CLI
A user name can be added to be used by the SNMP v1/v2 to access the configuration.
For non-authenticated access (SNMPv1 and SNMPv2), the Mediatrix will use the AAA user name from the
SnmpUser variable if it is not empty. If empty, the community name is used as the AAA user name.
To add an SNMP user name:

1.
In the snmpMIB, set the SnmpUser variable to a valid AAA user name.
snmp.SnmpUser="Value"
where Value is a valid AAA user name.
Caution: If the provided SNMP user name does not exist in the AAA.UsersStatus table or if the SNMP
user name is empty and the community name does not exist in the AAA.UsersStatus table, the SNMP
access will fail.
579
SNMP Statistics
Partial Reset
When a partial reset is triggered, the following parameters are affected:
Listening Port: Default value 161.

Enable SNMPv1: Default value disable.
Enable SNMPv2: Default value disable.
Enable SNMPv3: Default value enable.
See Partial Reset on page 15 for more details.
SNMP Statistics
The following are the statistics the Mediatrix unit keeps.
Table 484: SNMP Statistics
MIB Variable
580
statsGetRequest
Number of GET requests handled by the service.
statsGetNextRequest
Number of GET-NEXT requests handled by the service.
statsSetRequest
Number of SET requests handled by the service.
H A P T E R
49
CWMP Configuration
This chapter describes how to set the CWMP parameters of the Mediatrix unit. The CPE WAN Management
Protocol (Cwmp) service is an optional feature that may be added to a Mediatrix unit at purchase time. For
more details, contact your sales representative.
Standards Supported
TR-069: CPE WAN Management Protocol v1.1 (Version:

Issue 1 Amendment 2)a b
TR-098: Internet Gateway Device Data Model for TR-069

(Version: Issue 1 Amendment 2)c
TR-104: Provisioning Parameters for VoIP CPEd
TR-106: Data Model Template for TR-069-Enabled Devices

(Version: Issue 1 Amendment 2)e
TR-111: Applying TR-069 to Remote Management of Home

Networking Devicesf
a. Not compliant with:

- Signed Vouchers (Appendix C).
- Web Identity Management as (Appendix D).
- Signed Package Format (Appendix E).
- Connection Request via NAT Gateway (Appendix G).
b. Partially compliant with:
- Device-Gateway Association (Appendix F).
c. Partially compliant
d. Supports a single VoiceProfile in the VoiceService :1 object.
e. Supports:
- baseline:1
- GatewayInfo:1
- "Device.LAN:1
- "Device.Time:1 (except NtpServer2).
f. Supports TR-111 Part 2
- UDPConnectionRequestAddress
- STUNEnable
- STUNServerAddress
- STUNServerPort
- NATDetected
- STUNMinimumKeepAlivePeriod
Introduction
The CPE WAN Management Protocol service allows configuring the Mediatrix unit via the TR-069 protocol.
The supported data models are a subset of Internet Gateway Device (TR-098) and Device (TR-106) and
includes Mediatrix proprietary data models.
Using TR-069, the Mediatrix unit can get in contact with an Auto Configuration Server (ACS) to initiate a
configuration script transfer/execution and a firmware upgrade. The periodic informs are also supported to
allow the unit to contact the ACS when behind a NAT.
581
Chapter 49 - CWMP Configuration
Introduction
See ACS Access to the Local Log Table on page 597 for the list of supported methods and parameters.
Note: The Mediatrix units CWMP settings do not support IPv6. See IPv4 vs. IPv6 Availability on page 87
for more details.
Licence Key Activation of TR-069

You can enable the TR-069 feature on units that are already deployed via an encrypted licence key. See
Activate Licence on page 617 for more details.
582
CWMP Configuration Section

The CWMP Configuration section allows you to configure the CWMP general parameters, as well as the ACS
and Periodic Inform parameters.
The General Configuration part allows you to set the basic CWMP parameters.
To set general CWMP parameters:

1.
In the web interface, click the Management link, then the CWMP sub-link.
Figure 267: Management CWMP Web Page
2
3
2.
In the General Configuration part, set the Root Element drop-down menu with the data model used
for the configuration.
Table 485: Root Element Parameters
Parameter
Device
Description
The Mediatrix unit uses the Device data model as defined in TR-106.
Internet Gateway Device The Mediatrix unit uses the Internet Gateway Device data model as
defined in TR-098.
3.
Set the Listening Port field with the port number on which the unit listens for incoming CPE WAN
Management Protocol connections.
If you set the field to 0, the Mediatrix unit uses the default CPE WAN Management Protocol port
7547.
4.
583
ACS Configuration
The ACS Configuration part allows you to set how to contact the ACS server.
The ACS Status field displays the status of the connection with the ACS.
Table 486: ACS Status
Status
Description
Starting
Cwmp service is starting and no connection attempt have been

made so far.
Connected
The Cwmp service is connected to the ACS.
NoUrl
The Cwmp service cannot connect with the ACS because no ACS
URL is configured.
ErrorCannotResolve
The Cwmp service cannot connect with the ACS because the FQDN
cannot be resolved.
ErrorNotResponding
The Cwmp service cannot connect with the ACS because the ACS is
not responding.
ErrorAuthFailure
The Cwmp service failed authenticating to the ACS.
ErrorOther
The Cwmp service cannot connect to the ACS for unspecified

reason. See device and ACS logs.
To define the ACS parameters:

1.
In the ACS Configuration part, set the ACS URL Config Source field with the method to obtain the
URL of the ACS.
Table 487: ACS URL Config Source Parameters
Parameter
Description
DHCP
Uses the DHCP protocol to find the URL as defined in TR-106

section 3.1.
Static
Uses the URL defined in the ACS Static URL field (Step 2).
The Mediatrix unit uses the URL obtained via DHCP if ACS URL Config Source field is set to DHCP.
When the ACS URL Config Source field is set to DHCP but no ACS URL is returned, the unit will
re-attempt using the Static URL.
584
Figure 268: Management CWMP Web Page, ACS Part
1
2
3
2.
If the configuration source is set to Static, set the ACS Static URL field with the URL used by the
unit to connect to the ACS.
This parameter must be a valid HTTP or HTTPS URL.
Example:
http://somewhere.com
https://somewhere.secure.com
3.
Set the Username field with the username used to authenticate the unit when making a connection
to the ACS.
Caution: The Username field is not accessible if you have the User or Observer access right. See Users
4.
Set the Password field with the password used to authenticate the unit when making a connection
to the ACS.
Caution: The Password field is not accessible if you have the User or Observer access right. See Users
5.
ACS Configuration Parameters


by using the CLI
You can define a username and password to authenticate an ACS making a connection request to the CPE.
585
To set the ACS connection request username and password:

1.
2.
In the cwmpMIB, locate the AcsGroup folder.

Define the username to authenticate an ACS making a connection request to the CPE in the
ConnectionRequestUsername variable.
cwmp.ConnectionRequestUsername="Value"
The default Value is admin.

3.
Define the password to authenticate an ACS making a connection request to the CPE in the
ConnectionRequestPassword variable. This variable contains a secret value.
cwmp.ConnectionRequestPassword="Value"
The default Value is administrator.
Periodic Inform Configuration

The Periodic Inform Configuration part allows you to configure the parameters related to the Inform method
call.
When Periodic Inform is enabled, the Mediatrix unit periodically notifies the ACS of parameter changes and
the ACS can then execute commands on the unit. This method allows the unit to contact the ACS when behind
a NAT or firewall. Notifications to the ACS are also sent if there are no parameter changes.
To define the Periodic Inform parameters:

1.
In the Periodic Inform Configuration part, set the Periodic Inform Enable drop-down menu with the
proper behaviour.
This defines whether or not the unit needs to periodically send CPE information to the ACS using
the Inform method call.
Table 488: Periodic Inform Parameters
Parameter
586
Description
Disable
The unit does not send periodic inform.
Enable
The unit sends periodic inform.
Figure 269: Management CWMP Web Page, Periodic Inform Part
2.
Set the Periodic Inform Interval field with the duration, in seconds, of the interval for which the unit
needs to attempt to connect with the ACS and call the Inform method.
3.
Set the Periodic Inform Time field with an absolute time reference in UTC format to determine when
the unit initiates the periodic Inform method calls.
Each Inform call MUST occur at this reference time plus or minus an integer multiple of the Periodic
Inform Interval parameter (Step 2).
The Periodic Inform Time parameter is used only to set the "phase" of the periodic Informs. The
actual value of the Periodic Inform Time field can be arbitrarily far into the past or future.
For example, if Periodic Inform Interval is 86400 (a day) and if Periodic Inform Time is set to UTC
midnight on some day (in the past, present, or future), then periodic Informs will occur every day at
UTC midnight. These MUST begin on the very next midnight, even if Periodic Inform Time refers to
a day in the future.
The Unknown Time value indicates that no particular time reference is specified. That is, the unit
locally chooses the time reference and needs only to follow the specified Periodic Inform Interval.
If absolute time is not available to the unit, its periodic Inform behavior is the same as if the Periodic
Inform Time parameter was set to the Unknown Time value.
The format of the value is:
CCYY-MM-DDThh:mm:ssZ
where:
CCYY: Year number.
MM: Month number in the year.
DD: Day number in the month.
hh: Hour number in the day.
mm: Minute number in the hour.
ss: Second number in the minute.
Example:
1969-07-21T02:28:00Z
The Unknown Time value is defined as:
587
0001-01-01T00:00:00Z
4.
TR-069 Configuration
The TR-069 Configuration part allows you to enable or disable the support of variables under the TR-069
Device.GatewayInfo tree.
To set the TR-069 configuration:

1.
In the TR-069 Configuration part, set the Annex-F drop-down menu with the proper behaviour.
This parameter enables/disables the support of variables under the TR-069 Device.GatewayInfo
tree.
Figure 270: Management CWMP Web Page, TR-069 Configuration Part
2.
Note: This feature is available on the following models:
Mediatrix 3716 / 3731 / 3732 / 3741 / 3742
Mediatrix 4102S
Mediatrix 4104
Mediatrix 4108/4116/4124
Mediatrix LP series
Mediatrix C7 Series
The TR-104 Configuration part allows you to enable or disable the provisioning parameters for VoIP CPE.
588
To set the TR-104 configuration:

1.
In the TR-104 Configuration part, set the TR-104 Enable drop-down menu with the proper
behaviour.
This parameter enables/disables the support of variables under the TR-069
Device.Services.VoiceService tree.
2.
TR-104 Implementation Limitations
The Media5 implementation of TR-104 is limited to the SIPEndpoint profile :1 for the
VoiceService :1 object.
Active notifications are not supported for all objects except when explicitly mentioned under the
Notification field in the mapping table in section ACS Access to the Local Log Table on
page 597.
Only FXS lines can be managed by the TR-104 profile. BRI, PRI or FXO lines are not supported
but can be configured with configuration variables.
A single voice profile is supported, it is instantiated by default. No creation or deletion of voice

profile is allowed.
Only a subset of variables is currently supported. Refer to section ACS Access to the Local
Log Table on page 597.
When TR-104 is used, it is highly recommended not to use other means of configuration (since
TR-104 assumes some configuration).
The TR-106 Configuration part allows you to set the network interface referred by the TR-106 LAN profile.
589
To select a specific profile:

1.
In the TR-106 Configuration part, set the Network Interface drop-down menu with the network
interface referred by the TR-106 LAN profile.
If the menu is empty, the Mediatrix unit uses the network interface configured in Management
Interface Configuration on page 617. If this network interface is set to All, the Mediatrix unit uses
the network interface used for contacting the ACS.
2.
The TR-111 part allows you to set the TR-111 parameters for connection request via NAT Gateway.
The STUN Status field displays the status of the connection with the STUN server.
Table 489: STUN Server Status
Status
590
Description
Disabled
TR-111 is disabled.
Starting
Cwmp service is starting and no connection attempt have been

made so far.
Connected
The Cwmp service is connected to the STUN server.
ErrorCannotResolve
The Cwmp service cannot connect with the STUN server because
the FQDN cannot be resolved.
ErrorNotResponding
The Cwmp service cannot connect with the STUN server because
the server is not responding.
ErrorOther
The Cwmp service cannot connect to the STUN server for

unspecified reason or because no ACS URL is configured. See
device and ACS logs.
To set TR-111 parameters:

1.
In the TR-111 part, enable or disable the use of STUN binding requests to discover the units
external IP address and port in the STUN Enable drop-down menu.
This also sets the ManagementServer.UDPConnectionRequestAddress value accordingly.
The ACS can then send UDP connection requests to the unit at that IP address and port.
Table 490: STUN Parameters
Parameter
Description
Enable
Enables the use of STUN by the unit.
Disable
Disables the use of STUN by the unit.
1
2
When TR-111 STUN is enabled, the NAT Detected status parameter indicates whether or not the
unit has detected the address and/or port mapping in use.
Table 491: NAT Detected Parameters
Parameter
2.
Description
Yes
TR-111 STUN is enabled and the unit has detected the address and/
or port mapping.
No
TR-111 STUN is disabled or the unit has not detected the address
and/or port mapping.
If TR-111 STUN is enabled, set the host name or IP address of the STUN server for the unit to send
Binding Requests in the STUN Server field.
If the STUN server field is empty and TR-111 STUN is enabled, the unit uses the address of the
ACS extracted from the host portion of the ACS URL.
If the port is not specified or set to 0, the default STUN port (3478) is used.
3.
591
Additional TR-111 Parameters


by using the CLI
You can set the STUN username and STUN keep alive parameters. These parameters are effective only if the
STUN Enable drop-down menu is set to Enable (see TR-111 Configuration on page 590 for more details).
To set additional TR-111 parameters:

1.
In the cwmpMIB, set the period range, in seconds, at which STUN Binding Requests must be sent
by the unit for the purpose of maintaining the STUN connection in the tr111StunKeepAlivePeriod
variable.
cwmp.tr111StunKeepAlivePeriod="Value"
The timeouts must be entered in the format 'minimum-maximum'.

The value must be less than or equal to 3600 seconds.
Note: The current implementation does not allow a range. The minimum and maximum values must be the
same.
2.
Set the value of the STUN username attribute to be used in Binding Requests in the
tr111StunUsername variable.
cwmp.tr111StunUsername="Value"
Transport HTTPS Cipher Suite Settings

592

by using the CLI
You can define the allowed cipher suites for the network security settings when using the HTTPS protocol to
connect to the ACS. When the device initiates an HTTPS connection to the ACS, it will negotiate the cipher
suite according to its configuration.
Parameter
CS1
CS2
Description
This represents secure configuration using SHA-1. This value includes the following cipher
suites:
593

Parameter
CS3
Description
To set the HTTPS transport cipher suite configuration parameter:

1.
In the cwmpMIB, locate the transportGroup folder.
2.
Set the HTTPS transport cipher suite configuration in the TransportHttpsCipherSuite variable.
cwmp.TransportHttpsCipherSuite="Value"

Value
Meaning
100
CS1
200
CS2
300
CS3
Transport Certificate Validation

594

by using the CLI
You can define the level of security to use when validating the server's certificate when connecting to the ACS
using HTTPS. The following values are available:
Table 494: Transport Certificate Validation Parameters
Parameter
Description
NoValidati
on
Allows a connection to the server without validating its certificate. The only condition is to
receive a certificate from the server. This option provides partial security and should be
selected with care.
HostName
Allows a connection to the server by validating its certificate is trusted and valid. The
validations performed on the certificate include the expiration date and that the Subject
Alternate Name (SAN) or Common Name (CN) matches the FQDN or IP address of the
server. This is the default setting.
To set the level of security to use when validating the server's certificate when connecting to the ACS
using HTTPS:
1.
In the cwmpMIB, locate the TransportGroup folder.
2.
Set the level of security to use in the TransportCertificateValidation variable.

cwmp.TransportCertificateValidation="Value"

Table 495: Transport Certificate Validation Values
Value
Meaning
100
NoValidation
200
HostName
Allow Unauthenticated UDP Connection Requests


by using the CLI
You can define the CPE behaviour when receiving a UDP connection request message. The following values
are available:
Table 496: Allow Unauthenticated UDP Connection Requests Parameters
Parameter
Description
Enable
Allows any UDP connection request without authentication. This behaviour goes against the
UDP Connection Request described by TR-069 Amendment-2 specification.
Disable
Forces the authentication of a UDP connection request.
To set the CPE behaviour when receiving a UDP connection request message:
1.
In the cwmpMIB, set the behaviour when receiving a UDP connection request message in the
InteropAllowUnauthenticatedUDPConnectionRequests variable.
cwmp.InteropAllowUnauthenticatedUDPConnectionRequests="Value"
595

Table 497: Allow Unauthenticated UDP Connection Requests Values
Value Meaning
0
disable
enable
Parameter Type Validation


by using the CLI
You can define the parameter type validation when the ACS assigns a value to a parameter. The following
values are available:
Table 498: Parameter Type Validation Parameters
Parameter
Description
Tolerant
The client target type is evaluated and conversion is done (if possible): i.e., string to boolean,
string to int, string to unsigned int, string to datetime, etc.
Strict
The ACS and the client must have matching xsd:type otherwise the client rejects the
parameter value.
To set the parameter type validation:

1.
In the cwmpMIB, set the parameter type validation when the ACS assigns a value to a parameter
in the interopParameterTypeValidation variable.
cwmp.interopParameterTypeValidation="Value"

Table 499: Parameter Type Validation Values
Value Meaning
100
Tolerant
200
Strict
MAC Address Format


by using the CLI
You can define the LAN MAC address display format. The following values are available:
Table 500: MAC Address Format Parameters
Parameter
LowerCase
596
Description
MAC address is in lower case format. Ex.: 0090f80d5b4a
Table 500: MAC Address Format Parameters (Continued)

Parameter
UpperCaseWithColon
Description
MAC address is in upper case format, each octet separated by a colon. Ex.:
00:90:F8:0D:5B:4A
To set the parameter type validation:

1.
In the cwmpMIB, set the MAC address format in the interopMacAddressFormat variable.
cwmp.interopMacAddressFormat="Value"

Table 501: MAC Address Format Values
Value
Meaning
100
LowerCase
200
UpperCaseWithColon
ACS Access to the Local Log Table


by using the CLI
You can enable or disable access to the local log table from the ACS. The following values are available:
Table 502: Allow ACS Access to Local Log Table Parameters
Parameter
Description
Enable
Allow access to the local log table from the ACS.
Disable
Disallow access to the local log table from the ACS.
To set the access mode to the local log table from the ACS:
1.
In the cwmpMIB, set access to the local log table in the NlmLocalLogLogEnable variable.
cwmp.NlmLocalLogLogEnable="Value"
where Value may be Enable or Disable (default)
Warning: Enabling access will require significant CPU resources and impact performance of your unit
when notifications are sent frequently to the local log.
597
Supported TR-069 Methods and Parameters

The following lists the TR-069 methods and parameters that the Mediatrix unit supports.
Methods
The Mediatrix unit supports the following TR-069 methods:
GetRPCMethods
SetParameterValues
GetParameterValues
GetParameterNames
GetParameterAttributes
AddObject
DeleteObject
Download
Reboot
Factory Reset
Download Method Information

The supported file types for the Download method are "Firmware Upgrade Image" and "Vendor Configuration
File".
Table 503: Firmware Upgrade Image
Parameter
URL
Comment
Full path of the file "Dgw.mfp" of the MFP to install. The firmware name, version, host,
location and protocol to install are extracted from this path.
The transfer protocol can be "http://", "https://", "ftp://" or "tftp://".
Username
Username used for the firmware upgrade.
Password
Password used for the firmware upgrade.
FileSize
This parameter is not used.
TargetFileName
DelaySeconds
SuccessURL
FailureURL

Table 504: Vendor Configuration File
Parameter
URL
Comment
Full path of the script to install and execute. The script name, location, protocol and host
are extracted from this path.
The transfer protocol can be "http://", "https://", "ftp://" or "tftp://".
598
Username
Username used by the script download.
Password
Password used by the script download.
FileSize
TargetFileName
Table 504: Vendor Configuration File (Continued)

Parameter
Comment
DelaySeconds
SuccessURL
FailureURL
Parameters
The following section lists the parameters that the Mediatrix unit supports according to the "Device" and
"Gateway" profiles.
Device Profile
The Mediatrix unit supports the data model template for device (TR-106) if the Root Element drop-down menu
(General Configuration on page 583 ) is set to "Device".
Supported data models for the Device:1 object:
Baseline:1
LAN:1
Time:1
VoiceService:1 (Partial Support)
GatewayInfo:1
The following table shows the supported parameters access and notifications.
Table 505: Device Profile Parameters
Name
Access
Notification
Device.
Present
N/A
DeviceSummary
Read-only
Always Passive
Device.DeviceInfo.
Present
N/A
Manufacturer
Read-only
Passive
ManufacturerOUI
Read-only
Passive
ModelName
Read-only
Passive
Description
Read-only
Passive
ProductClass
Read-only
Passive
SerialNumber
Read-only
Passive
HardwareVersion
Read-only
Always Passive
SoftwareVersion
Read-only
Always Active
DeviceStatus
Read-only
Passive
UpTime
Read-only
Passive
Device.Time.
Present
N/A
NTPServer1
Read/Write
Passive
NTPServer2
Read/Write
Passive
NTPServer3
Read/Write
Passive
NTPServer4
Read/Write
Passive
CurrentLocalTime
Read-only
Passive
599
Table 505: Device Profile Parameters (Continued)

Name
600
Access
Notification
LocalTimeZone
Read/Write
Passive
Device.GatewayInfo.
Present
N/A
ManufacturerOUI
Read-only
Active
ProductClass
Read-only
Active
SerialNumber
Read-only
Active
Device.LAN.
Present
N/A
AddressingType
Read-only
Passive
IPAddress
Read-only
Always Active
SubnetMask
Read-only
Passive
DefaultGateway
Read-only
Passive
DNSServers
Read-only
Passive
MACAddress
Read-only
Passive
Device.LAN.Stats.
Present
N/A
ConnectionUpTime
Read-only
Passive
TotalBytesSent
Read-only
Passive
TotalBytesReceived
Read-only
Passive
TotalPacketsSent
Read-only
Passive
TotalPacketsReceived
Read-only
Passive
Device.ManagementServer.
Present
N/A
URL
Read/Write
Passive
Username
Read/Write
Passive
Password
Read/Write
Passive
PeriodicInformEnable
Read/Write
Passive
PeriodicInformInterval
Read/Write
Passive
PeriodicInformTime
Read/Write
Passive
ParameterKey
Read/Write
Always
ConnectionRequestURL
Read-only
Active
ConnectionRequestUsername
Read-only
Active
ConnectionRequestPassword
Write
Passive
UpgradesManaged
Read/Write
Passive
UDPConnectionRequestAddress
Read-only
Active
STUNEnable
Read/Write
Active
STUNServerAddress
Read/Write
Active
STUNServerPort
Read/Write
Active
NATDetected
Read-only
Active
STUNMinimumKeepAlivePeriod
Read/Write
Active

Name
Access
Notification
STUNUsername
Read/Write
Active
Services
Present
N/A
.VoiceService.{i}.
Present
N/A
VoiceProfileNumberOfEntries
Read-only
None
.VoiceService.{i}.Capabilities.
Present
N/A
MaxProfileCount
Read-only
None
MaxLineCount
Read-only
None
SignalingProtocols
Read-only
None
DigitMap
Read-only
None
.VoiceService.{i}.Capabilities.SIP.
Present
N/A
Role
Read-only
None
Transports
Read-only
None
URISchemes
Read-only
None
DigitMap
Read/Write
None
.VoiceService.{i}.VoiceProfile.{i}.SIP.
Present
N/A
ProxyServer
Read/Write
None
ProxyServerPort
Read/Write
None
RegistrarServer
Read/Write
None
RegistrarServerPort
Read/Write
None
UserAgentDomain
Read/Write
None
UserAgentPort
Read/Write
None
UserAgentTransport
Read/Write
None
OutboundProxy
Read/Write
None
OutboundProxyPort
Read/Write
None
ConferenceCallURI
Read/Write
Passive
.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.
Present
N/A
Enable
Read/Write
Passive
DirectoryNumber
Read/Write
Passive
Status
Read-only
Passive
.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.
Present
N/A
AuthUserName
Read/Write
None
AuthPassword
Write-only
None
URI
Read/Write
Active
.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Stats.a
Present
N/A
ResetStatistics
Write only
None
PacketsSent
Read-only
Passive
601

Name
602
Access
Notification
PacketsReceived
Read-only
Passive
BytesSent
Read-only
Passive
BytesReceived
Read-only
Passive
Read-only
Passive
Read-only
Passive
IncomingCallsConnected
Read-only
Passive
IncomingCallsFailed
Read-only
Passive
Read-only
Passive
Read-only
Passive
OutgoingCallsConnected
Read-only
Passive
OutgoingCallsFailed
Read-only
Passive
CallsDropped
Read-only
Passive
TotalCallTime
Read-only
Passive
AverageReceiveInterarrivalJitter
Read-only
Passive
.X_0090F8_EpServ.Call.DtmfMap.Allowed{i}.
Present
N/A
Enable
Read/Write
Passive
ApplyTo
Read/Write
Passive
EpId
Read/Write
Passive
DtmfMap
Read/Write
Passive
DtmfTransformation
Read/Write
Passive
TargetHost
Read/Write
Passive
Emergency
Read/Write
Passive
.X_0090F8_EpServ.Call.DtmfMap.Refuse{i}.
Present
N/A
Enable
Read/Write
Passive
ApplyTo
Read/Write
Passive
EpId
Read/Write
Passive
DtmfMap
Read/Write
Passive
.X_0090F8_SipEp.Registration.Status{i}.
Present
N/A
Gateway
Read-only
Passive
Endpoint
Read-only
Passive
State
Read-only
Passive
Registrar
Read-only
Passive
UserName
Read-only
Passive
.X_0090F8_Nlm.LocalLog.
Present
N/A
MaxNbEntries
Read-only
Passive
NbEntries
Read-only
Passive

Name
Access
Notification
NbErrorEntries
Read-only
Active
NbCriticalEntries
Read-only
Active
ResetLog
Write-only
None
.X_0090F8_Nlm.LocalLog.Log{i}.
Present
N/A
LocalTime
Read-only
None
ServiceNumkey
Read-only
None
NotificationId
Read-only
None
Severity
Read-only
None
ServiceTextkey
Read-only
None
Message
Read-only
None
.X_0090F8_Nlm.Events
Present
N/A
Activation
Read/Write
Passive
Type
Read/Write
Passive
Criteria
Read/Write
Passive
Action
Read/Write
Passive
ConfigStatus
Read-only
Passive
a. These statistics are cumulative for all the calls on the line. They are updated at
the end of each call.
Gateway Profile
The Mediatrix unit supports the data model template for internet gateway device (TR-98) if the Root Element
drop-down menu (General Configuration on page 583) is set to "Internet Gateway Device".
The Mediatrix unit supports a subset of the parameters specified in the Baseline:1.
The following table shows the supported parameters access and notification. Parameters not present are not
supported.
Table 506: Internet Gateway Device Profile Parameters
Name
Access
Notification
InternetGatewayDevice.
Present
DeviceSummary
Read-only
Passive
LANDeviceNumberOfEntries
Read-only
Not supported
WANDeviceNumberOfEntries
Read-only
Not supported
InternetGatewayDevice.DeviceInfo.
Present
N/A
Manufacturer
Read-only
Passive
ManufacturerOUI
Read-only
Passive
ModelName
Read-only
Passive
Description
Read-only
Passive
ProductClass
Read-only
Passive
603
Table 506: Internet Gateway Device Profile Parameters (Continued)

Name
604
Access
Notification
SerialNumber
Read-only
Passive
HardwareVersion
Read-only
Always Passive
SoftwareVersion
Read-only
Always Active
SpecVersion
Read-only
Passive
ProvisioningCode
Read/Write
Active
UpTime
Read-only
Passive
DeviceLog
Not supported
Not supported
InternetGatewayDevice.ManagementServer.
Present
N/A
URL
Read/Write
Passive
Username
Read/Write
Passive
Password
Write-only
Passive
Read/Write
Passive
Read/Write
Passive
PeriodicInformTime
Read/Write
Passive
ParameterKey
Read-only
Always
ConnectionRequestURL
Read-only
Always Active
ConnectionRequestUsername
Read/Write
Active
ConnectionRequestPassword
Write-only
Passive
UpgradesManaged
Read/Write
Passive
UDPConnectionRequestAddress
Read-only
Active
STUNEnable
Read/Write
Active
STUNServerAddress
Read/Write
Active
STUNServerPort
Read/Write
Active
NATDetected
Read-only
Active
STUNMinimumKeepAlivePeriod
Read/Write
Active
STUNUsername
Read/Write
Active
InternetGatewayDevice.Time.
Present
N/A
NTPServer1
Read/Write
Passive
NTPServer2
Read/Write
Passive
NTPServer3
Read/Write
Passive
NTPServer4
Read/Write
Passive
CurrentLocalTime
Read-only
Passive
Services
Present
N/A
.VoiceService.{i}.
Present
N/A
VoiceProfileNumberOfEntries
Read-only
None
.VoiceService.{i}.Capabilities.
Present
N/A

Name
Access
Notification
MaxProfileCount
Read-only
None
MaxLineCount
Read-only
None
SignalingProtocols
Read-only
None
DigitMap
Read-only
None
.VoiceService.{i}.Capabilities.SIP.
Present
N/A
Role
Read-only
None
Transports
Read-only
None
URISchemes
Read-only
None
DigitMap
Read/Write
None
.VoiceService.{i}.VoiceProfile.{i}.SIP.
Present
N/A
ProxyServer
Read/Write
None
ProxyServerPort
Read/Write
None
RegistrarServer
Read/Write
None
RegistrarServerPort
Read/Write
None
UserAgentDomain
Read/Write
None
UserAgentPort
Read/Write
None
UserAgentTransport
Read/Write
None
OutboundProxy
Read/Write
None
OutboundProxyPort
Read/Write
None
ConferenceCallURI
Read/Write
Passive
.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.
Present
N/A
Enable
Read/Write
Passive
DirectoryNumber
Read/Write
Passive
Status
Read-only
Passive
.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.SIP.
Present
N/A
AuthUserName
Read/Write
None
AuthPassword
Read/Write
None
URI
Read/Write
Active
.VoiceService.{i}.VoiceProfile.{i}.Line.{i}.Stats.a
Present
N/A
ResetStatistics
Write only
None
PacketsSent
Read-only
Passive
PacketsReceived
Read-only
Passive
BytesSent
Read-only
Passive
BytesReceived
Read-only
Passive
Read-only
Passive
Read-only
Passive
605

Name
606
Access
Notification
IncomingCallsConnected
Read-only
Passive
IncomingCallsFailed
Read-only
Passive
Read-only
Passive
Read-only
Passive
OutgoingCallsConnected
Read-only
Passive
OutgoingCallsFailed
Read-only
Passive
CallsDropped
Read-only
Passive
TotalCallTime
Read-only
Passive
AverageReceiveInterarrivalJitter
Read-only
Passive
.X_0090F8_EpServ.Call.DtmfMap.Allowed{i}.
Present
N/A
Enable
Read/Write
Passive
ApplyTo
Read/Write
Passive
EpId
Read/Write
Passive
DtmfMap
Read/Write
Passive
DtmfTransformation
Read/Write
Passive
.X_0090F8_EpServ.Call.DtmfMap.Refuse{i}.
Present
N/A
Enable
Read/Write
Passive
ApplyTo
Read/Write
Passive
EpId
Read/Write
Passive
DtmfMap
Read/Write
Passive
TargetHost
Read/Write
Passive
Emergency
Read/Write
Passive
.X_0090F8_SipEp.Registration.Status{i}.
Present
N/A
Gateway
Read-only
Passive
Endpoint
Read-only
Passive
State
Read-only
Passive
Registrar
Read-only
Passive
UserName
Read-only
Passive
.X_0090F8_Nlm.LocalLog.
Present
N/A
MaxNbEntries
Read-only
Passive
NbEntries
Read-only
Passive
NbErrorEntries
Read-only
Active
NbCriticalEntries
Read-only
Active
ResetLog
Write-only
None
.X_0090F8_Nlm.LocalLog.Log{i}.
Present
N/A
LocalTime
Read-only
None

Name
Access
Notification
ServiceNumkey
Read-only
None
NotificationId
Read-only
None
Severity
Read-only
None
ServiceTextkey
Read-only
None
Message
Read-only
None
.X_0090F8_Nlm.Events
Present
N/A
Activation
Read/Write
Passive
Type
Read/Write
Passive
Criteria
Read/Write
Passive
Action
Read/Write
Passive
ConfigStatus
Read-only
Passive
a. These statistics are cumulative for all the calls on the line. They are updated at
the end of each call.
607
608
H A P T E R
50
Access Control Configuration

This chapter describes how to set the Access Control parameters of the Mediatrix unit.
Standards Supported

Authentication
RFC 2865: Remote Authentication Dial In User Service

(RADIUS)
RFC 2866: RADIUS Accounting
Caution: The Access Control page is not accessible if you have the User or Observer access right. See
Users
The Users section allows you to manage the users that can access the web interface. You can add a maximum
of 10 users.
To manage users:
1.
In the web interface, click the Management link, then the Access Control sub-link.
Figure 274: Management Access Control Web Page
2.
If you want to add a new user, enter its name in the blank User Name field in the bottom left of the
window, enter the corresponding password in the blank Password field, then click the
button.
The name is case-sensitive.
3.
If you want to delete an existing user, click the corresponding
button.
If you delete all users in the table, the profiles default user(s) will be used upon unit restart.
Note: A system restart is required to completely remove the user. The current activities of this user are not
terminated on removal.
4.
If you want to change the password of an existing user, type it in the corresponding Password field.
The password is case sensitive. All characters are allowed.
5.
Define the access rights template applying to a user in the corresponding Access Rights drop-down
menu.
607
Chapter 50 - Access Control Configuration
Services Access Control Type

Table 507: Access Rights
Access Right
Description
Admin
User is allowed to read and modify all variables of the unit.
User
User is allowed to read and modify all variables except passwords and secrets.
Observer
User is only allowed to read variables that are not passwords or secrets.
See Access Rights Description on page 611 for mode details on the various operations allowed
with each access right.
6.
Partial Reset
When a partial reset is triggered, the password and access rights reset back to the default value (see Partial
Reset on page 15 for more details).
Services Access Control Type

The Services Access Control Type section allows you to define the type of authentication and accounting to
use for the CLI, SNMP, and Web services.
Authentication provides a way of identifying a user, typically by having the user enter a valid user name and
valid password before access is granted.
Accounting measures the resources a user consumes during access. This can include the amount of system
time or the amount of data a user has sent and/or received during a session.
To set the Services Access Control type:

1.
In the Services Access Control Type section of the Access Control page, set the authentication type
a service uses for incoming authentication requests in the corresponding Authentication Type
column.
Table 508: Authentication Types
Type
Description
Local
Incoming authentication attempts are validated against the user names and
passwords stored in the Local Users table (see Users on page 607 for more
details).
Radius
Incoming authentication attempts are validated against the first responding

Radius server configured in the Radius Servers section (Radius Servers on
page 609). When no server is configured or the servers are unreachable, an
authentication attempt of type Local is performed against the user names and
passwords stored in the Local Users table (see Users on page 607 for more
details).
Note: This type is not available for the SNMP interface.

Figure 275: Access Control Services Access Control Type Section
1
608
Radius Servers
2.
Set the accounting type a service uses in the corresponding Accounting Type column.
Accounting starts once users are successfully authenticated and stops when their session is over.
Table 509: Accounting Types
Type
3.
Description
None
Accounting is disabled.
Radius
Accounting is done by the first responding Radius server configured in the

Radius Servers section (Radius Servers on page 609).
Partial Reset
When a partial reset is triggered, the Radius authentication is disabled (see Partial Reset on page 15 for
more details).
Radius Servers
The Radius Servers section allows you to define up to three Radius servers. It also allows you to define
authentication server and accounting server information, for the CLI, SNMP, and Web services.
Note: The Mediatrix units Radius server settings do not support IPv6. See IPv4 vs. IPv6 on page 87 for
more details.
Radius Authentication occurs when the Authentication Type column of the Services Access Control Type
section (Services Access Control Type on page 608) is set to Radius for the service from which the
authentication request is coming. You can configure up to three Radius servers for each service listed in the
Select a Service drop-down menu. The first authentication attempt is sent to the Radius server with the highest
priority, which is set in the Priority column (1 being the highest priority). When authentication fails or the
request reaches the timeout set in the Server Request Timeout field, the next server with the highest priority
is used. When all servers have failed to reply or no servers are configured for the service asking for
authentication, authentication is attempted against local user names and passwords as a fallback strategy.
Radius authentication is available for the CLI and Web services.
Radius Accounting is enabled by setting the Accounting Type column of the Services Access Control Type
section (Services Access Control Type on page 608) to Radius for one or more services. When such a
configuration is set, accounting requests made through those services are forwarded to a Radius server
configured in the Radius Servers section. You can configure up to three Radius servers for each service listed
in the Select a Service drop-down menu. The first accounting request is sent to the Radius server with the
highest priority, which is set in the Priority column (1 being the highest priority). When the accounting request
fails or the request reaches the timeout set in the Server Request Timeout field, the next server with the
highest priority is used. The CLI, Web, and SNMP services can use the accounting functionality.
To set the Radius servers information:

1.
Select to which service you want to apply the changes in the Select a Service drop-down menu
above the Radius Servers section.
You can copy the configuration of the selected service to one or more services of the Mediatrix unit
in the Apply to the Following Services section at the bottom of the page. You can select specific
services by checking them, as well as use the Check All or Uncheck All buttons.
2.
In the Authentication part of the Radius Servers section, set the host name and port of a Radius
server used for authentication requests in the corresponding Host field.
609
Radius Servers
Figure 276: Access Control Radius Servers Section

1
2
3
4
5
6
7
You can configure up to three Radius servers with a different priority.
Note: This parameter is not available for the SNMP service.

3.
Set the secret key shared between the Radius server and the unit in the corresponding Server
Secret field.
The Authentication Secret key must be the same as the secret key stored on the corresponding
Radius authentication server.
Note: This parameter is not available for the SNMP service.

4.
In the Accounting part, set the host name and port of a Radius server used for accounting requests
in the corresponding Host field.
You can configure up to three Radius servers with a different priority.
5.
Set the secret key shared between the Radius server and the unit in the corresponding Server
Secret field.
The Accounting Secret key must be the same as the secret key stored on the corresponding Radius
accounting server.
6.
Set the Server Request Timeout field with the maximum time, in milliseconds, the unit waits for a
reply from a Radius server.
This parameter applies to all services. Upon reaching the timeout, the request is sent to the next
configured server.
7.
Define the access rights template applying to a user in the corresponding Radius Users Access
Rights drop-down menu.
This parameter applies to all services. You have the following choices:
Table 510: Radius Users Access Rights
Access Right
Description
Admin
User is allowed to read and modify all variables of the unit.
User
User is allowed to read and modify all variables except passwords and secrets.
Observer
User is only allowed to read variables that are not passwords or secrets.
See Access Rights Description on page 611 for mode details on the various operations allowed
with each access right.
8.
610
Access Rights Description

You have three templates of rights from which you can select the permissions given to each user allowed in a
unit (see Users on page 607 and Radius Servers on page 609).
The following table describes the various operations allowed with each access right.
Table 511: Access Rights Description
Access Right
Read configuration
Observer
Yes
User
Yes
a
Admin
Yes
Modify Configuration
No
Yes
Yes
Read/Write Passwords
No
No
Yes
Change Access Rights
No
No
Yes
Execute Configuration Script No
Yesa
Yes
Export Configuration
No
Yesa
Yes
Backup/Restore
configuration
No
No
Yes
Firmware updates
No
No
Yes
a. Passwords cannot be changed and will not be exported to a configuration script.
611
612
H A P T E R
51
File Manager
This chapter describes how to use the units File Manager.
File Manager
The File page allows you to view and delete the files you have created with the File transfer protocol, for
instance, a configuration backup.
It also allows you to see the default and user-defined configuration presets for the ISDN, R2 CAS and E&M
protocols. They differ depending on the Mediatrix unit you are using. Depending on your unit's profile, it may
be possible that no preset files are available.
Note: The files under the File service are not included in the backup process. In the same way, the restore
process will not remove any file under the File service.
To use the File manager:

1.
In the web interface, click the Management link, then the File sub-link.
Figure 277: Management File Web Page
If you want to delete an existing file, click the corresponding
button.
You can directly download a file via your web browser by clicking it. You will then be able to see its
contents.
2.
To add a file to the units File System, type the path and name of the file to add in the field of the
Upload File Through Web Browser section, or select an existing one on the PC with the Browse
button.
If you are currently using an unsecure HTTP access, the Upload File Through Web Browser section
is disabled. This is to avoid transferring a file in clear text. To enable the section, access the secure
site by clicking the Activate unsecure file transfer through web browser link at the top of the window.
3.
Click Submit.
Partial Reset
613
Chapter 51 - File Manager
File Manager
Transfer Protocols
Table 512:
Name
Status
HTTP
Download only. Basic and digest authentication

supported.
HTTPS
Download only. Requires a valid trusted

certificate matching the remote servers
certificate to be available through Cert. Basic
and digest authentication supported.
TFTP
Download only.
FTP
Download only.
Security Certificates
This service makes use of security certificates as configured in the Certificate Manager service (Cert). It
retrieves the certificates from Cert and then uses them as needed to authenticate remote servers.
HTTPS Transfer Settings

If the TLS server requests a certificate from the client (with a CertificateRequest message), the connection
must be mutually authenticated by sending a message containing the clients certificate during the TLS
handshake. The corresponding certificate is retrieve from the Cert.HostCertificateAssociation table.
When the transfer method is HTTPS, the negotiated network security settings depend on the CipherSuite
configuration. The current cipher suite choices:
CS1: This is the default value and represents the cipher suites configuration prior to this
variable introduction. This should be changed if additional network security is required.
CS2: This represents a secure configuration using SHA-1.

CS3: This represents the most secure configuration using SHA-2. Only the most secure cipher
HTTPS Transfer Cipher Suite Settings

614

by using the CLI
File Manager
Parameter
CS1
Description
CS2
suites:
615
Chapter 51 - File Manager
File Manager

Parameter
CS3
Description

1.
In the MIB, locate the transferGroup folder.
2.
Set the HTTPS transfer cipher suite configuration in the TransferHttpsCipherSuite variable.
file.TransferHttpsCipherSuite="Value"

616
Value
Meaning
100
CS1
200
CS2
300
CS3
H A P T E R
52
Miscellaneous
This chapter describes how to set various parameters used to manage the Mediatrix unit.
Management Interface Configuration

The Miscellaneous page allows you to specify which one of the existing network interfaces is used to manage
the Mediatrix unit.
To set the system management interface:

1.
In the web interface, click the Management link, then the Misc sub-link.
Figure 278: Management Misc Web Page
2.
Select which one of the existing network interfaces is used to manage the device in the Network
Interface drop-down menu.
The management services (typically Web and/or SNMP) can be reached through this network
interface.
Before the system management services can be used, they need to be bound (or linked) to a
physical port of your Mediatrix unit.
The special value "All" means to bind all network interfaces.
3.
Partial Reset
When a partial reset is triggered, the Management Interface reverts back to its default value.
Activate Licence
The Activate Licence section allows you to activate a feature via an encrypted licence key. Once decrypted,
the licence key will contain the feature to activate.
This functionality currently allows you to enable the TR-069 feature on units that are already deployed (see
Chapter 49 - CWMP Configuration on page 581 for more details on TR-069).
To use the Feature Activation:

1.
In the Activate Licence section of the Misc page, enter the licence key in the Licence Key field.
Figure 279: Management Activate Licence Section
1
617
Chapter 52 - Miscellaneous
Activate Licence
2.
618
Click Submit & Activate Now to activate the feature.
Appendices
P P E N D I X
Country-Specific Parameters
The following parameters differ depending on the country in which you are.
Definitions
The following are some useful definitions.
Table 514: Definitions
Term
Description
Dial Tone
Indicates the endpoint is ready to receive dialing.
Busy Tone
Indicates the endpoint or equipment is in use, engaged or occupied.
Ringback Tone
Indicates the called line is ringing out.
Special Information Tone
Identifies network-provided announcements.
Stutter Dial Tone
Notifies the user that they have a voice mail message when the phone
does not or cannot have a message-waiting light.
Confirmation Tone
Confirms a command performed by the user (such as activate a

service).
Receiver Off Hook (ROH) Tone
Indicates that the telephone is not hung up correctly.
Message Waiting Indicator Tone
Indicates there is a message waiting somewhere for the owner of the

phone
Network Congestion Tone
Indicates that all switching paths are busy, all toll trunks are busy, or
there are equipment blockages.
Intercept Tone
Indicates that you have dialed incorrectly or that the feature you've
requested is not available on your terminal.
Preemption Tone
In military telephone systems, a distinctive tone that is used to indicate

to connected users, i.e., subscribers, that their call has been
preempted by a call of higher precedence.
Reorder Tone
Indicates that all switching paths are busy, all toll trunks are busy, there
are equipment blockages, the caller dialled an unassigned code, or the
digits dialled got messed up along the way.
FED Tone
Indicates the far end tone detection.
Conventions
The following conventions apply to this Appendix.
Frequencies
Symbol * means modulated. For instance: 425 Hz * 25 means 425 Hz modulated at 25 Hz.
Symbol + means added. For instance: 425 Hz + 330 Hz means that both 425 Hz and 330 Hz
621
Appendix A - Country-Specific Parameters
Definitions
sines are played at the same time.
When a tone is composed of more than one frequency, if not otherwise specified, the given
electrical level applies to each frequency taken separately.
Impedance
Impedance is the apparent resistance, in an electric circuit, to the flow of an alternating current, analogous to
the actual electrical resistance to a direct current, being the ratio of electromotive force to the current.
When representing an impedance, the following applies:
Symbol // means parallel.

Symbol + means serial.
Furthermore, there are two types of impedances:
Input Impedance
Terminal Balance Return Loss (TBRL) Impedance
Input Impedance
Impedance of the Mediatrix at the Tip and Ring wires.
Terminal Balance Return Loss (TBRL) Impedance

Balance return loss attributable to transmission loss between two points. It is used to characterize an
impedance balancing property of the 2-wire analog equipment port.
Each country has its own definition of the TBRL value. For instance, in North America, TIA/EIA 464 (and TIA/
EIA 912) define two TBRL values:
600 for on-premise or short loop ports.

350 + (1000 || 21 nF) for off-premise or long loop ports.
A wire length above 2.5 km is considered long loop according to TIA/EIA 912 section 6.4 (7)(b)).
In Europe, ETSI 300 439 also mentions a TBRL value. However, most European countries have different
requirements regarding the TBRL Impedance. This is also true for other countries around the world. Each one
of them has different requirements.
Line Attenuation
Values are given in dBr (deciBel relative):
A + for input means that the digital side is attenuated by x decibels relative to the analog side.
A + for output means that the analog side is amplified by x decibels relative to the digital side.
A - for input means that the digital side is amplified by x decibels relative to the analog side.
A - for output means that the analog side is attenuated by x decibels relative to the digital side.
On-Off Sequences
Values in bold are on cycles, where tones are audible. Values in normal style are off cycles, where tones
are not audible. When not otherwise specified, sequences repeat forever. A x symbol means that the
sequences between parenthesis is repeated x times. The next cycle(s) repeat forever, unless otherwise
specified. Values are in seconds.
For instance:
3*(0.1
0.1) then 0.6 1.0 - 0.2 0.2
means that the 0.1s on and 0.1s off sequence is repeated 3 times, afterwards the 0.6s on, 1.0s off, 0.2s on
and 0.2s off sequence repeats forever.
622
Definitions
Distinctive Ring
Note: This section applies to the Mediatrix LP/4100/C7 series only.
The distinctive ring service allows you to have four different numbers with each their own ring. The numbers
ring through a single line coming into the business or residence and each number can be distinguished by the
pattern of the ring. These ring patterns are made up of various combinations of ring bursts.
This feature uses the Alert-Info header from the initial INVITE of a call to know if the call requires a distinctive
ringing.
The supported value of the Alert-Info are:
Table 515: Distinctive RIng Patterns
Alert-Info value
Ring Name
On Off Sequence (s)
<http://127.0.0.1/Bellcore-dr2>
Bellcore-dr2
0.8 0.4, 0.8 4.0
Bellcore-dr3
0.4 0.2, 0.4 0.2, 0.8 4.0
Bellcore-dr4
0.3 0.2, 1.0 0.2, 0.3 4.0
All other values or call

property
not present
Country normal ring
The Mediatrix plays the default ring of the country selected if the Alert-Info value is not present or the value is
not supported.
Note: Since the first pause of the distinctive ring is lower that 1 second, a splash ring followed by an Off of
1 second precedes the distinctive ring pattern.
623
Australia
Australia
The following parameters apply if you have selected Australia as location.
Australia 1
The following parameters apply if you have selected Australia 1 as location.
Table 516: Australia 1 Parameters
Parameter
Value
Elect. Levels
Busy Tone
425 Hz
0.38 0.38
-18 dBm
Call Waiting Tone
425 Hz
0.2 - 0.2, 0.2 - 4.4, 0.2 - 0.2, 0.2 - 4.4
-23 dBm
Dial Tone
425 Hz
400 Hz
450 Hz
CONTINUOUS
CONTINUOUS
CONTINUOUS
-18 dBm
-24 dBm
-24 dBm
425 Hz
400 Hz
450 Hz
(0.1 - 0.04)x72, CONTINUOUS

-18 dBm
-24 dBm
-24 dBm
425 Hz
425 Hz
0.38 - 0.38, 0.38 - 0.38

0.38 - 0.38, 0.38 - 0.38
-13 dBm
-23 dBm
2350 Hz
CONTINUOUS
-5 dBm
Reorder Tone
425 Hz
2.5 - 0.5
-18 dBm
Ringback Tone
425 Hz
400 Hz
450 Hz
0.4 - 0.2, 0.4 - 2.0

0.4 - 0.2, 0.4 - 2.0
0.4 - 0.2, 0.4 - 2.0
-18 dBm
-24 dBm
-24 dBm
425 Hz
2.5 - 0.5
-18 dBm
Stutter Dial Tone
425 Hz
400 Hz
450 Hz
CONTINUOUS
CONTINUOUS
CONTINUOUS
-18 dBm
-24 dBm
-24 dBm
Ring
AC: 53 VRMS, 25 Hz
DC: -10 Vdc
0.4 - 0.2, 0.4 - 2.0
Loop Current
30 ma
Input Impedance (FXS)
600
Input Impedance (FXO)
600
Tbrl-Impedance
600
FED Tone
425 Hz
Default Caller ID (FXS)
BELLCORE
8.0
FXS Line Attenuation (Input)
+0 dBr
FXS Line Attenuation (Output)
-6 dBr
FXO Line Attenuation (Input)
+6 dBr
+0 dBr
FXO Line Attenuation (Output)
624
On Off - CID Sequence (s)
Delay Before Answering
2 seconds
Delay Before Dialing (No Dial

Tone Detection)
4 seconds
Australia
Australia 2
The following parameters apply if you have selected Australia 2 as location.
Table 517: Australia 2 Parameters
Parameter
Value
Elect. Levels
Busy Tone
425 Hz
0.38 0.38
-18 dBm
Call Waiting Tone
425 Hz
0.2 - 0.2, 0.2 - 4.4, 0.2 - 0.2, 0.2 - 4.4
-23 dBm
Dial Tone
425 Hz
400 Hz
450 Hz
CONTINUOUS
CONTINUOUS
CONTINUOUS
-18 dBm
-24 dBm
-24 dBm
425 Hz
400 Hz
450 Hz

-18 dBm
-24 dBm
-24 dBm
425 Hz
425 Hz
0.38 - 0.38, 0.38 - 0.38

0.38 - 0.38, 0.38 - 0.38
-13 dBm
-23 dBm
2350 Hz
CONTINUOUS
-5 dBm
Reorder Tone
425 Hz
2.5 - 0.5
-18 dBm
Ringback Tone
425 Hz
400 Hz
450 Hz
0.4 - 0.2, 0.4 - 2.0

0.4 - 0.2, 0.4 - 2.0
0.4 - 0.2, 0.4 - 2.0
-18 dBm
-24 dBm
-24 dBm
425 Hz
2.5 - 0.5
-18 dBm
Stutter Dial Tone
425 Hz
400 Hz
450 Hz
CONTINUOUS
CONTINUOUS
CONTINUOUS
-18 dBm
-24 dBm
-24 dBm
Ring
AC: 53 VRMS, 25 Hz
DC: -10 Vdc
0.4 - 0.2, 0.4 - 2.0
Loop Current
30 ma
220 + 820 // 120 nF
220 + 820 // 115 nF
Tbrl-Impedance
220 + 820 // 120 nF
FED Tone
425 Hz
BELLCORE
8.0
-3 dBr
-9 dBr
+6 dBr
+0 dBr
2 seconds

Tone Detection)
4 seconds
625
Austria
Austria
The following parameters apply if you have selected Austria1 as location.
Table 518: Austria1 Parameters
Parameter
626
Value
Elect. Levels
Busy Tone
450 Hz
0.3 0.3
-20 dBm
Call Waiting Tone
440 Hz
2.0 0.3, 10.0 0.3, 10.0
-17 dBm
Confirmation Tone
450 Hz
(0.1 0.1) x 3 End
-20 dBm
Dial Tone
450 Hz
CONTINUOUS
-20 dBm
450 Hz
(0.1 0.1) x 10, CONTINUOUS
-20 dBm
450 Hz
0.3 0.3
-20 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Reorder Tone
950 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-20 dBm
-20 dBm
-20 dBm
Ringback Tone
450 Hz
1.0 5.0
-20 dBm
950 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-20 dBm
-20 dBm
-20 dBm
Stutter Dial Tone
450 Hz
-20 dBm
Ring (FXS)
AC: 45 VRMS, 50 Hz
DC: -15 Vdc
1.0 5.0
Loop Current
30 ma
Flash Hook Detection Range
Min: 170 ms
Max: 900 ms
220 + 820 // 115 nF
270 + 750 // 150 nF
Tbrl-Impedance
600
FED Tone
450 Hz
BELLCORE
8.0
-3 dBr
-10 dBr
+6 dBr
-1 dBr
0 seconds

Tone Detection)
4 seconds
Brazil
Brazil
The following parameters apply if you have selected Brazil as location.
Table 519: Brazil Parameters
Parameter
Value
Elect. Levels
Busy Tone
425 Hz
0.25 0.25
-10 dBm
Call Waiting Tone
440 Hz
2.0 0.3, 10.0 0.3, 10.0
-17 dBm
Confirmation Tone
425 Hz
(0.1 0.1) x 3 End
-15 dBm
Dial Tone
425 Hz
CONTINUOUS
-15 dBm
425 Hz
-15 dBm
425 Hz
0.2 0.2
-10 dBm
425 Hz
0.25 0.25
-10 dBm
Reorder Tone
425 Hz
0.75 0.25, 0.25 0.25
-10 dBm
Ringback Tone
425 Hz
1.0 4.0
-15 dBm
950 Hz
1400 Hz
1800 Hz
0.33 - 0.03 - 0.33 - 0.03 - 0.33 - 1.0

0.33 - 0.03 - 0.33 - 0.03 - 0.33 - 1.0
0.33 - 0.03 - 0.33 - 0.03 - 0.33 - 1.0
-15 dBm
-15 dBm
-15 dBm
Stutter Dial Tone
425 Hz
-15 dBm
Ring (FXS)
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
1.0 4.0
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
900
900
Tbrl-Impedance
800 // 50 nF
FED Tone
425 Hz
TELEBRAS_DTMF
8.0
0 dBr
-7 dBr
6 dBr
0 dBr
0 seconds

Tone Detection)
0 seconds
627
China
China
The following parameters apply if you have selected China as location.
Table 520: China Parameters
Parameter
Value
Elect. Levels
Busy Tone
450 Hz
0.35 0.35
-10 dBm
Call Waiting Tone
450 Hz
0.4 4.0, 0.4 4.0
-20 dBm
Confirmation Tone
450 Hz
(0.1 0.1) x 3, End
-10 dBm
Dial Tone
450 Hz
CONTINUOUS
-10 dBm
Intercept Tone
450 Hz
0.2 0.2, 0.2 0.6
-20 dBm
450 Hz
0.4 0.04
-10 dBm
450 Hz
0.7 0.7
-10 dBm
Preemption Tone
450 Hz
0.2 0.2, 0.2 0.6
-20 dBm
950 Hz
950 Hz
950 Hz
950 Hz
5.0 5.0 5.0 5.0

5.0 5.0 5.0 5.0
5.0 5.0 5.0 5.0
5.0 5.0 5.0 5.0
-25 dBm
-16 dBm
-8 dBm
-6 dBm
Reorder Tone
450 Hz
0.1 0.1, 0.1 0.1, 0.1 0.1, 0.4 0.4
-10 dBm
Ringback Tone
450 Hz
1.0 4.0
-10 dBm
950 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-10 dBm
-10 dBm
-10 dBm
-10 dBm
Stutter Dial Tone
450 Hz
0.4 0.04
Ring
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
1.0 4.0
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
600
600
Tbrl-Impedance
600
FED Tone
450 Hz
Default Caller ID
BELLCORE
8.0
0 dBr
-9 dBr
0 dBr
0 dBr
628
0 seconds

Tone Detection)
0 seconds
Czech Republic
Czech Republic
The following parameters apply if you have selected Czech Republic1 as location.
Table 521: Czech Republic1 Parameters
Parameter
Value
Elect. Levels
Busy Tone
425 Hz
0.33 0.33
-12 dBm
Call Waiting Tone
425 Hz
2.0 0.33, 10.0 0.33, 10.0
-11 dBm
Confirmation Tone
425 Hz
(0.1 0.1) x 3, End
-12 dBm
Dial Tone
425 Hz
0.33 0.33, 0.66 0.66
-12 dBm
425 Hz
(0.1 0.1) x 10, 0.33 0.33, 0.66 0.66
-12 dBm
425 Hz
0.17 0.17
-12 dBm
425 Hz
0.17 0.17
-12 dBm
Ringback Tone
425 Hz
1.0 4.0
-12 dBm
950 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-12 dBm
-12 dBm
-12 dBm
Stutter Dial Tone
425 Hz
(0.17 0.17) x 3, 0.66 0.66
-12 dBm
Ring (FXS)
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
1.0 4.0
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
600
270 + 750 // 150 nF
Tbrl Impedance
220 + 820 // 115 nF
FED Tone
425 Hz
V23
0.165 0.165
0 dBr
-7 dBr
+6 dBr
-1 dBr
0 seconds

Tone Detection)
4 seconds
629
Denmark
Denmark
The following parameters apply if you have selected Denmark1 as location.
Table 522: Denmark1 Parameters
Parameter
Value
Elect. Levels
Busy Tone
425 Hz
0.5 0.5
-10 dBm
Call Waiting Tone
440 Hz
2.0 0.3 10.0 0.3 10.0
-17 dBm
Confirmation Tone
425 Hz
(0.1 0.1) x 3, End
-15 dBm
Dial Tone
425 Hz
CONTINUOUS
-15 dBm
425 Hz
-15 dBm
425 Hz
0.2 0.2
-10 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Ringback Tone
425 Hz
1.0 4.0
-15 dBm
950 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-15 dBm
-15 dBm
-15 dBm
Stutter Dial Tone
425 Hz
-15 dBm
Ring
AC: 45 VRMS, 25Hz

DC: -15 Vdc
1.0 4.0
Loop Current
30 ma
300 + 1000 // 220 nF
270 + 750 // 150 nF
Tbrl Impedance
400 + 500 // 330 nF
FED Tone
425 Hz
TDK DTMF
8.0
+0 dBr
-6 dBr
-6 dBr
+0 dBr
630
0 seconds

Tone Detection)
0 seconds
France
France
The following parameters apply if you have selected France1 as location.
Table 523: France1 Parameters
Parameter
Value
Elect. Levels
Busy Tone
440 Hz
0.5 0.5
-19.9 dBm
Call Waiting Tone
440 Hz
2.0 0.3, 10.0 0.3, 10.0
-17 dBm
Confirmation Tone
440 Hz
(0.1 0.1) x 3, End
-16.9 dBm
Dial Tone
440 Hz
CONTINUOUS
-16.9 dBm
440 Hz
-16.9 dBm
440 Hz
0.25 0.25
-19.9 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Ringback Tone
440 Hz
1.5 3.5
-19.9 dBm
950 Hz
1400 Hz
1800 Hz
0.3 - 0.03 - 0.3 - 0.03 - 0.3 - 1.0

0.3 - 0.03 - 0.3 - 0.03 - 0.3 - 1.0
0.3 - 0.03 - 0.3 - 0.03 - 0.3 - 1.0
-19.9 dBm
-19.9 dBm
-19.9 dBm
Stutter Dial Tone
440 Hz
-19.9 dBm
Ring (FXS)
AC: 45 VRMS, 50 Hz
DC: -15 Vdc
1.5 3.5
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
215 + 1000 // 137 nF
270 + 750 // 150 nF
Tbrl-Impedance
600
FED Tone
440 Hz
BELLCORE
8.0
+1.9 dBr
-8.9 dBr
+6 dBr
-1 dBr
0 seconds

Tone Detection)
4 seconds
631
Germany
Germany
The following parameters apply if you have selected Germany as location.
Germany 1
The following parameters apply if you have selected Germany 1 as location.
Table 524: Germany 1 Parametersa
Parameter
Value
Elect. Levels
Busy Tone
425 Hz
0.48 0.48
-16 dBm
Call Waiting Tone
440 Hz
0.3 End
-17 dBm
Confirmation Tone
425 Hz
(0.1 0.1) x 3, End
-16 dBm
Dial Tone
425 Hz
CONTINUOUS
-16 dBm
425 Hz
-16 dBm
425 Hz
0.24 0.24
-16 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Ringback Tone
425 Hz
1.0 4.0
-16 dBm
900 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-16 dBm
-16 dBm
-16 dBm
Stutter Dial Tone
425 Hz
-16 dBm
Ring (FXS)
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
1.0 4.0
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
220 + 820 // 115 nF
270 + 750 // 150 nF
Tbrl-Impedance
220 + 820 // 115 nF
FED Tone
425 Hz
BELLCORE
8.0
-3 dBr
-10 dBr
+6 dBr
-1 dBr
0 seconds

Tone Detection)
4 seconds
a. The Germany 2 choice in the MIB is exactly the same as Germany 1.
632
Germany
Germany 2
The following parameters apply if you have selected Germany 2 as location.
Table 525: Germany 2 Parameters
Parameter
Value
Elect. Levels
Busy Tone
425 Hz
0.48 0.48
-13 dBm
Call Waiting Tone
440 Hz
2.0 0.3, 10.0 0.3, 10.0
-17 dBm
Confirmation Tone
425 Hz
(0.1 0.1) x 3, End
-13 dBm
Dial Tone
425 Hz
CONTINUOUS
-13 dBm
425 Hz
-13 dBm
425 Hz
0.24 0.24
-13 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-16 dBm
Ringback Tone
425 Hz
1.0 4.0
-13 dBm
900 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-13 dBm
-13 dBm
-13 dBm
Stutter Dial Tone
425 Hz
-13 dBm
Ring (FXS)
AC: 57 VRMS, 25 Hz
DC: -5 Vdc
1.0 4.0
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
220 + 820 // 115 nF
270 + 750 // 150 nF
Tbrl-Impedance
220 + 820 // 115 nF
FED Tone
425 Hz
BELLCORE
8.0
0 dBr
-7 dBr
+6 dBr
-1 dBr
0 seconds

Tone Detection)
4 seconds
633
Israel2
Israel2
The following parameters apply if you have selected Israel2 as location.
Table 526: Israel2 Parameters
Parameter
634
Value
Elect. Levels
Busy Tone
400 Hz
0.5 0.5
-14 dBm
Call Waiting Tone
400 Hz
0.5 10.0, 0.5 10.0
-16 dBm
Confirmation Tone
400 Hz
0.17 0.34, 0.14 0.14, End
-14 dBm
Dial Tone
400 Hz
CONTINUOUS
-14 dBm
Hold Tone
400 Hz
0.05 2.0, End
-16 dBm
400 Hz
(0.16 0.16) x 10, CONTINUOUS
-14 dBm
400 Hz
0.25 0.25
-14 dBm
1440+2060+2452+2600 Hz
0.12 0.1
-14 dBm
Reorder Tone
1000 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-14 dBm
-14 dBm
-14 dBm
Ringback Tone
400 Hz
1.0 3.0
-14 dBm
450 + 150 Hz
0.5 End
-14 dBm
Stutter Dial Tone
400 Hz
-15 dBm
Ring
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
1.0 3.0
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
600
600
Tbrl-impedance
600
FED Tone
400 Hz
BELLCORE
8.0
0 dBr
-9 dBr
0 dBr
0 dBr
0 seconds

Tone Detection)
0 seconds
Italy
Italy
The following parameters apply if you have selected Italy1 as location.
Table 527: Italy1 Parameters
Parameter
Value
Elect. Levels
Busy Tone
425 Hz
0.5 0.5
-13 dBm
Call Waiting Tone
440 Hz
2.0 0.3, 10.0 0.3, 10.0
-17 dBm
Confirmation Tone
425 Hz
(0.1 0.1) x 3, End
-13 dBm
Dial Tone
425 Hz
0.2 0.2, 0.6 1.0
-13 dBm
425 Hz
(0.1 0.1) x 10, 0.2 0.2, 0.6 1.0
-13 dBm
425 Hz
0.2 0.2
-13 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Ringback Tone
425 Hz
1.0 4.0
-13 dBm
950 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-20 dBm
-20 dBm
-20 dBm
Stutter Dial Tone
425 Hz
(0.1 0.1) x 3, 0.2 0.2, 0.6 1.0
-13 dBm
Ring (FXS)
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
1.0 4.0
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
180 + 630 // 60 nF
270 + 750 // 150 nF
Tbrl-impedance
750 // 18 nF
FED Tone
425 Hz
BELLCORE
0.2 0.2, 0.6 1.0
0 dBr
-7 dBr
+6 dBr
-1 dBr
0 seconds

Tone Detection)
4 seconds
635
Japan
Japan
The following parameters apply if you have selected Japan 2 as location.
Table 528: Japan 2 Parameters
Parameter
Value
Elect. Levels
Busy Tone
400 Hz
0.5 0.5
-13 dBm
Call Waiting Tone
400 Hz
2.0 - 0.3, 10.0 - 0.3, 10.0
-17 dBm
Confirmation Tone
400 Hz
(0.1 0.1) x 3, End
-13 dBm
Dial Tone
400 Hz
CONTINUOUS
-19 dBm
400 Hz
-13 dBm
400 Hz
0.5 0.5
-13 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Ringback Tone
400 Hz
420 Hz
380 Hz
1.0 2.0
1.0 2.0
1.0 2.0
-16 dBm
-22 dBm
-22 dBm
400 Hz
0.1 0.1
-13 dBm
Stutter Dial Tone
400 Hz
-13 dBm
Ring
AC: 45 VRMS, 20 Hz
DC: -15 Vdc
1.0 2.0
Loop Current
30 ma
600 1000 nF
600
Tbrl-Impedance
600 1000 nF
FED Tone
400 Hz
Default Caller ID
BELLCORE
8.0
+0 dBr
-9 dBr
+0 dBr
+0 dBr
636
2 seconds

Tone Detection)
3 seconds
Mexico
Mexico
The following parameters apply if you have selected Mexico as location.
Table 529: Mexico Parameters
Parameter
Value
Elect. Levels
Busy Tone
425 Hz
0.25 0.25
-18 dBm
Call Waiting Tone
440 Hz
2.0 0.3, 10.0 0.3, 10.0
-17 dBm
Confirmation Tone
425 Hz
(0.1 0.1) x 3, End
-14 dBm
Dial Tone
425 Hz
CONTINUOUS
-14 dBm
425 Hz
(0.1 0.1) x 10 CONTINUOUS
-14 dBm
425 Hz
0.25 0.25
-18 dBm
Preemption Tone
425 Hz
0.5 0.17, 0.17 0.17
-18 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Ringback Tone
425 Hz
1.0 4.0
-16 dBm
900 Hz
1400 Hz
1800 Hz
1.0 - 1.0 - 1.0 - 1.0

1.0 - 1.0 - 1.0 - 1.0
1.0 - 1.0 - 1.0 - 1.0
-14 dBm
-14 dBm
-14 dBm
Stutter Dial Tone
425 Hz
-14 dBm
Ring
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
1.0 4.0
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
600
600
Tbrl-impedance
600
FED Tone
425 Hz
Default Caller ID
BELLCORE
8.0
-3 dBr
-3 dBr
0 dBr
0 dBr
0 second

Tone Detection)
0 second
637
Netherlands
Netherlands
The following parameters apply if you have selected Netherlands1 as location.
Table 530: Netherlands1 Parameters
Parameter
Value
Elect. Levels
Busy Tone
425 Hz
0.5 0.5
-17 dBm
Call Waiting Tone
440 Hz
2.0 0.3 10.0 0.3 10.0
-17 dBm
Confirmation Tone
425 Hz
(0.1 0.1) x 3, End
-17 dBm
Dial Tone
425 Hz
CONTINUOUS
-17 dBm
425 Hz
-17 dBm
425 Hz
0.25 0.25
-17 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Ringback Tone
425 Hz
1.0 4.0
-17 dBm
950 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-17 dBm
-17 dBm
-17 dBm
Stutter Dial Tone
425 Hz
-17 dBm
Ring
AC: 45 VRMS, 25Hz

DC: -15 Vdc
1.0 4.0
Loop Current
30 ma
600
270 + 750 // 150 nF
Tbrl Impedance
340 + 422 // 100 nF
FED Tone
425 Hz
Bell 202
8.0
+0 dBr
-7 dBr
+6 dBr
-1 dBr
638
0 seconds

Tone Detection)
4 seconds
New Zealand
New Zealand
The following parameters apply if you have selected New Zealand1 as location.
Table 531: NewZealand1 Parameters
Parameter
Value
Elect. Levels
Busy Tone
400 Hz
0.5 0.5
-17 dBm
Call Waiting Tone
440 Hz
2.0 0.3 10.0 0.3 10.0
-17 dBm
Confirmation Tone
400 Hz
(0.1 0.1) x 3, End
-17 dBm
Dial Tone
400 Hz
CONTINUOUS
-17 dBm
Intercept Tone
1400 Hz
0.4 4.0
-17 dBm
400 Hz
-17 dBm
400 Hz
0.25 0.25
-17 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Reorder Tone
400 Hz
0.07 0.1 0.07 0.1 0.07 0.1 0.07 0.4
-17 dBm
Ringback Tone
400+450 Hz
0.4 0.2 0.4 2.0
-19 dBm
1400 Hz
0.1 0.1
-17 dBm
Stutter Dial Tone
400 Hz
-17 dBm
Ring
AC: 45 VRMS, 25Hz

DC: -15 Vdc
0.4 0.2 0.4 2.0
Loop Current
30 ma
370 + 620 // 310 nF
370 + 620 // 310 nF
Tbrl Impedance
370 + 620 // 310 nF
FED Tone
400 Hz
Bell 202
8.0
-3 dBr
-9 dBr
+6 dBr
-1 dBr
0 seconds

Tone Detection)
0 seconds
639
North America
North America
The following parameters apply if you have selected North America as location.
North America 1
The following parameters apply if you have selected North America 1 as location.
Table 532: North America 1 Parameters
Parameter
Value
Elect. Levels
Busy Tone
480+620 Hz
0.5 0.5
-21 dBm
Call Waiting Tone
440 Hz
2.0 0.3, 10.0 0.3, 10.0
-17 dBm
Confirmation Tone
350+440 Hz
(0.1 0.1) x 3, End
-17 dBm
Dial Tone
350+440 Hz
CONTINUOUS
-17 dBm
Intercept Tone
440+620 Hz
0.5 0.5
-14 dBm
350+440 Hz
-17 dBm
480+620 Hz
0.25 0.25
-21 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Reorder Tone
480+620 Hz
0.3 0.2
-21 dBm
Ringback Tone
440+480 Hz
2.0 4.0
-19 dBm
950 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-14 dBm
Stutter Dial Tone
350+440 Hz
-17 dBm
Ring (FXS)
AC: 45 VRMS, 20 Hz
DC: -15 Vdc
2.0 4.0
Loop Current
30 ma
Min: 300 ms
Max: 1100 ms
600
600
Tbrl-Impedancea (FXS)
600
FED Tone
440 Hz
BELLCORE
8.0
-3 dBr
-3 dBr
0 dBr
0 dBr
0 seconds

Tone Detection)
0.7 seconds
a. TBRL-Impedance for on-premise or short loop ports.
640
Russia
Russia
The following parameters apply if you have selected Russia1 as location.
Table 533: Russia1 Parameters
Parameter
Value
Elect. Levels
Busy Tone
425 Hz
0.4 0.4
-10 dBm
Call Waiting Tone
440 Hz
2.0 0.3 10.0 0.3 10.0
-17 dBm
Confirmation Tone
425 Hz
(0.1 0.1) x 3, End
-10 dBm
Dial Tone
425 Hz
CONTINUOUS
-10 dBm
425 Hz
-10 dBm
425 Hz
0.2 0.2
-10 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Ringback Tone
425 Hz
0.8 3.2
-10 dBm
950 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-17 dBm
-17 dBm
-17 dBm
Stutter Dial Tone
425 Hz
-10 dBm
Ring
AC: 45 VRMS, 25Hz

DC: -15 Vdc
0.8 3.2
Loop Current
30 ma
600 + 2160
600
Tbrl Impedance
350 + 1000 // 210 nF
FED Tone
425 Hz
Bell 202
8.0
+2 dBr
-2 dBr
+0 dBr
+0 dBr
0 seconds

Tone Detection)
0 seconds
641
Spain
Spain
The following parameters apply if you have selected Spain1 as location.
Table 534: Spain1 Parameters
Parameter
642
Value
Elect. Levels
Busy Tone
425 Hz
0.2 0.2
-13 dBm
Call Waiting Tone
440 Hz
2.0 0.3, 10.0 0.3, 10.0
-17 dBm
Confirmation Tone
425 Hz
(0.1 0.1) x 3, End
-10 dBm
Dial Tone
425 Hz
CONTINUOUS
-10 dBm
425 Hz
-10 dBm
425 Hz
0.2 0.2, 0.2 0.2, 0.2 0.6
-13 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Reorder Tone
425 Hz
0.2 0.2, 0.2 0.6
-13 dBm
Ringback Tone
425 Hz
1.5 3.0
-13 dBm
950 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-20 dBm
-20 dBm
-20 dBm
Stutter Dial Tone
425 Hz
-10 dBm
Ring (FXS)
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
1.5 3.0
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
220 + 820 // 120 nF
270 + 750 // 150 nF
Tbrl-Impedance
220 + 820 // 120 nF
FED Tone
425 Hz
BELLCORE
8.0
0 dBr
-7 dBr
+6 dBr
-1 dBr
0 seconds

Tone Detection)
4 seconds
Sweden
Sweden
The following parameters apply if you have selected Sweden1 as location.
Table 535: Sweden1 Parameters
Parameter
Value
Elect. Levels
Busy Tone
425 Hz
0.25 0.25
-12.5 dBm
Call Waiting Tone
440 Hz
2.0 0.3 10.0 0.3 10.0
-17 dBm
Confirmation Tone
425 Hz
(0.1 0.1) x 3, End
-12.5 dBm
Dial Tone
425 Hz
CONTINUOUS
-12.5 dBm
425 Hz
-12.5 dBm
425 Hz
0.25 0.75
-12.5 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Ringback Tone
425 Hz
1.0 5.0
-12.5 dBm
950 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-22 dBm
-22 dBm
-22 dBm
Stutter Dial Tone
425 Hz
-12.5 dBm
Ring
AC: 45 VRMS, 25Hz

DC: -15 Vdc
1.0 5.0
Loop Current
30 ma
200 + 1000 // 100 nF
270 + 750 // 150 nF
Tbrl Impedance
900 // 30 nF
FED Tone
425 Hz
Bell 202
8.0
+0 dBr
-5 dBr
+2 dBr
-3 dBr
0 seconds

Tone Detection)
4 seconds
643
Switzerland
Switzerland
The following parameters apply if you have selected Switzerland as location.
Table 536: Switzerland Parameters
Parameter
644
Value
Elect. Levels
Busy Tone
425 Hz
0.5 0.5
-13 dBm
Call Waiting Tone
440 Hz
2.0 0.3, 10.0 0.3, 10.0
-17 dBm
Confirmation Tone
425 Hz
(0.1 0.1) x 3, End
-8 dBm
Dial Tone
425 Hz
CONTINUOUS
-8 dBm
425 Hz
-8 dBm
425 Hz
0.2 0.2
-13 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Ringback Tone
425 Hz
1.0 4.0
-13 dBm
950 Hz
1400 Hz
1800 Hz
0.33 - 0.33 - 0.33 - 1.0

0.33 - 0.33 - 0.33 - 1.0
0.33 - 0.33 - 0.33 - 1.0
-13 dBm
-13 dBm
-13 dBm
Stutter Dial Tone
425 Hz
-8 dBm
Ring (FXS)
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
1.0 4.0
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
220 + 820 // 115 nF
270 + 750 // 150 nF
Tbrl-impedance
220 + 820 // 115 nF
FED Tone
425 Hz
BELLCORE
8.0
0 dBr
-6.5 dBr
+6 dBr
-1 dBr
0 seconds

Tone Detection)
4 seconds
United Arab Emirates

The following parameters apply if you have selected United Arab Emirates as location.
United Arab Emirates 2

The following parameters apply if you have selected the United Arab Emirates 2 as location.
Table 537: United Arab Emirates 2 Parameters
Parameter
Value
Elect. Levels
Busy Tone
400 Hz
0.38 0.38
-13 dBm
Call Waiting Tone
425 Hz
(0.2 12.0, 0.2 12.0)x2 End
-13 dBm
Confirmation Tone
400 Hz
(0.1 0.1) x 3 End
-13 dBm
Dial Tone
350+450 Hz
CONTINUOUS
-13 dBm
350+440 Hz
-13 dBm
400 Hz
0.4 0.35, 0.23 0.53
-13 dBm
1400+2060+2450+2600 Hz
(0.1 0.1)
-19 dBm
Reorder Tone
400 Hz
CONTINUOUS
-13 dBm
Ringback Tone
425 Hz
0.4 0.2, 0.4 2.0
-13 dBm
950 Hz
1400 Hz
1800 Hz
0.33 0.33, 0.33 1.0

0.33 0.33, 0.33 1.0
0.33 0.33, 0.33 1.0
-13 dBm
-13 dBm
-13 dBm
Stutter Dial Tone
350+450 Hz
(0.4 0.04-) x 5 CONTINUOUS
-13 dBm
Ring
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
0.4 0.2, 0.4 2.0
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
600
600
Tbrl-impedance
600
FED Tone
440 Hz
Default Caller ID
BELLCORE
8.0
-3 dBr
-3 dBr
+0 dBr
+0 dBr
0 seconds

Tone Detection)
0 seconds

Parameter
Busy Tone
Value
400 Hz

0.38 0.38
Elect. Levels
-19 dBm
645
Table 538: United Arab Emirates 3 Parameters (Continued)

Parameter
Value
Elect. Levels
Call Waiting Tone
440 Hz
2.0 0.3, 10.0 0.3, 10.0
-17 dBm
Confirmation Tone
350+440 Hz
(0.1 0.1) x 3 End
-22 dBm
Dial Tone
350+440 Hz
CONTINUOUS
-22 dBm
350+440 Hz
-22 dBm
400 Hz
0.4 0.35, 0.23 0.53
-19 dBm
1400+2060+2450+2600 Hz
(0.1 0.1)
-19 dBm
Reorder Tone
400 Hz
CONTINUOUS
-19 dBm
Ringback Tone
400+450 Hz
0.4 2.0, 0.4 0.2
-22 dBm
950 Hz
1400 Hz
1800 Hz
0.33 0.33, 0.33 1.0

0.33 0.33, 0.33 1.0
0.33 0.33, 0.33 1.0
-19 dBm
-19 dBm
-19 dBm
Stutter Dial Tone
350+440 Hz
-22 dBm
Ring
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
0.4 2.0, 0.4 0.2
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
600
600
Tbrl-impedance
600
FED Tone
440 Hz
Default Caller ID
BELLCORE
8.0
-3 dBr
-3 dBr
+0 dBr
+0 dBr
0 seconds

Tone Detection)
0 seconds

Parameter
646
Value
Elect. Levels
Busy Tone
400 Hz
0.38 0.38
-19 dBm
Call Waiting Tone
440 Hz
2.0 0.3, 10.0 0.3, 10.0
-17 dBm
Confirmation Tone
350+440 Hz
(0.1 0.1) x 3 End
-22 dBm
Dial Tone
350+440 Hz
CONTINUOUS
-13 dBm
350+440 Hz
-22 dBm
400 Hz
0.4 0.35, 0.23 0.53
-19 dBm
1400+2060+2450+2600 Hz
(0.1 0.1)
-19 dBm
Reorder Tone
400 Hz
CONTINUOUS
-19 dBm
Ringback Tone
400+450 Hz
0.4 2.0, 0.4 0.2
-22 dBm
Table 539: United Arab Emirates 4 Parameters (Continued)

Parameter
Value
Elect. Levels
950 Hz
1400 Hz
1800 Hz
0.33 0.33, 0.33 1.0

0.33 0.33, 0.33 1.0
0.33 0.33, 0.33 1.0
-19 dBm
-19 dBm
-19 dBm
Stutter Dial Tone
350+440 Hz
-22 dBm
Ring
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
0.4 2.0, 0.4 0.2
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
600
600
Tbrl-impedance
600
FED Tone
440 Hz
Default Caller ID
BELLCORE
8.0
-3 dBr
-3 dBr
+0 dBr
+0 dBr
0 seconds

Tone Detection)
0 seconds
647
UK
UK
The following parameters apply if you have selected the United Kingdom as location.
Table 540: UK Parameters
Parameter
648
Value
Elect. Levels
Busy Tone
400 Hz
0.38 0.38
-19 dBm
Call Waiting Tone
440 Hz
2.0 0.3, 10.0 0.3, 10.0
-17 dBm
Confirmation Tone
350+440 Hz
(0.1 0.1) x 3, End
-22 dBm
Dial Tone
350+440 Hz
CONTINUOUS
-22 dBm
350+440 Hz
-22 dBm
400 Hz
0.4 0.35, 0.23 0.53
-19 dBm
1400+2060+2450+2600 Hz
0.1 0.1
-19 dBm
Reorder Tone
400 Hz
CONTINUOUS
-19 dBm
Ringback Tone
400+450 Hz
0.4 0.2, 0.4 2.0
-22 dBm
950 Hz
1400 Hz
1800 Hz
0.33 0.33, 0.33 1.0

0.33 0.33, 0.33 1.0
0.33 0.33, 0.33 1.0
-19 dBm
-19 dBm
-19 dBm
Stutter Dial Tone
350+440 Hz
-22 dBm
Ring (FXS)
AC: 45 VRMS, 25 Hz
DC: -15 Vdc
0.4 0.2, 0.4 2.0
Loop Current
30 ma
Min: 170 ms
Max: 900 ms
300 + 1000 // 220 nF
320 + 1050 // 230 nF
Tbrl-impedance
370 + 620 // 310 nF
FED Tone
440 Hz
V23
8.0
-3 dBr
-9 dBr
+6 dBr
-1 dBr
0 seconds

Tone Detection)
4 seconds
P P E N D I X
Scripting Language
This appendix describes the Media5 proprietary scripting language. It also lists a few configuration samples
that can be pasted or typed into the CLI (see Chapter 2 - Command Line Interface (CLI) on page 19 for more
details) or downloaded into the Mediatrix via the Configuration Script feature (see Chapter 44 - Configuration
Script on page 531 for more details).
You can substitute the values listed in these examples with your own values. When enums are involved, refer
to the MIB structure with a MIB browser to determine the actual value you need to insert. You can also refer
to the Configuration Reference Guide, which lists all the parameters, tables, and commands available in the
Mediatrix.
This appendix covers the following topics:
General Scripting Language Syntax

Assigning scalar values
Assigning table cell values
Executing commands
Variable Values (Enums)
Call Router Specific Information
Examples

The Media5 proprietary scripting language can be used to assign values to configuration variables and
execute configuration commands. The scripting language may be used when creating configuration scripts
(Creating a Configuration Script on page 548) and when working with the Command Line Interface (Chapter
2 - Command Line Interface (CLI) on page 19).
Using the scripting language requires a bit of knowledge about the Mediatrixs configuration variables tree
structure.
The scripting language uses the following general syntax:
[keyword] [Context_Name [separator expression [operator constant]]] [#comment]
All specific syntaxes in this Appendix are derived from this general syntax.
Note that the brackets ([ and ]) are used to mark optional arguments. They are not part of the syntax.
Table 541: Scripting Language Syntax
Token Types
Description
keyword
A token that defines the type of operation to execute on expression or the type of data to
retrieve from expression. Currently, only the set keyword is supported, which assigns
value of constant to expression.
Context_Name
Defines to which service the following expression belongs. For instance, the
Configuration Manager context is Conf, the Firmware Pack Updater context is Fpu, and
the Host Configuration context is Hoc.
separator
Delimiter defined as . (dot).
649
Appendix B - Scripting Language
Table 541: Scripting Language Syntax (Continued)

Token Types
expression
Description
String that describes a configuration object. It can resolve to either:
a scalar variable
a cell
a column
a row
a table
When a service, a scalar, a command or a table variable has the same name as a
keyword, you should use the "get" and "set " keywords to access this variable.
operator
Only the assignment operator (=) is defined.
constant
A textual string or a number to assign to expression.
comment
Anything following the comment marker (#) up to the end of the line is ignored.
Supported Characters
When using the scripting language, the following ASCII codes are supported:
10
13
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
LF, line feed

CR, carriage return
space
!, exclamation mark
", double quote
#, hash
$, dollar
%, percent
&, ampersand
', quote
(, open parenthesis
), close parenthesis
*, asterisk
+, plus
,, comma
-, minus
., full stop
/, oblique stroke
0, zero
1
2
3
4
5
6
7
8
9
:, colon
;, semicolon
<, less than
=, equals
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
>,
?,
@,
A
B
C
D
E
F
G
H
I
J
K
L
M
N
O
P
Q
R
S
T
U
V
W
X
Y
Z
[,
\,
],
greater than
question mark
commercial at
open square bracket

backslash
close square bracket
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
^,
_,
`,
a
b
c
d
e
f
g
h
i
j
k
l
m
n
o
p
q
r
s
t
u
v
w
x
y
z
{,
|,
},
~,
caret
underscore
back quote
open curly bracket

vertical bar
close curly bracket
tilde
All other ASCII codes are invalid.

Note that you must escape the XML reserved characters when inserting them in a configuration script:
650
< : <
> : >
% : %
Assigning Scalar Values
& : &
Assigning Scalar Values

The following is a sample script command assigning a value to a scalar configuration variable:
Service_Name.Scalar_Name=value
Table 542: Scalar Syntax

Command
Description
Service_Name
Defines which service should process the expression. For instance, the
Configuration Manager context is Conf, the Firmware Pack Updater context is
Fpu, and the Host Configuration context is Hoc.
Scalar_Name
Name of the specific variable to which assign a value.
value
A textual string or a number to assign to the argument.
A valid script line would be:

Cli.InactivityTimeOut=25
Assigning Table Cell Values

When you want to get the value of a specific table cell or set the value of a specific table cell, you must follow
a particular syntax:
[get]Service_Name.Table_Name[Index=key].Column_Name
[set]Service_Name.Table_Name[Index=key].Column_Name=<value>
Table 543: Table Cell Syntax

Command
Description
Service_Name
Defines which service should process the expression. For instance, the
Table_Name
Name of the table that contains the cell.
index=key
List of index values identifying the row on which the cell is located. The list of
indexes is in the name=value form, separated by spaces and enclosed within
brackets. The index is always the first column of a table.
Column_Name
Name of the column that contains the cell.
value
Lets take for instance the NetworkInterfacesStatus table:

InterfaceName
Interface1
InterfaceStatus
100a
LinkName
eth1-4b
IpAddr
10.1.1.1
Int1c
Interface2
400d
eth1-4e
Int1
10.1.1.2
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Executing Commands
InterfaceName
Interface3
InterfaceStatus
LinkName
eth5g
400
IpAddr
10.1.1.3
eth1h
a. This enum means disabled
b. Mediatrix 3000 Series only
c. Mediatrix 4400 Series only
d. This enum means ok
e. Mediatrix 3000 Series only
f. Mediatrix 4400 Series only
g. Mediatrix 3000 Series only
h. Mediatrix 4400 Series only
If you want to get the IP address value of Interface 3, you would have to enter the following command:
get Bni.NetworkInterfacesStatus[InterfaceName=Interface3].IpAddr
Executing Commands
Configuration commands are used to make the Mediatrix perform actions such as restarting the unit, restarting
a service, refreshing its SIP registration, etc.
There are two types of commands you can execute:
Normal Commands
Row Commands
Normal Commands
The normal command feature has the following syntax:
Service_Name.Command_Name arg1=value1 b arg2=[value2 value3 value4]
Table 544: Normal Command Syntax

Command
Description
Service_Name
Defines which service should process the command. For instance, the
Command_Name
The command to execute.
argn
Name of the argument for which you want to assign a value. Three types of
arguments are allowed:
flags (beginning with -, without anything else, for instance, -b in the

command syntax above). Flags are optional.
scalar arguments (with mandatory = and following value). They are

mandatory unless they have a default value, in which case they are
optional.
vector arguments (with mandatory = and following a list of values

enclosed within brackets and separated by spaces). They are
mandatory unless they have a default value, in which case they are
optional.
The number and types of arguments depend on the specific command you are
using.
valuen
652
Executing Commands
For instance, a valid command would be:

Conf.BackupImage FileName=backup_test_1 Location=/testfiles/Conf/v1/ManualTests/
TransferProtocol=400 TransferUsername=testuser TransferPassword=test
TransferSrvHostname=test1.mediatrix.com
Another valid command (without arguments) would be:

SipEp.RegistrationRefresh
Double Quotes
You must use double quotes when the text parameter contains special characters such as dot or "#". For
instance, entering the following command results in a bad command:
Conf.BackupImage FileName=test.cfg Location=config TransferProtocol=400
TransferUsername=Usr1 TransferPassword=Pwd1 TransferSrvHostname=192.168.6.3
You must enclose each text parameter that contains special characters such as dot or "#" with double quotes.
In the above example, you must enclose FileName=test.cfg and TransferSrvHostname=192.168.6.3 in double
quotes:
Conf.BackupImage FileName="test.cfg" Location=config TransferProtocol=400
TransferUsername=Usr1 TransferPassword=Pwd1 TransferSrvHostname="192.168.6.3"
Row Commands
Row commands appear as table cells and allow you to perform an action on a specific row of the relevant table.
Row commands are available in several services of the Mediatrix. For instance, the Call Router service uses
the Up, Down, Insert, and Delete commands in its various tables.
The row command feature has the following syntax:
Context_Name.Table_Name[index1=value1 index2=value2].Row_Command=execute_value
Table 545: Row Command Syntax

Command
Description
Context_Name
Defines which service should process the command. For instance, the
Table_Name
Table where the row command is located.
indexn=valuen
List of index values identifying the row on which to execute the command. The
list of indexes is in the name=value form, separated by spaces and enclosed
within brackets. The index is always the first column of a table. See Assigning
Table Cell Values on page 651 for more details.
Row_Command
The row command to execute.
execute_value
Numerical value of the enum.
For instance, the following executes the service Dhcps StaticLeases Delete row command on one of the
tables rows. The StaticLeases table only has one index column: the MacAddress column. The Delete row
command is an enum that has two possible values: noOp (0) and delete (10). The command is executed by
assigning the execute value (10) to the Delete cell.
Dhcp.StaticLeases[MacAddress="0090F8001234"].Delete=10
DeleteAllRows Command
The DeleteAllRows command is a table command that you can use to delete all rows of a specific table to start
anew. You can use it as follows:
Service_Name.Table_Name.DeleteAllRows
A valid command would be:

CRout.MappingExpression.DeleteAllRows
653

The scripting language represents enums with their numeric value, and not their textual value. For instance,
the TFTP transfer protocol values available are as follows:
100
200
300
400
500
This does not mean much. By looking into the MIB structure of the Mediatrix with a MIB browser or requesting
help on the variable in the CLI, you will be able to determine that the values really mean the following:
100: HTTP
200: HTTPS
300: TFTP
400: FTP
500: FILE
Call Router Specific Information

When working with call router parameters, you must be aware of the following:
You must prefix the name of a route with route-, for instance: route-isdn_sip.
You must prefix the name of a SIP interface with sip-, for instance: sip-default.
You must prefix the name of an ISDN interface with isdn-, for instance: isdn-default.
You must prefix the name of a hunt with hunt-, for instance: hunt-hunt1.
Examples
This section gives a few configuration samples that can be used both in the CLI or as part of a configuration
script.
Basic Call
The following sections describe the minimal configuration you must apply to perform a basic call between an
ISDN telephone and an analog telephone.
Performing a Firmware Upgrade

The first step to do upon receiving a new Mediatrix is to upgrade its firmware to the latest version available.
Fpu.MfpVersion="1.1.8.73"
Fpu.MfpLocation=""
Fpu.MfpTransferProtocol=300
Fpu.MfpTransferUsername=""
Fpu.MfpTransferPassword=""
Fpu.MfpTransferSrvHostname="192.168.3.4"
Fpu.AutomaticRestartEnable=1
Fpu.Install
654
Examples
Configuration of the ISDN Interface

You must minimally configure the ISDN interface so that it can send and receive calls. If you have a Mediatrix
4400plus model, this includes enabling the ISDN power feeding feature to power your ISDN telephone.
-----------------------------------------------------------Mediatrix 4400 Series
-----------------------------------------------------------MbXgw.BriPorts[Name=Bri1].PowerFeedingEnable=1
Isdn.BasicRateInterface[Name=Bri1].EndpointType=200
Isdn.BasicRateInterface[Name=Bri1].ConnectionType=200
-----------------------------------------------------------Isdn.BasicRateInterface[Name=Slot2/E1T1].EndpointType=200
Isdn.BasicRateInterface[Name=Slot2/E1T1].ConnectionType=200
Configuration of the SIP Endpoint

Configuring the SIP endpoint allows you to register your ISDN telephone to a SIP server.
SipEp.DefaultStaticRegistrarServerHost=192.168.3.3
SipEp.DefaultStaticProxyHomeDomainHost=192.168.3.3
-----------------------------------------------------------sipEp.userAgent[EpId=Bri1].Username=4401
sipEp.userAgent[EpId=Bri1].FriendlyName=4401
sipEp.userAgent[EpId=Bri1].GatewayName="default"
sipEp.userAgent[EpId=Bri1].Publish=0
sipEp.userAgent[EpId=Bri1].Register=1
-----------------------------------------------------------sipEp.userAgent[EpId=Slot2/E1T1].Username=4401
sipEp.userAgent[EpId=Slot2/E1T1].FriendlyName=4401
sipEp.userAgent[EpId=Slot2/E1T1].GatewayName="default"
sipEp.userAgent[EpId=Slot2/E1T1].Publish=0
sipEp.userAgent[EpId=Slot2/E1T1].Register=1
SipEp.registrationrefresh
Configuration of the Call Router: Routes

You must create routes that will route calls from ISDN to SIP and from SIP to ISDN.
Crout.InsertRoute
Crout.Route[Index=1].SourceCriteria="Slot2/E1T1" (Mediatrix 3000 Series)
Crout.Route[Index=1].SourceCriteria="isdn-Bri1" (Mediatrix 4400 Series)
Crout.Route[Index=1].PropertiesCriteria=200
Crout.Route[Index=1].ExpressionCriteria=""
Crout.Route[Index=1].Destination="sip-default"
Crout.InsertRoute
Crout.Route[Index=2].SourceCriteria="sip-default"
Crout.Route[Index=2].PropertiesCriteria=200
Crout.Route[Index=2].ExpressionCriteria=""
Crout.Route[Index=2].Destination="Slot2/E1T1" (Mediatrix 3000 Series)
Crout.Route[Index=2].Destination="isdn-Bri1" (Mediatrix 4400 Series)
Crout.ApplyConfig
655
Examples
Configuration of the Call Router: Mapping

You must create mappings that will allow you to properly communicate between an ISDN telephone and an
analog telephone. In this example, it is assumed that the ISDN telephone number is 800 and the analog
telephone number is 101.
Crout.InsertMappingType
Crout.mappingtype[Index=1].Name="ISDN_To_SIP"
Crout.mappingtype[Index=1].Criteria=300
Crout.mappingtype[Index=1].Transformation=300
Crout.InsertMappingExpression
Crout.mappingexpression[Index=1].Name="ISDN_To_SIP"
Crout.mappingexpression[Index=1].Criteria=4400
Crout.mappingexpression[Index=1].Transformation=101
Crout.mappingexpression[Index=1].SubMappings=""
Crout.InsertMappingType
Crout.mappingtype[Index=2].Name="SIP_to_ISDN"
Crout.mappingtype[Index=2].Criteria=300
Crout.mappingtype[Index=2].Transformation=300
Crout.InsertMappingExpression
Crout.mappingexpression[Index=2].Name="SIP_to_ISDN"
Crout.mappingexpression[Index=2].Criteria=4400
Crout.mappingexpression[Index=2].Transformation=800
Crout.mappingexpression[Index=2].SubMappings=""
Crout.Route[Index=1].Mappings="ISDN_To_SIP
Crout.Route[Index=2].Mappings="SIP_to_ISDN
Crout.ApplyConfig
Management Functions
The following sections describe how to perform some useful management functions such as a configuration
backup/restore and changing the default user password.
Configuration Backup / Restore

Each of the two following commands must be created in one line.
Conf.BackupImage FileName="image.text" Location="resultfolder" TransferProtocol=300
TransferUsername="" TransferPassword="" TransferSrvHostname=192.168.3.4"
Conf.RestoreImage FileName="image.text" Location="resultfolder" TransferProtocol=300
TransferUsername="" TransferPassword="" TransferSrvHostname=192.168.3.4"
Configuration of a User Password

If you are using the CLI, the new password will be used the next time you connect to the Mediatrix.
Aaa.Users[UserName=public].Password=TestPwd
Debugging
The following sections allow you to enable two useful debugging tools of the Mediatrix: syslog messages and
PCM traces.
Enabling Syslog
This example assumes that you run a syslog server at address 192.168.3.4.
Nlm.SyslogRemoteHost=192.168.3.4
Cli.MinSeverity=300
Bni.MinSeverity=400
656
Examples
Hoc.MinSeverity=500
Configuring PCM Capture

The PCM traces are two different RTP streams made specifically to record all analog signals that are either
sent or received on the analog side of the Mediatrix. Only the configured port, port #1 and/or #2 are sending
the PCM traces for a maximum of four simultaneous RTP streams.
The RTP streams are sent to a configurable IP address, normally an IP address on your network where it can
be recorded with a packet sniffer (such as Wireshark). Moreover, they are independent from the regular RTP
streams of the VoIP call.
All streams are sent instantly at startup with an average ptime of 15 ms. This means that until the PCM traces
are disabled, even an idle unit will continuously send up to 66.6 packets/s X 4 streams = 267 packets/s using
approximately 174 bytes each, for a total of 46 Kbytes of upstream bandwidth.
Mipt.PcmCaptureEnable=1
Mipt.PcmCaptureEndpoint=Bri1-1
Mipt.PcmCaptureIpAddr=192.168.3.3
Mipt.restart
Advanced Configuration
The following sections describe more advanced configuration such as creating and enabling network
interfaces and configuring the ISDN interface of the Mediatrix.
Network Interfaces
Bni.pppServiceName="test"
Bni.pppRetryInterval=20
Bni.pppAuthenticationProtocol=100
Bni.pppIdentity="user"
Bni.pppSecret="pass"
Bni.networkInterfaces[InterfaceName=Lan1].LinkName=eth1-4 (Mediatrix 3000 Series)
Bni.networkInterfaces[InterfaceName=Lan1].ConnectionType=100 (Mediatrix 3000 Series)
Bni.networkInterfaces[InterfaceName=Lan1].StaticIpAddr=192.168.0.100/24 (Mediatrix
3000 Series)
Bni.networkInterfaces[InterfaceName=Lan1].Activation=1 (Mediatrix 3000 Series)
Bni.networkInterfaces[InterfaceName=Lan1].StatusInterfaceName=Loop (Mediatrix 3000
Series)
Bni.networkInterfaces[InterfaceName=Rescue].LinkName=eth1 (Mediatrix 4400 Series)
Bni.networkInterfaces[InterfaceName=Rescue].LinkName=eth1-4 (Mediatrix 3000 Series)
Bni.networkInterfaces[InterfaceName=Rescue].ConnectionType=100
Bni.networkInterfaces[InterfaceName=Rescue].StaticIpAddr=192.168.0.110/24
Bni.networkInterfaces[InterfaceName=Rescue].Activation=0
Bni.networkInterfaces[InterfaceName=Rescue].StatusInterfaceName=Loop
Bni.networkInterfaces[InterfaceName=Uplink].LinkName=eth1 (Mediatrix 4400 Series)
Bni.networkInterfaces[InterfaceName=Uplink].LinkName=eth5 (Mediatrix 3000 Series)
Bni.networkInterfaces[InterfaceName=Uplink].ConnectionType=100
Bni.networkInterfaces[InterfaceName=Uplink].StaticIpAddr=192.168.0.120/24
Bni.networkInterfaces[InterfaceName=Uplink].Activation=1
Bni.networkInterfaces[InterfaceName=Uplink].StatusInterfaceName=Loop
Bni.MinSeverity=100
ISDN Service Mediatrix 3400

Isdn.BasicRateInterface[Name=Slot2/Bri4].EndpointType=200
Isdn.BasicRateInterface[Name=Slot2/Bri4].ConnectionType=200
Isdn.BasicRateInterface[Name=Slot2/Bri4].NetworkLocation=500
Isdn.BasicRateInterface[Name=Slot2/Bri4].PreferredEncodingScheme=200
Isdn.BasicRateInterface[Name=Slot2/Bri4].FallbackEncodingScheme=100
Isdn.BasicRateInterface[Name=Slot2/Bri4].ChannelAllocationStrategy=400
Isdn.BasicRateInterface[Name=Slot2/Bri4].MaxActiveCalls=2
Isdn.BasicRateInterface[Name=Slot2/Bri4].SignalInformationElementEnable=1
Isdn.BasicRateInterface[Name=Slot2/Bri4].InbandToneGenerationEnable=0
Isdn.BasicRateInterface[Name=Slot2/Bri4].InbandDtmfDialingEnable=0
657
Examples
Isdn.BasicRateInterface[Name=Slot2/Bri4].OverlapDialingEnable=0
Isdn.BasicRateInterface[Name=Slot2/Bri4].CallingNameMaxLength=82
Isdn.BasicRateInterface[Name=Slot2/Bri4].ExclusiveBChannelSelectionEnable=1
Isdn.BasicRateInterface[Name=Slot2/Bri4].SendingCompleteEnable=0
Isdn.BasicRateInterface[Name=Slot2/Bri4].ClipEnable=1
Isdn.BasicRateInterface[Name=Slot2/Bri4].ClirEnable=1
Isdn.BasicRateInterface[Name=Slot2/Bri4].ClirOverrideEnable=1
Isdn.BasicRateInterface[Name=Slot2/Bri4].SendRestartOnStartupEnable=0
Isdn.BasicRateInterface[Name=Slot2/Bri4].HookFlashKeypad=*25
Isdn.BasicRateInterface[Name=Slot2/Bri4].KeypadReceptionTimeout=4
Isdn.BasicRateInterface[Name=Slot2/Bri4].SupplementaryServicesEnable=1
Isdn.BasicRateInterface[Name=Slot2/Bri4].Msn1=340401
Isdn.BasicRateInterfaceInterop[Name=Slot2/Bri4].ProgressIndicatorInSetupEnable=0
Isdn.BasicRateInterfaceInterop[Name=Slot2/Bri4].ProgressIndicatorInSetupAckEnable=0
Isdn.BasicRateInterfaceInterop[Name=Slot2/
Bri4].ProgressIndicatorInCallProgressForSetupEnable=0
Isdn.BasicRateInterfaceInterop[Name=Slot2/Bri4].ProgressIndicatorInAlertingEnable=0
Isdn.BasicRateInterfaceInterop[Name=Slot2/Bri4].ProgressIndicatorInConnectEnable=0
Isdn.BasicRateInterfaceInterop[Name=Slot2/Bri4].MaximumFacilityWaitingDelay=1500
Isdn.SignalingChannel[InterfaceName=Slot2/Bri4].Protocol=100
Isdn.SignalingChannel[InterfaceName=Slot2/Bri4].ColpEnable=1
Isdn.SignalingChannel[InterfaceName=Slot2/Bri4].ColrEnable=1
Isdn.SignalingChannel[InterfaceName=Slot2/Bri4].ColrOverrideEnable=1
Isdn.PhysicalLink[InterfaceName=Slot2/Bri4].L1TimerT3=30000
Isdn.PhysicalLink[InterfaceName=Slot2/Bri4].ClockMode=300
Isdn.MinSeverity=500
Isdn.Restart

Isdn.PrimaryRateInterface[Name=Slot2/E1T1].EndpointType=200
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].LineType=100
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].LineCoding=300
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].LineFraming=300
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].NetworkLocation=500
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].PreferredEncodingScheme=200
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].FallbackEncodingScheme=100
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].ChannelRange=1-15
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].ChannelAllocationStrategy=400
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].MaxActiveCalls=30
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].SignalInformationElementEnable=1
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].InbandToneGenerationEnable=0
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].InbandDtmfDialingEnable=0
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].OverlapDialingEnable=0
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].CallingNameMaxLength=82
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].ExclusiveBChannelSelectionEnable=1
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].SendingCompleteEnable=0
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].ClipEnable=1
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].ClirEnable=1
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].ClirOverrideEnable=1
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].SendRestartOnStartupEnable=0
Isdn.PrimaryRateInterface[Name=Slot2/E1T1].SupplementaryServicesEnable=1
Isdn.primaryRateInterfaceInterop[Name=Slot2/E1T1].ProgressIndicatorInSetupEnable=0
Isdn.primaryRateInterfaceInterop[Name=Slot2/E1T1].ProgressIndicatorInSetupAckEnable=0
Isdn.primaryRateInterfaceInterop[Name=Slot2/
E1T1].ProgressIndicatorInCallProgressForSetupEnable=0
Isdn.primaryRateInterfaceInterop[Name=Slot2/E1T1].ProgressIndicatorInAlertingEnable=0
Isdn.primaryRateInterfaceInterop[Name=Slot2/E1T1].ProgressIndicatorInConnectEnable=0
Isdn.primaryRateInterfaceInterop[Name=Slot2/E1T1].MaximumFacilityWaitingDelay=1500
Isdn.SignalingChannel[InterfaceName=Slot2/E1T1].Protocol=400
Isdn.SignalingChannel[InterfaceName=Slot2/E1T1].ColpEnable=1
Isdn.SignalingChannel[InterfaceName=Slot2/E1T1].ColrEnable=1
Isdn.SignalingChannel[InterfaceName=Slot2/E1T1].ColrOverrideEnable=1
Isdn.PhysicalLink[InterfaceName=Slot2/E1T1].L1TimerT3=30000
658
Examples
Isdn.PhysicalLink[InterfaceName=Slot2/E1T1].ClockMode=300
Isdn.Restart

Isdn.BasicRateInterface[Name=Bri4].EndpointType=200
Isdn.BasicRateInterface[Name=Bri4].ConnectionType=200
Isdn.BasicRateInterface[Name=Bri4].NetworkLocation=500
Isdn.BasicRateInterface[Name=Bri4].PreferredEncodingScheme=200
Isdn.BasicRateInterface[Name=Bri4].FallbackEncodingScheme=100
Isdn.BasicRateInterface[Name=Bri4].ChannelAllocationStrategy=400
Isdn.BasicRateInterface[Name=Bri4].MaxActiveCalls=2
Isdn.BasicRateInterface[Name=Bri4].SignalInformationElementEnable=1
Isdn.BasicRateInterface[Name=Bri4].InbandToneGenerationEnable=0
Isdn.BasicRateInterface[Name=Bri4].InbandDtmfDialingEnable=0
Isdn.BasicRateInterface[Name=Bri4].OverlapDialingEnable=0
Isdn.BasicRateInterface[Name=Bri4].CallingNameMaxLength=82
Isdn.BasicRateInterface[Name=Bri4].ExclusiveBChannelSelectionEnable=1
Isdn.BasicRateInterface[Name=Bri4].SendingCompleteEnable=0
Isdn.BasicRateInterface[Name=Bri4].ClipEnable=1
Isdn.BasicRateInterface[Name=Bri4].ClirEnable=1
Isdn.BasicRateInterface[Name=Bri4].ClirOverrideEnable=1
Isdn.BasicRateInterface[Name=Bri4].SendRestartOnStartupEnable=0
Isdn.BasicRateInterface[Name=Bri4].HookFlashKeypad=*25
Isdn.BasicRateInterface[Name=Bri4].KeypadReceptionTimeout=4
Isdn.BasicRateInterface[Name=Bri4].SupplementaryServicesEnable=1
Isdn.BasicRateInterface[Name=Bri4].Msn1=440401
Isdn.BasicRateInterfaceInterop[Name=Bri4].ProgressIndicatorInSetupEnable=0
Isdn.BasicRateInterfaceInterop[Name=Bri4].ProgressIndicatorInSetupAckEnable=0
Isdn.BasicRateInterfaceInterop[Name=Bri4].ProgressIndicatorInCallProgressForSetupEnable=
0
Isdn.BasicRateInterfaceInterop[Name=Bri4].ProgressIndicatorInAlertingEnable=0
Isdn.BasicRateInterfaceInterop[Name=Bri4].ProgressIndicatorInConnectEnable=0
Isdn.BasicRateInterfaceInterop[Name=Bri4].MaximumFacilityWaitingDelay=1500
Isdn.SignalingChannel[InterfaceName=Bri4].Protocol=100
Isdn.SignalingChannel[InterfaceName=Bri4].ColpEnable=1
Isdn.SignalingChannel[InterfaceName=Bri4].ColrEnable=1
Isdn.SignalingChannel[InterfaceName=Bri4].ColrOverrideEnable=1
Isdn.PhysicalLink[InterfaceName=Bri4].L1TimerT3=30000
Isdn.PhysicalLink[InterfaceName=Bri4].ClockMode=300
Isdn.Restart
659
660
Examples
P P E N D I X
Maximum Transmission Unit

(MTU)
This appendix describes the MTU (Maximum Transmission Unit) requirements of the Mediatrix.
What is MTU?
The Maximum Transmission Unit (MTU) is a parameter that determines the largest packet than can be
transmitted by an IP interface (without it needing to be broken down into smaller units). Each interface used
by TCP/IP may have a different MTU value specified.
The MTU should be larger than or equal to the largest packet you wish to transmit unfragmented. Note that
this only prevents fragmentation locally. Some other link in the path may have a smaller MTU: the packet will
be fragmented at that point, although some routers may refuse packets larger than their MTU.
Mediatrixs MTU
The Mediatrixs MTU is 1500 bytes, which is the Ethernet typical value.
Possible Hardware Problem

The implementation of the IEEE Standard 802.1q in the Mediatrix may have a minor problem because of
hardware limitations.
802.1q increases the Ethernet frame header by 4 bytes, adding a Virtual LAN ID and a user_priority. This is
useful to limit broadcasts that cross bridges, and it may also prioritize frames in the queuing algorithm of
switches. However, it also increases the maximum possible size of Ethernet frames from 1518 to 1522 bytes,
and this might not be handled adequately by every hardware.
A workaround is available for PCs running Windows to avoid sending 1522 bytes packets (note that this
happens only in special and rare cases). The workaround is to reduce the MTU of the interface (the one that
sends packets with 802.1q framing) by 4 bytes.
1.
Use the registry editor (regedt32) and go to the key:

Windows 2000 and later:
HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\Tcpip\Parameters\Interfaces
\<ethernet adapter>
Windows NT4 and 98:

\HKEY_LOCAL_MACHINE\SYSTEM\CurrentControlSet\Services\<ethernet
adapter>\Parameters\Tcpip
where <Ethernet adapter> can be found by using the command ipconfig /all.
2.
Add (or modify) a value named MTU of type REG_DWORD. Set it to 1496 (instead of 1500), in
decimal. Restart the computer to have those changes in effect.
In Windows 2000 and later this value is under the following key:
Key: Tcpip\Parameters\Interfaces\ID for Adapter2
661
Appendix C - Maximum Transmission Unit (MTU)
3.
Possible Hardware Problem
Value Type: REG_DWORD Number
Valid Range: 68 - the MTU of the underlying network
Default: 0xFFFFFFFF
Description: This parameter overrides the default MTU for a network interface. The
MTU is the maximum packet size in bytes that the transport will transmit over the
underlying network. The size includes the transport header. Note that an IP datagram
may span multiple packets. Values larger than the default for the underlying network
will result in the transport using the network default MTU. Values smaller than 68 will
result in the transport using an MTU of 68.
To validate that the changes are correct, try to ping the Mediatrix with large packets once restarted:
ping -l 2000
This will cause IP fragmentation, the first fragment being as large as the interface allows it. With the
MTU reduced, you should now receive an answer. For more informations, see:
http://support.microsoft.com/default.aspx?scid=kb;en-us;120642.
662
P P E N D I X
Web Interface SNMP Variables

Mapping
All parameters available in the Mediatrix web interface may also be configured via SNMP. The Mediatrix
SNMP feature offers the following options:
Remote management
This Appendix lists the mapping between the web interface fields and the corresponding SNMP variables of
the Mediatrix.
System Page
Information Sub-Page
Current Status Section
Field Name
System Description
SNMP Variable
unitInfoProductName
Description
Product name of the unit.
Firmware
mfpInstalledInfoMfpVersion
Version of the Firmware Pack installed.
Profile
mfpInstalledInfoMfpProfileName
Name of the profile.
MAC Address
unitInfoMacAddress
MAC address of the unit.
Serial Number
unitInfoSerialNumber
Serial number of the unit.
System Uptime
sysUpTime
Time since the last restart.
System Time
currentTimeSystem
Current date and time system configured in the unit.
Installed Hardware Section (Mediatrix 3000 Series only)

Field Name
SNMP Variable
Description
Name
hardwareExtInfoProductName
Name of the hardware extension.
Serial Number
hardwareExtInfoSerialNumber
Serial number of the hardware extension.
Location
hardwareExtInfoLocation
Location in the unit of the hardware extension.
Services Sub-Page
System Service
Field Name
SNMP Variable
Description
System Service
servicesInfoName
Current service name.
Status
servicesInfoExecState
Shows the execution state of the service.
663
Appendix D - Web Interface SNMP Variables Mapping
System Page
User Service
Field Name
SNMP Variable
Description
User Service
servicesInfoName
Current service name.
Status
servicesInfoExecState
Shows the execution state of the service.
Startup Type
servicesInfoStartupType
Selects the service startup type.
Action
serviceCommandsRestart
Restarts, starts or stops the service.
Comment
servicesInfoComment
Comments on the service's current state.
Start (command)
Starts the service.
Stop (command)
Stops the service.
Restart (command)
Restarts the service.
Restart Required Services

Field Name
Graceful Delay (min)
SNMP Variable
Description
graceDelay
The delay (in minutes) allowed for telephony calls to be all

completed.
CancelRestartRequiredServices (command)
Cancels the restart during the grace delay period.
RestartRequiredServices (command)
Restarts all required services
Hardware Sub-Page
Unit Configuration Section
Field Name
SNMP Variable
Description
Unit Configuration
portsConfiguration
Configures how each port provides a link interface.
Clock Reference
physicalLinkClockMode
A port can either generate the clocking for the line or accept
the clock from the line.
BRI Cards Configuration Section

Field Name
SNMP Variable
Description
Slot
physicalLinkInterfaceName
Identifies the interface.
Clock Reference
PRI Cards Configuration Section

Field Name
664
SNMP Variable
Description
Slot
physicalLinkInterfaceName
Identifies the interface.
Clock Reference
Indicates the preferred synchronisation source to use for the

internal clock of this digital card.
Line Type
physicalLinkLineCoding
Defines the transmission encoding of bits.
System Page
BRI Cards Configuration Section (Mediatrix 4400 Series only)

Field Name
Power Feeding
SNMP Variable
briPortsPowerFeedingEnable
Description
Enables/disables the ISDN power feeding for a given port.
Endpoints Sub-Page
Unit States Section
Field Name
SNMP Variable
Description
Administrative
unitAdminState
Indicates the current maintenance state of a unit.
Operational
unitOpState
The operational state of the unit reflects the unit's internal

state.
Usage
unitUsageState
The usage state of the unit indicates its running state.
Action
unitUnlock
unitLock
unitForceLock
Allows to use a unit.

Gracefully disallows to use a unit.
Forcefully disallows to use a unit.
Endpoint States Section

Field Name
SNMP Variable
Description
Endpoint
endpointEpId
String that identifies an endpoint in other tables.
Administrative
endpointAdminState
Administrative state of an endpoint.
Operational
endpointOpState
Operational state of an endpoint.
Usage
endpointUsageState
Ru7ning state of an endpoint.
Initial Administrative
endpointInitialAdminStateConfig
Initial administrative state of an endpoint.
Action
endpointUnlock
endpointLock
endpointForceLock
Allows using the endpoint.

Gracefully disallows using the endpoint.
Forcefully disallows using the endpoint.
Administration Section
Field Name
SNMP Variable
Description
Disable Unit When No

Gateways Are In State
Ready
unitDisabledWhenNoGatewayReadyEnable
Indicates if the unit operational state is automatically set to

disable when all signaling gateways are not ready.
Shutdown Endpoint
When Operational
State is 'Disable' And
Its Usage State Is 'idleunusable'
endpointAutomaticShutdownEnable
Indicates if an endpoint is physically shutdown when in the

'idle-unusable' usage state.
Syslog Sub-Page
Field Name
SNMP Variable
Description
Remote Host
syslogRemoteHost
Host name and port number of the device that archives log
entries.
Authentication,
Authorization and
Accounting (AAA)
minSeverity (aaaMIB)
Minimal Severity of Notification
Basic Network
Interface (BNI)
minSeverity (bniMIB)
Call Routing (CROUT)
minSeverity (cRoutMIB)
665
Field Name
System Page
SNMP Variable
Description
Certificate Manager
(CERT)
minSeverity (certMIB)
Command Line
Interface (CLI)
minSeverity (cliMIB)
Configuration Manager
(CONF)
minSeverity (confMIB)
Device Control
Manager (DCM)
minSeverity (dcmMIB)
DHCP (DHCP)
minSeverity (dhcpMIB)
Endpoint
Administration
(EpAdm)
minSeverity (epAdmMIB)
Endpoint Services
(EpServ)
minSeverity (epServMIB)
Ethernet manager
(Eth)
minSeverity (ethMIB)
Firmware Pack
Updater (FPU)
minSeverity (fpuMIB)
Host Configuration
(HOC)
minSeverity (hocMIB)
Integrated Services
Digital Network (ISDN)
minSeverity (isdnMIB)
Local Quality Of
Service (LQOS)
minSeverity (lQosMIB)
Media IP Transport
(MIPT)
minSeverity (miptMIB)
Notifications and
Logging Manager
(NLM)
minSeverity (nlmMIB)
Plain Old Telephony

System Lines service
(POTS)
minSeverity (potsMIB)
Process Control
Manager (PCM)
minSeverity (pcmMIB)
Service Controller
Manager (SCM)
minSeverity (scmMIB)
SIP ALG (SipAlg)
minSeverity (sipAlgMIB)
SIP Endpoint (SipEp)
minSeverity (sipEpMIB)
Diagnostic Traces
diagnosticTracesEnable
Enables traces allowing the Technical Assistance Centre to

further assist in resolving some issues.
Filter
diagnosticTracesFilter
Filter applied to diagnostic traces.
Events Sub-Page
Field Name
666
SNMP Variable
Description
Activation
eventsActivation
Current activation state for this system event.
Criteria
eventsCriteria
Expression an event must match in order to apply the

specified action.
Service
N/A
N/A
Notification
N/A
N/A
Action
eventsAction
Action to apply to the system event if the criteria matches.
Config Status
eventsConfigStatus
Configuration status of the row.
Network Page
Field Name
SNMP Variable
Description
InsertEvent (command)
Inserts a row to the EventsTable
eventsDelete
Deletes this row.
Local Log Sub-Page

Local Log Status Section
Field Name
SNMP Variable
Description
Maximum Number of
Entries
LocalLogMaxNbEntries
Maximum number of entries that the local log can contain.

When adding a new entry while the local log is full, the
oldest entry is erased to make room for the new one.
Number of Error
Entries
LocalLogNbErrorEntries
Current number of error entries in the local log.
Number of Critical
Entries
LocalLogNbCriticalEntries
Current number of critical entries in the local log.
Local Log Entries Section

Field Name
SNMP Variable
Description
Local Time
LocalTime
Local date and time at which the log entry was inserted.
Format is YYYY-MM-DD HH:MM:SS.
Severity
Severity
Severity of the log entry.
Service Name
ServiceTextkey
Textual identifier of the service that issued the log entry.
Service Key
ServiceNumkey
Numerical identifier of the service that issued the log entry.
Message Key
NotificationId
Numerical identifier of the notification message.
Message Content
Message
The readable content of the log message.
Network Page
Status Sub-Page
Interfaces Status Section
Field Name
SNMP Variable
Description
Interface
networkInterfaceStatusInterfaceName
Network interface name.
Link
networkInterfaceStatusLinkName
Name of the link interface associated with the network

interface.
IP Address
networkInterfacesStatusIpAddr
Current address and network mask of the network interface.
Default Router
networkInterfacesDefaultRouter
Current default gateway of the network interface.
Connection Uptime
networkInterfacesConnectionUptime
The time, in seconds, for which this IP interface has been

connected.
Status
uplinkInterfaceStatus
Operational status of the Uplink network interface.
VLAN Override
netorknterfacesStatusVlanOverrideEnable
Indicates if the VLAN ID of the current network interface has

been overridden by the values received from the LLDP
protocol.
667
Network Page
LLDP Status Section

Field Name
SNMP Variable
Description
Type
remoteMediaPolicyStateAppType
The type of application.
Vlan ID
remoteMediaPolicyStateVlanId
VLAN ID.
User Priority (802.1Q)
remoteMediaPolicyStatePriority
802.1Q User Priority.
DiffServ (DSCP)
remoteMediaPolicyStateDscp
DSCP (DiffServ).
Policy Flag
remoteMediaPolicyStatePolicyFlag
Indicates if an Endpoint Device wants to explicitly advertise

that the network policy for a specific application type is
required but is currently unknown.
Tagged Flag
remoteMediaPolicyStateTaggedFlag
The Tagged flag.
Host Status Section

Field Name
SNMP Variable
Description
Automatic
Configuration Interface
subnetsAutomaticConfigurationInterface
The network interface that provides the automatic

configuration (E.g.: DNS servers, NTP server, etc.) to this
subnet.
Host Name Configuration

Host Name
domainNamesInfoSubnetName
Name of the subnet.
Domain Name
domainNamesInfoDomainName
Indicates the subnet's current domain name.

IPv4 Default Gateway
defaultRoutersInfoDefaultRouter
Indicates the subnet's current default gateway.
IPv6 Default Gateway
DNS Configuration
Primary DNS
dnsServersInfoIpAddress1
Indicates the subnets' first DNS server.
Secondary DNS
Indicates the subnets' secondary DNS server.
Third DNS
Indicates the subnets' third DNS server.
Fourth DNS
Indicates the subnets' fourth DNS server.

SNTP Configuration
Primary SNTP Host
sntpServersInfoHostName1
Indicates the subnets' first NTP server.
Secondary SNTP Host
Indicates the subnets' second NTP server.
Third SNTP Host
Indicates the subnets' third NTP server.
Fourth SNTP Host
Indicates the subnets' fourth NTP server.
Advanced IP Routes Section

Field Name
668
SNMP Variable
Description
advancedIpRoutesStatusIPriority
Unique identifier of the row in the table.
Source Address
advancedIpRoutesStatusSourceAddress
Source address[/mask] criteria used to match the rule.
Source Link
advancedIpRoutesStatusSourceLink
Source link criteria used to match the rule.
Forward To Network
advancedIpRoutesStatusForwardToNetwork
Network on which the packet is forwarded.
State
advancedIpRoutesStatusStatus
Status of the rule.
Network Page
IPv4 Routes Section

Field Name
Link
SNMP Variable
ipRoutesStatus
Description
Link (interface) ID.
Destination
ipRoutesStatus
Destination IP address or network address.
Gateway
ipRoutesStatus
Specifies the gateway IP address.
Protocol
ipRoutesStatus
Identifies the entity that installed the route.
Firewall Section
Field Name
SNMP Variable
Description
networkRulesStatusPriority
Source Address
networkRulesStatusSourceAddress
Source address[/mask] criteria an incoming packet must

have to match this rule.
Source Port
networkRulesStatusSourcePort
Source port[-port] criteria an incoming packet must have to

match this rule.
Destination Address
networkRulesStatusDestinationAddress
Destination address[/mask] criteria an incoming packet must

Destination Port
networkRulesStatusDestinationPort
Destination port[-port] criteria an incoming packet must have

to match this rule.
Protocol
networkRulesStatusProtocol
Protocol criteria an incoming packet must have to match this

rule.
Connection State
networkRulesStatusConnectionState
Connection state associated with the incoming packet.
Action
networkRulesStatusAction
Action taken when this rule matches a packet.
Network Address Translation Section

Field Name
SNMP Variable
Description
sNatRulesStatusPriority
dNatRulesStatusPriority
Source Address
sNatRulesStatusSourceAddress
dNatRulesStatusSourceAddress
Source address[/mask] criteria an incoming packet must

Source Port
sNatRulesStatusSourcePort
dNatRulesStatusSourcePort
Source port[-port] criteria an incoming packet must have to

match this rule.
Destination Address
sNatRulesStatusDestinationAddress
dNatRulesStatusDestinationAddress
Destination address[/mask] criteria an incoming packet must

Destination Port
sNatRulesStatusDestinationPort
dNatRulesStatusDestinationPort
Destination port[-port] criteria an incoming packet must have

to match this rule.
Protocol
sNatRulesStatusProtocol
dNatRulesStatusProtocol
Protocol criteria an incoming packet must have to match this

rule.
New Address
sNatRulesStatusNewAddress
dNatRulesStatusNewAddress
New address[:port] applied to the source of the packet.
Host Sub-Page
General Configuration Section
Field Name
Automatic
SNMP Variable
automaticConfigurationInterface
Description
configuration used by the unit (e.g.: Default gateway, DNS
servers, NTP server, etc.).
669
Field Name
Network Page
SNMP Variable
Description
Automatic IPv4 config

source network:
automaticConfigurationInterface

configuration used by the unit (e.g.: Default gateway, DNS
servers, NTP server, etc.).
Automatic IPv6 config

source network
ipv6AutomaticConfigurationInterface
The network interface that provides the IPv6 automatic

configuration (Default Router, domain name, DNS servers
and NTP server) used by the unit.
Host Name Configuration Section

Field Name
SNMP Variable
Description
Domain Name
Configuration Source
domainNameConfigSource
Configuration source for the domain name.
Domain Name
staticDomainName
Static domain name.
Host Name
hostName
Systems host name.
Default Gateway Configuration Section

Field Name
SNMP Variable
Description
IPv4
defaultRouterConfigSource
Default Gateway
staticDefaultRouter
Configuration source for the default gateway.

Static default gateway address.
IPv6
defaultRouterConfigSource
Configuration source for the default gateway.
Default Gateway
staticDefaultRouter
Static default gateway address.
DNS Configuration Section

Field Name
SNMP Variable
Description
dnsServersConfigSource
Configuration source for the DNS servers.
Primary DNS
staticDnsServersIpAddress1
Secondary DNS
Third DNS
Fourth DNS
SNTP Configuration Section

Field Name
670
SNMP Variable
Description
sntpConfigSource
Configuration source for the SNTP parameters.
Primary SNTP
staticSntpServersHostName1
Secondary SNTP
Third SNTP
Fourth SNTP
Synchronization Period
sntpSynchronizationPeriod
Time interval between system time synchronization cycles.
Synchronization Period
On Error
sntpSynchronizationPeriodOnError
Time interval between retries after an unsuccessful request

to the SNTP server.
Network Page
Time Configuration Section

Field Name
Static Time Zone
SNMP Variable
staticTimeZone
Description
Specifies the time zone in which the system is located.
Interfaces Sub-Page
Interface Configuration Section
Field Name
SNMP Variable
Description
Interface
networkInterfacesInterfaceName
Network interface name.
Link
networkInterfacesLinkName
Name of the link interface associated with the network

interface.
Type
networkInterfacesConnectionType
Connection type of the network interface.
Static IP Address
networkInterfacesStaticIpAddr
IPv4 address and network mask of the network interface.
Static Default Router
networkInterfacesStaticDefaultRouter
IPv4 address of the default gateway for the network

interface when the ConnectionType is set to ipStatic.
Activation
networkInterfacesActivation
Attempts to activate the network interface.
AddNetwork (command)
Adds a new network interface.
networkInterfacesDelete
Deletes the network interface and removes it from the

system.
PPPoE Configuration Section

Field Name
SNMP Variable
Description
Service Name
pppServiceName
Name of the service requested to the access concentrator

when establishing the next PPPoE connection.
Protocol
pppAuthenticationProtocol
Authentication protocol to use for authenticating the system

to the PPP peer.
User Name
pppIdentity
Name that identifies the system to the PPP peer during the
authentication process.
Password
pppSecret
Secret that identifies the system to the PPP peer during the
authentication process.
LLDP Configuration Section

Field Name
SNMP Variable
Description
Network Interface
NetworkInterface
The network interface name on which LLDP should be

enabled.
Chassis ID
ChassisId
The address type to populate the chassis ID.
Override Network
Policy
OverrideNetworkPolicyEnable
Enables the LLDP-MED protocol override of the VLAN ID,

User Priority and DiffServ values.
Ethernet Link Configuration Section

Field Name
SNMP Variable
Description
Link
linksName
The name of the Ethernet link.
MTU
linksMtu
Configures the MTU (Maximum Transmission Unit) of a

specific Ethernet link.
802.1x Authentication
linksIeee8021XAuthentication
Configures the IEEE 802.1x authentication protocol

activation on the Ethernet link interface.
671
Field Name
Network Page
SNMP Variable
Description
EAP Username
eapUserName
Username used to authenticate each Ethernet link

interfaces during the IEEE 802.1x EAP-TLS authentication
process.
eapCertificationValidation
Level of validation used by the device to authenticate the

IEEE 802.1x EAP-TLS peer's certificate. This variable
controls also the criteria used to select the host certificate
sent during the authentitication handshake
EAP 802.1x Configuration Section

Field Name
EAP 802.1x Version
SNMP Variable
Ieee8021XVersion
Description
Configures the IEEE 802.1x version of the unit.
VLAN Sub-Page
Field Name
SNMP Variable
Description
Link
vlanLinkName
Name of the Ethernet link over which the VLAN interface is

built.
Id
vlanId
VLAN ID used by the VLAN interface.
Default User Priority
vlanDefaultUserPriority
Default User Priority value the interface uses when tagging

packets.
AddVlan (command)
Adds a new virtual LAN.
vlanDelete
Deletes the VLAN interface and removes it from the system.
Local Firewall Sub-Page

Field Name
Config Modified
SNMP Variable
configModifiedStatus
Description
Shows whether the configuration of the local firewall was
modified without being applied.
Local Firewall Configuration Section

Field Name
Default Policy
SNMP Variable
defaultPolicy
Description
Action taken when a packet doesn't match any rules.
Local Firewall Rules Section

Field Name
672
SNMP Variable
Description
localRulesPriority
Activation
localRulesActivation
Current state for this rule.
Source Address
localRulesSourceAddress
Source address of the incoming packet.
Source Port
localRulesSourcePort
Source port of the incoming packet.
Destination Address
localRulesDestinationAddress
Destination address of the incoming packet.
Destination Port
localRulesDestinationPort
Destination port of the incoming packet.
Protocol
localRulesProtocol
Protocol of the incoming packet.
Action
localRulesAction
Action that will be taken in the matching packets.
localRulesInsert
Inserts a new row before this row.
Network Page
Field Name
SNMP Variable
Description
localRulesDelete
Deletes this row.
localRulesDown
Moves the current row downside.
localRulesUp
Moves the current row upside.
IP Routing Sub-Page
Field Name
Config Modified
SNMP Variable
Description
Shows whether or not the Network Address Translation

configuration has been modified without being applied.
Rollback (command)
Rolls back the current configuration to the running

configuration as showed in the status.
IP Routing Configuration Section

Field Name
IPv4 Forwarding
SNMP Variable
ipv4ForwardingEnable
Description
Enables/disables IPv4 forwarding.
Advanced IP Routes Section

Field Name
SNMP Variable
Description
advancedIpRoutesPriority
Activation
advancedIpRoutesActivation
Activates this route.
Source Address
advancedIpRoutesSourceAddress
Specifies the source IP address criteria an incoming packet

must have to match this rule.
Source Link
advancedIpRoutesSourceLink
Specifies the source link criteria an incoming packet must

Forward to Network
advancedIpRoutesForwardToNetwork
Network on which to route the packet.
advancedIpRoutesInsert
advancedIpRoutesDelete
Deletes this row.
advancedIpRoutesDown
advancedIpRoutesUp
Static IP Routes Section

Field Name
SNMP Variable
Description
Index
staticIpRoutesIndex
Destination
staticIpRoutesDestination
Specifies the destination IP address criteria that an outgoing

packet must have to match this route.
Link
staticIpRoutesLink
Output link (interface) name.
Gateway
staticIpRoutesGateway
Specifies the IP address of the gateway used by the route.
insertStaticIpRoute (command)
Inserts a new row at the end of the StaticIpRoutes table.
staticIpRoutesDelete
Deletes this row.
673
Network Page
Network Firewall Sub-Page

Field Name
Config Modified
SNMP Variable
Description
Shows whether the configuration of the network firewall was

Rollback (command)

Network Firewall Configuration Section

Field Name
Default Policy
SNMP Variable
defaultPolicy
Description
Action taken when a packet does not match any rules.
Network Firewall Rules Section

Field Name
SNMP Variable
Description
networkRulesPriority
Activation
networkRulesActivation
Activates this rule.
Source Address
networkRulesSourceAddress
Source Port
networkRulesSourcePort
Destination Address
networkRulesDestinationAddress
Destination Port
networkRulesDestinationPort
Protocol
networkRulesProtocol
Connection State
networkRulesConnectionState
Connection state associated with the incoming packet.
Action
networkRulesAction
Action that will be taken in the matching packets.
networkRulesInsert
networkRulesDelete
Deletes this row.
networkRulesDown
networkRulesUp
NAT Sub-Page
Field Name
Config Modified
SNMP Variable
Description
Shows whether or not the Network Address Translation

configuration has been modified without being applied.
Rollback (command)

Source Network Address Translation Rules Section

Field Name
674
SNMP Variable
Description
sNatRulesPriority
Activation
sNatRulesActivation
Source Address
sNatRulesSourceAddress
Source Port
sNatRulesSourcePort
Destination Address
sNatRulesDestinationAddress
Network Page
Field Name
Destination Port
SNMP Variable
Description
sNatRulesDestinationPort
Protocol
sNatRulesProtocol
New Address
sNatRulesNewAddress
New address applied to the destination of the packet.
sNatRulesInsert
sNatRulesDelete
Deletes this row.
sNatRulesDown
sNatRulesUp
Destination Network Address Translation Rules Section

Field Name
SNMP Variable
Description
dNatRulesPriority
Activation
dNatRulesActivation
Source Address
dNatRulesSourceAddress
Source Port
dNatRulesSourcePort
Destination Address
dNatRulesDestinationAddress
Destination Port
dNatRulesDestinationPort
Protocol
dNatRulesProtocol
New Address
dNatRulesNewAddress
New address applyed to the destination of the packet.
dNatRulesInsert
dNatRulesDelete
Deletes this row.
dNatRulesDown
dNatRulesUp
DHCP Server Sub-Page

DHCP Server Status Section
Field Name
Subnet Status
SNMP Variable
Description
subnetsConfigStatus
Subnet configuration status.

Lease Time
leaseTimesInfoDefault
Indicates the subnet's current default lease time in seconds.
Default Gateway

DNS (Option 6)
Primary DNS
dnsServersInfoDns1
Secondary DNS:
dnsServersInfoDns2
Third DNS
dnsServersInfoDns3
Fourth DNS
dnsServersInfoDns4

NTP (Option 42)
Primary SNTP Host
ntpServersInfoNtp1
675
Field Name
Network Page
SNMP Variable
Description
Secondary SNTP Host
ntpServersInfoNtp2
Third SNTP Host
ntpServersInfoNtp3
Fourth SNTP Host
ntpServersInfoNtp4
DHCP Server Leases Section

Field Name
SNMP Variable
Description
MAC Address
assignedLeasesInfoMacAddress
MAC address of the host.
IP Address
assignedLeasesInfoIpAddress
IP Address of the host.
Subnet Name
assignedLeasesInfoSubnetName
Indicates on which subnet the host is located.
Time Left
assignedLeasesInfoLeaseTimeLeft
Indicates the lease time left in seconds.
DHCP Server Configuration Section

Field Name
SNMP Variable
Description
DHCP Server Enable
subnetsEnableSubnet
Enables the subnet configuration.
Start IP Address
subnetsStartAddress
Start address of the subnet range.
End IP Address
subnetsEndAddress
End address of the subnet range.
Automatic
subnetsAutomaticConfigurationInterface
Interface that will provide the automatic configuration to this

subnet.
Subnet Specific
specificLeaseTimesEnableConfig
Defines the lease time configuration to use for a specific

subnet.
Lease Time
defaultLeaseTime
specificLeaseTimesLeaseTime
Specifies the lease time (in seconds) default setting for all
subnets.
Specifies the subnet's specific lease time in seconds.

Enable Option
specificDomainNamesEnableOption
Enables the domain name option.
Subnet Specific
specificDomainNamesEnableConfig
Defines the domain name configuration to use for this

specific subnet.
defaultDomainNameConfigSource
specificDomainNamesConfigSource
Default configuration source of all subnets.

Subnet's domain name specific configuration source.
Domain Name
defaultStaticDomainName
specificDomainNamesStaticName
Default static domain name for all subnets.

Static Domain Name Configuration.

Enable Option
specificDefaultRoutersEnableOption
Enables the default gateway option.
specificDefaultRoutersConfigSource
The subnet's specific router configuration source.
Default Gateway
specificDefaultRoutersStaticRouter
Specifies the subnet's default gateway.
DNS (Option 6)
676
Enable Option
specificDnsServersEnableOption
Enables the DNS servers option.
Subnet Specific
specificDnsServersEnableConfig
Defines the DNS servers configuration to use for a specific

subnet.
defaultDnsServersConfigSource
specificDnsServersConfigSource
Default configuration source for the DNS servers of all

subnets.
DNS servers specific configuration source for the subnet.
Primary DNS
specificDnsServersStaticDns1
IP address of the first DNS server of the subnet.
Secondary DNS:
IP address of the second DNS server of the subnet.
Third DNS
IP address of the third DNS server of the subnet
Network Page
Field Name
SNMP Variable
Description
Fourth DNS
IP address of the fourth DNS server of the subnet.
Enable Option
specificNtpServersEnableOption
Enables the NTP servers option.
Subnet Specific
specificNtpServersEnableConfig
Defines the NTP servers configuration to use for a specific

subnet.
defaultNtpServersConfigSource
specificNtpServersConfigSource
Default configuration source for the NTP servers of all

subnets.
NTP servers specific configuration source for the subnet.
Primary NTP
specificNtpServersStaticNtp1
IP address of the first NTP server of the subnet.
NTP (Option 42)
Secondary NTP
IP address of the second NTP server of the subnet.
Third NTP
IP address of the third NTP server of the subnet.
Fourth NTP
IP address of the fourth NTP server of the subnet.

NBNS (Option 44)
Enable Option
specificNbnsServersEnableOption
Enable NBNS servers option.
Subnet Specific
specificNbnsServersEnableConfig
Defines the NBNS servers configuration to use for a specific

subnet.
Primary NBNS
specificNbnsServersStaticNbns1
IP address of the first NBNS server of the subnet.
Secondary NBNS
IP address of the second NBNS server of the subnet.
Third NBNS
IP address of the third NBNS server of the subnet.
Fourth NBNS
IP address of the fourth NBNS server of the subnet.
QoS Sub-Page
Differentiated Service Field Configuration Section
Field Name
SNMP Variable
Description
Default DiffServ (IPv4)
defaultDiffServ
Default Differentiated Services value used by the unit for all

generated packets.
Default Traffic Class

(IPv6)
defaultTrafficClass
Default Traffic Class value used by the unit for all generated
IPv6 packets.
Ethernet 802.1Q Tagging Configuration Section

Field Name
SNMP Variable
Description
Enable
ethernet8021QTaggingEnablePriorityTagging
Enables or disables user priority tagging on the interface.
Default User Priority
ethernet8021QTaggingDefaultUserPriority
Default User Priority value the interface uses when tagging

packets.
Service Class Configuration Section

Field Name
DiffServ (IPv4)
SNMP Variable
Description
serviceClassesDiffServ
Differentiated Services value for a specific service class.
Traffic Class (IPv6)
serviceClassesTrafficClass
Default Traffic Class value used in IPv6 packets.
User Priority
servicesClassesUserPriority
User priority for a specific service class.
Network Traffic Control Configuration Section

Field Name
Physical Link
SNMP Variable
linkBandwidthControlLinkName
Description
Name of the Ethernet link over which the bandwidth
limitation is applied.
677
Field Name
Egress Limit
ISDN Page
SNMP Variable
linkBandwidthControlEgressLimit
Description
Indicates the bandwidth limitation for the selected link
interface.
ISDN Page
Field Name
Select Interface
SNMP Variable
primaryRateInterfaceName
basicRateInterfaceName
Description
Name that identifies the interface.
Status Sub-Page
Automatic Configuration Section
Field Name
Status
SNMP Variable
autoConfigureStatus
Description
Indicates the current status of the ISDN automatic
configuration mechanism.
Last Result
lastAutoConfigureResult
Result of the last ISDN automatic configuration.
Start Sensing (button)
AutoConfigure (command)
Auto-detect and auto-configure ISDN interfaces.
ISDN Preset Configuration Section

Field Name
SNMP Variable
Description
SNMP Variable
Description
Local Preset
Preset Name
Interface Status Section

Field Name
Signalling Channel
signallingChannelInfoState
Current call control state for this channel.
Bearer Group
bearerChannelInfoState
Current call control state for this channel.
Statistics Section (PRI)

Field Name
678
SNMP Variable
Description
TxFrames
statsInfoFramesTransmitted
RxFrames
statsInfoFramesReceived
Number of HDLC frames received.
TxOctets
statsInfoOctetsTransmitted
Number of octets transmitted.
RxOctets
statsInfoOctetsReceived
Number of octets received.
FcsErrors
statsInfoFcsErrors
Frame check sequence (FCS) errors indicate that frames of

data are being corrupted during transmission.
FramesDropped
statsInfoFramesDropped
Number of frames dropped.
OctetsDropped
statsInfoOctetsDropped
Number of octets dropped.
NegFrameSlipsTx
statsInfoNegativeFrameSlipsTransmitted
Negative Frame Slips Transmitted
NegFrameSlipsRx
statsInfoNegativeFrameSlipsReceived
Negative Frame Slips Received
ISDN Page
Field Name
SNMP Variable
Description
PosFrameSlipsTx
statsInfoPositiveFrameSlipsTransmitted
Positive Frame Slips Transmitted
PosFrameSlipsRx
statsInfoPositiveFrameSlipsReceived
Positive Frame Slips Received
FramingErrors
statsInfoFramingErrors
The framing error count indicates that a FAS (Frame

Alignment Signal) word has been received with an error.
CodeViolations
statsInfoCodeViolations
The code violations count indicates that an encoding error

on the PCM line has been detected.
CRCErrors
statsInfoCRCErrors
The CRC errors count is incremented when a multiframe

has been received with a CRC error.
EBitErrors
statsInfoEBitErrors
The E-Bit error count gives information about the outgoing

transmit PCM line if the E-bits are used by the remote end
for submultiframe error indication.
BlockErrors
statsInfoBlockErrors
The Block Error count is incremented once per multiframe if

a multirame has been received with a CRC error or a bad
frame alignment has been detected.
Interface Configuration Section (PRI)

Field Name
SNMP Variable
Description
Endpoint Type
primaryRateInterfaceEndpointType
Defines the endpoint type.
Clock Mode
Defines the clock mode of the interface.
Port Pinout
primaryRateInterfacePortPinout
Defines the port pinout.
Monitor Link State
physicalLinkMonitorLinkStateEnable
Monitors the physical link state of an ISDN interface.
Line Coding
primaryRateInterfaceLineCoding
Line encoding scheme to use.
Line Framing
primaryRateInterfaceLineFraming
Defines the frame format.
Signalling Protocol
signalingChannelProtocol
Protocol to use for the signalling channel.
Network Location
primaryRateInterfaceNetworkLocation
Value of the network location in the progress indicator

messages sent by the unit.
Preferred Encoding
Scheme
primaryRateInterfacePreferredEncodingSchem
e
Preferred data encoding scheme in the bearer capabilities.
Fallback Encoding
Scheme
primaryRateInterfaceFallbackEncodingScheme
Fallback data encoding scheme in the bearer capabilities.
Channel Range
primaryRateInterfaceChannelRange
Range of active bearer channels.
Channels Reserved for

Incoming Calls
primaryRateInterfaceIncomingChannelRange
Channel range reserved for incoming calls
Channels Reserved for

Outgoing Calls
primaryRateInterfaceOutgoingChannelRange
Channel range reserved for outgoing calls
Channel Allocation
Strategy
primaryRateInterfaceChannelAllocationStrategy
Strategy for selecting bearer channels.
Maximum Active Calls
primaryRateInterfaceMaxActiveCalls
Maximum number of active calls on the interface.
Signal Information
Element
primaryRateInterfaceSignalInformationElement
Enable
Defines whether the signal information element is enabled

on the interface.
Inband Tone
Generation
primaryRateInterfaceInbandTonesEnable
Defines whether inband tones are enabled.
Inband DTMF Dialing
primaryRateInterfaceInbandDtmfDialingEnable
Defines whether inband DTMF dialing is enabled.
Overlap Dialing
primaryRateInterfaceOverlapDialingEnable
Defines whether overlap dialing is enabled.
Calling Name Max

Length
primaryRateInterfaceCallingNameMaxLength
Maximum length of the calling party name for calls from SIP
to ISDN.
Exclusive B-Channel
Selection
primaryRateInterfaceExclusiveBChannelSelecti
onEnable
Defines whether exclusive B-Channel selection is enabled

for calls from SIP to ISDN.
Sending Complete
primaryRateInterfaceSendingCompleteEnable
Defines whether sending complete information element into

SETUP messages is enabled for calls from SIP to ISDN.
679
Field Name
Send Restart On
Startup
ISDN Page
SNMP Variable
Description
primaryRateInterfaceSendRestartOnStartupEna
ble
Defines whether the RESTART message is sent upon a

signaling channel "UP" event.
Link Establishment
signalingChannelLinkEstablishment
Sets the link establishment strategy.
Accepted Status
Causes
signalingChannelAcceptedStatusCauses
Sets the STATUS causes that can be received without

automatically clearing the call.
Accepted Progress
Causes
signalingChannelAcceptedProgressCauses
Sets the PROGRESS causes that can be received without

automatically disconnecting the call.
Send Isdn Progress
signalingChannelSendIsdnProgress
Sets the strategy for sending ISDN Progress messages.
Send Progress
Indicator IE
signalingChannelSendProgressIndicatorIE
Sets the strategy for sending the Progress Indicator

Information Element.
Default TON for Calling

Party Number IE
signalingChannelDefaultCallingTon
Default value to insert in the "Type of Number" parameter of

the "Calling Party Number" IE when "Type of Number" is not
already defined in the call properties.
Default NPI for Calling

Party Number IE
signalingChannelDefaultCallingNpi
Default value to insert in the "Numbering Plan Identification"

parameter of the "Calling Party Number" IE when
"Numbering Plan Identification" is not already defined in the
call properties.
Default PI for Calling

Party Number IE
signalingChannelDefaultCallingPi
Default value to insert in the "Presentation Indicator"

"Presentation Indicator" is not already defined in the call
properties.
Default SI for Calling

Party Number IE
signalingChannelDefaultCallingSi
Default value to insert in the "Screening Indicator"

"Screening Indicator" is not already defined in the call
properties.
Default TON for Called

Party Number IE
signalingChannelDefaultCalledTon

the "Called Party Number" IE when "Type of Number" is not
Default NPI for Called

Party Number IE
signalingChannelDefaultCalledNpi

parameter of the "Called Party Number" IE when
call properties.
Notification User
Suspended
signalingChannelUserSuspendedHandling
Defines the unit's behaviour when it receives a Notification

Indicator IE with the description set to User suspended.
Interface Configuration Section (BRI)

Field Name
Endpoint Type
680
SNMP Variable
basicRateInterfaceEndpointType
Description
Defines the endpoint type.
Clock Mode
Defines the clock mode of the interface.
Monitor Link State
Monitors the physical link state of an ISDN interface.
Connection Type
basicRateInterfaceConnectionType
Defines the connection type.
Signalling Protocol
signalingChannelProtocol
Protocol to use for the signalling channel.
Network Location
basicRateInterfaceNetworkLocation
Value of the network location in the progress indicator

messages sent by the unit.
Preferred Encoding
Scheme
basicRateInterfacePreferredEncodingScheme
Preferred data encoding scheme in the bearer capabilities.
Fallback Encoding
Scheme
basicRateInterfaceFallbackEncodingScheme
Fallback data encoding scheme in the bearer capabilities.
Channel Allocation
Strategy
basicRateInterfaceChannelAllocationStrategy
Strategy for selecting bearer channels.
basicRateInterfaceMaxActiveCalls
Maximum number of active calls on the interface.
Signal Information
Element
basicRateInterfaceSignalInformationElementEn
able
Defines whether the signal information element is enabled

on the interface.
ISDN Page
Field Name
Inband Tone
Generation
SNMP Variable
basicRateInterfaceInbandTonesEnable
Description
Defines whether inband tones are enabled.
Inband DTMF Dialing
basicRateInterfaceInbandDtmfDialingEnable
Defines whether inband DTMF dialing is enabled.
Overlap Dialing
basicRateInterfaceOverlapDialingEnable
Defines whether overlap dialing is enabled.
Calling Name Max

Length
basicRateInterfaceCallingNameMaxLength
Maximum length of the calling party name for calls from SIP
to ISDN.
Exclusive B-Channel
Selection
basicRateInterfaceExclusiveBChannelSelection
Enable
Defines whether exclusive B-Channel selection is enabled

for calls from SIP to ISDN.
Sending Complete
basicRateInterfaceSendingCompleteEnable
Defines whether sending complete information element into

SETUP messages is enabled for calls from SIP to ISDN.
Send Restart On
Startup
basicRateInterfaceSendRestartOnStartupEnabl
e
Defines whether the RESTART message is sent upon a

signaling channel "UP" event.
Link Establishment
signalingChannelLinkEstablishment
Sets the link establishment strategy.
Hook-Flash Keypad
basicRateInterfaceHookFlashKeypad
The actual keypad string that is to be considered as a hookflash.
Accepted Status
Causes
signalingChannelAcceptedStatusCauses
Sets the STATUS causes that can be received without

automatically clearing the call.
Send Isdn Progress
signalingChannelSendIsdnProgress
Sets the strategy for sending ISDN Progress messages.
Send Progress
Indicator IE
signalingChannelSendProgressIndicatorIE
Sets the strategy for sending the Progress Indicator

Information Element.
TEI Negotiation
basicRateInterfaceTeiNegotiation
Sets the Terminal Endpoint Identifier (TEI) negotiation

strategy.
Default TON for Calling

Party Number IE
signalingChannelDefaultCallingTon

the "Calling Party Number" IE when "Type of Number" is not
Default NPI for Calling

Party Number IE
signalingChannelDefaultCallingNpi

call properties.
Default PI for Calling

Party Number IE
signalingChannelDefaultCallingPi
Default value to insert in the "Presentation Indicator"

"Presentation Indicator" is not already defined in the call
properties.
Default SI for Calling

Party Number IE
signalingChannelDefaultCallingSi
Default value to insert in the "Screening Indicator"

"Screening Indicator" is not already defined in the call
properties.
Default TON for Called

Party Number IE
signalingChannelDefaultCalledTon

the "Called Party Number" IE when "Type of Number" is not
Default NPI for Called

Party Number IE
signalingChannelDefaultCalledNpi

parameter of the "Called Party Number" IE when
call properties.
Notification User
Suspended
signalingChannelUserSuspendedHandling
Defines the unit's behaviour when it receives a Notification

Indicator IE with the description set to User suspended.
Interop Configuration Section

Field Name
SNMP Variable
Description
Progress Indicator In
Setup
basicRateInterfaceInteropProgressIndicatorInS
etupEnable (BRI)
primaryRateInterfaceInteropProgressIndicatorIn
SetupEnable (PRI)
Defines whether or not the Progress Indicator IE is present

in the SETUP message when acting as the originating side.
Setup Ack
basicRateInterfaceInteropProgressIndicatorInS
etupAckEnable (BRI)
SetupAckEnable (PRI)

in the SETUP ACK when acting as the terminating side.
681
Field Name
ISDN Page
SNMP Variable
Description
Call Proceeding
basicRateInterfaceInteropProgressIndicatorInC
allProgressForSetupEnable (BRI)
CallProgressForSetupEnable (PRI)

in the CALL PROCEEDING message in response to a
SETUP message when acting as the terminating side.
Progress Indicator in
Alerting
basicRateInterfaceInteropProgressIndicatorInAl
ertingEnable (BRI)
AlertingEnable (PRI)
Defines whether or not the Progress Indicator information

element is present in the ALERTING message.
Progress Indicator in
Connect
basicRateInterfaceInteropProgressIndicatorInC
onnectEnable (BRI)
ConnectEnable (PRI)
Defines whether or not the Progress Indicator information

element is present in the CONNECT message.
Maximum Facility
Waiting Delay (ms)
basicRateInterfaceInteropMaximumFacilityWaiti
ngDelay (BRI)
primaryRateInterfaceInteropMaximumFacilityW
aitingDelay (PRI)
Maximum amount of time to wait for a FACILITY message,

after receiving a SETUP message, before going on with
normal call processing.
Use Implicit Inband

Info
basicRateInterfaceInteropUseImplicitInbandInfo
Enable (BRI)
primaryRateInterfaceInteropUseImplicitInbandIn
foEnable (PRI)
Defines if a message with progress indicator No. 1 MUST be

considered as offering inband information available.
Allow TEI Broadcast in

PTP
basicRateInterfaceInteropAllowTeiBroadcastInP
tpEnable
Defines whether or not an ISDN message with a TEI

broadcast needs to be interpreted as a TEI 0 when the
connection type is 'PointToPoint'
Call Proceeding Delay
signalingChannelInteropCallProceedingDelay
Defines the maximum time to wait after receiving a SETUP

message before sending a CALL PROCEEDING message
and going on with normal call processing.
Calling Name Delivery
signalingChannelInteropCallingNameDelivery
Defines how the Calling Name is delivered.
Timer Configuration Section

Field Name
SNMP Variable
Description
Auto Cancel Timeout
signalingChannelAutoCancelTimeout
Time, in seconds, the endpoint rings before the call is

L1 Timer T3
physicalLinkL1TimerT3
Sets the value, in milliseconds (ms), of the Layer 1 Timer

T3.
Services Configuration Section

Field Name
682
SNMP Variable
Description
Facility Services
signalingChannelFacilityServicesEnable
Defines whether or not supplementary services FACILITY

messages should be accepted on the ISDN interface.
Calling Line
Information
Presentation
basicRateInterfaceClipEnable (BRI)
primaryRateInterfaceClipEnable (PRI)
Defines whether or not the calling number information

element is sent.
Calling Line
Information Restriction
basicRateInterfaceClirEnable (BRI)
primaryRateInterfaceClirEnable (PRI)
Defines whether or not the calling number restriction at the

originating ISDN side is activated.
Calling Line
Information Restriction
Override
basicRateInterfaceClirOverrideEnable (BRI)
primaryRateInterfaceClirOverrideEnable (PRI)
At the destination ISDN side, activated the delivery of the

Calling Number digits even if the Calling Party information
element's Presentation Indicator is set to "restricted".
Connected Line
Identification
Presentation
signalingChannelColpEnable
Defines the Connected Number Information Element at the

originating ISDN side.
Connected Line
Identification
Restriction
signalingChannelColrEnable
Defines the Connected Number Information Element

restriction at the destination ISDN side.
Connected Line
Identification
Restriction Override
signalingChannelColrOverrideEnable
Defines the Connected Number Information Element

restriction override at the originating ISDN side.
R2 Page (Mediatrix 3600 Series only)
Field Name
SNMP Variable
Description
Outgoing Notify
signalingChannelOutgoingNotifyEnable
Defines if NOTIFY messages can be sent.
Maintenance Service
Call Termination
signalingChannelMaintenanceServiceCallTermi
nation
Describes the call termination strategy after reception of a

service message requesting a maintenance on the
associated bearer channel.
Date/Time IE Support
DateTimeIeSupport
Defines if the optional Date/Time Information Element (IE)

can be sent in the CONNECT and SETUP messages issued
over BRI and PRI.
AOC-E Support
signalingChannelAocESupport
Send the total charge at the (E)nd of the call in AOC-E

messages.
AOC-D Support
signalingChannelAocDSupport
Send the current charge (D)uring the call in AOC-D

messages.
Forward Call Rerouting
signalingChannelCallReroutingBehavior
Defines how the call rerouting request received from the

private network side is supported.
MSN
basicRateInterfaceMsn
Enables the MSN supplementary service with a commaseparated list of numbers.
IP Sync Section
Clock Synchronization Status Section
Field Name
SNMP Variable
Description
Unit
unitStatus
Current status of the unit in regard to the Clock

Precision
tdmBusPrecisionStatus
Estimate of the drift in PPB (parts per billion), of the local

TDM clock with regards to the reference (master) clock.
Jitter
jitterStatus
On a Slave unit, this variable displays the jitter observed

over incoming synchronization packets.
Clock Synchronization Configuration Section

Field Name
SNMP Variable
Description
Port
port
Port on which the unit listens for and sends clock

synchronization packets.
Static Master Host
staticMasterHost
FQDN or IP address of the Master unit on which a Slave

Static Redundant
Master Host
redundantStaticMasterHost
FQDN or IP address of the redundant Master unit on which

a Slave attempts to synchronize when the StaticMasterHost
is unreachable.

Field Name
Select Interface
SNMP Variable
Description
Status Sub-Page
683

Field Name
Status
SNMP Variable
autoConfigureStatus
Description
Indicates the current status of the R2 automatic
Last Result
Result of the last R2 automatic configuration.
Auto-detect and auto-configure R2 interfaces.
R2 Preset Configuration Section

Field Name
SNMP Variable
Description
Local Preset
Preset Name
R2 Chanel Associated Signaling Config Sub-Page

Field Name
SNMP Variable
Description
Clock Mode
Port Pinout
physicalLinkPortPinout
Defines the port pinout
Monitor Link State
r2PhysicalLinkMonitorLinkStateEnable
Monitors the physical link state of an R2 interface.
Line Coding
Line Framing
physicalLinkLineFraming
Channel Range
r2ChannelRange
Defines the range of active bearer channels.
Channel Allocation
Strategy
r2ChannelAllocationStrategy
Defines the strategy for selecting bearer channels.
r2MaxActiveCalls
Defines the maximum number of active calls on the

interface.
Encoding Scheme
r2EncodingScheme
Defines the voice encoding scheme in the bearer

capabilities.
Line Signaling Protocol
r2LineSignaling
Defines the protocol to use for the line signaling.
Incoming Digit
Signaling
r2IncomingDigitSignaling
Defines the R2 incoming digit signaling method.
Outgoing Digit
Signaling
r2OutgoingDigitSignaling
Defines the R2 outgoing digit signaling method.
Country Selection
r2CountrySelection
Identifies the country.
Digit Attenuation
r2DigitAttenuation
Additional attenuation in dB for MFR2/DTMF digits

generation.
Signaling Sub-Page
Field Name
SNMP Variable
r2SignalingVariantsResetSpecific
Description
Resets specific settings of R2 signaling parameters with the
default country settings.
Country Section
Field Name
Country Selection
684
SNMP Variable
r2CountrySelection
Description
Identifies the current country selected in the R2 table for the
specified interface.
Field Name
Override Default
Country Settings
SNMP Variable
r2SignalingVariantsOverrideDefault
Description
Overrides the default setting of R2 signaling parameters for
the Country Selection and Interface Name.
R2 Signaling Variants Section

Field Name
SNMP Variable
Description
Bits CD
r2SignalingVariantsBitsCD
Settings of the C and D bits when the device transmits line

signals.
ANI Length
r2SignalingVariantsAniLength
Defines the length of ANI (Automatic Number Identification)

to be requested or sent.
DNIS Length
r2SignalingVariantsDnisLength
Specifies the length of the DNIS expected.
ANI Request
r2SignalingVariantsAniRequestEnable
Defines whether or not ANI should be requested.
Send ANI request after

nth DNIS Digits
r2SignalingVariantsSendAniRequestAfterDnisDi
gits
Defines the number of DNIS digits to be received before

sending the ANI request.
Collect Call Blocked

Enabled
r2SignalingVariantsCollectCallBlocked
Defines if the Collect Call Blocked Option is used.
ANI Category
r2SignalingVariantsCategoryAni
Specifies the group II forward signal to be sent on receiving

calling party category request.
Line Free Category
r2SignalingVariantsLineFreeCategory
Specifies the group B backward signal to be sent by the

incoming R2 register to indicate line free condition of the
destination party.
ANI Restricted
r2SignalingVariantsAniRestrictedEnable
Specifies the behaviour of the unit following the reception of

a reject request after sending the Send next digit (ANI)
request.
Incoming Decline
Method
r2SignalingVariantsIncomingDeclineMethod
Sets how to cancel a call attempt from R2 if the called party

rejects the call when in Seizure Acknowledged state (waiting
for answer).
Timers Sub-Page
Field Name
SNMP Variable
r2TimerVariantsResetSpecific
Description
Resets specific settings of R2 timers parameters with the
Country Section
Field Name
SNMP Variable
Description
Country Selection
r2CountrySelection

Override Default
Country Settings
r2TimerVariantsOverrideDefault
Overrides the default setting of R2 timers parameters for the

Country Selection and Interface Name.
R2 Timers Variants Section

Field Name
SNMP Variable
Description
Seizure Ack Timeout
r2TimerVariantsSeizureAckTimeout
Maximum time an outgoing R2 register waits for the seizure

acknowledgement signal after sending a seizure signal
Fault Seizure Ack

Timeout
r2TimerVariantsFaultSeizureAckTimeout
Maximum time an incoming R2 register waits for a seizure

acknowledge failure condition to clear.
Double Seizure
Timeout
r2TimerVariantsDoubleSeizureTimeout
Minimum time an outgoing R2 register waits after a double

seizure is recognized before releasing the connection.
Double Answer
Timeout
r2TimerVariantsDoubleAnswerTimeout
Maximum time an outgoing R2 register waits after receiving

a clear-backward signal before releasing the connection.
685
Field Name
SNMP Variable
Description
Answer Timeout
r2TimerVariantsAnswerTimeout
Maximum time the answer signal AB=01 is applied before

the clear backward signal AB=11 is sent.
ReAnswerTimeout
r2TimerVariantsReAnswerTimeout
Maximum time the clear backward signal AB=11 is applied

before the answer signal AB=01 is reapplied again.
Release Guard
Timeout
r2TimerVariantsReleaseGuardTimeout
Maximum time an incoming R2 register waits after sending

an idle line signal to prevent a new seizure of the line when
a clear forward line signal is received.
InterCall Guard
Timeout
r2TimerVariantsInterCallGuardTimeout
Maximum time an outgoing R2 register waits after receiving

an idle line signal before attempting a new seizure of the
line.
Congestion Tone
Guard Timeout
r2TimerVariantsCongestionToneGuardTimeout
Maximum time an incoming R2 register waits after sending a

congestion tone before sending a clear forward line signal
and transit to the idle state.
Unblocking Timeout
r2TimerVariantsUnblockingTimeout
Sets the unblocking timeout to avoid a too aggresive seizure

of the trunk.
Address Complete
Timeout
r2TimerVariantsAddressCompleteTimeout
The Address Complete Timeout is the maximum time that

the caller waits for a reception of AddressComplete Tone
after sending all ANI or DNIS Digits.
Wait Answer Timeout
r2TimerVariantsWaitAnswerTimeout
The maximal time that the caller waits for an Answer signal
(ANSW) after receiving a Group B Line Free Signal Tone.
Digit Complete
Timeout
r2TimerVariantsDigitCompleteTimeout
The maximal time that the caller waits for a Group I forward
tone after sending either a Group A next DNIS digit, next
ANI digit or next calling category Tone.
Wait GroupB
Response Complete
Timeout
r2TimerVariantsWaitGroupBResponseComplet
eTimeout
The maximal time that the caller waits for the confirmation of
the end of the compelled sequence after receiving a Group
B Signal Tone.
Wait Immediate
Response Complete
Timeout
r2TimerVariantsWaitImmediateResponseCompl
eteTimeout
The maximal time that the caller waits for the confirmation of
the end of the compelled sequence after receiving a Group
B Signal Tone.
Play Tone Guard

Timeout
r2TimerVariantsPlayToneGuardTimeout
The maximum time an incoming R2 registers waits after

receiving the confirmation of the reception of the Group B
Signal Tone before playing one of the calling tones in the
caller direction.
Accept Call Timeout
r2TimerVariantsAcceptCallTimeout
Time that the unit waits to accept a R2 call.
Clear Forward Guard

Timeout
r2TimerVariantsClearForwardGuardTimeout
The maximum time an outgoing R2 register waits after

sending a clear forward line signal before transiting to the
idle state.
Clear Backward Guard

Timeout
r2TimerVariantsClearBackwardGuardTimeout
The maximum time an incoming R2 register waits after

sending a clear backward line signal before sending the idle
line signal and transit in the idle state.
Fault On Answered
Guard Timeout
r2TimerVariantsFaultOnAnsweredGuardTimeou
t
The maximum time an outgoing R2 register waits for the

fault to clear before sending the clear forward line signal.
Fault On Clear
Backward Guard
Timeout
r2TimerVariantsFaultOnClearBackwardGuardTi
meout

Fault On Seize Ack

Guard Timeout
r2TimerVariantsFaultOnSeizeAckGuardTimeout

Fault On Seize Guard

Timeout
r2TimerVariantsFaultOnSeizeGuardTimeout

Decline Guard Timeout
r2TimerVariantsDeclineGuardTimeout
The maximum time the AB=10 release signal is applied

before the AB=11 clearback signal. This variable applies
when IncomingDelcineMethod is set to ClearBack while
declining a call.
Digit Timers Sub-Page

Field Name
SNMP Variable
r2DigitTimerVariantsResetSpecific
686
Description
Country Section
Field Name
SNMP Variable
Description
Country Selection
r2CountrySelection

Override Default
Country Settings
r2DigitTimerVariantsOverrideDefault

R2 Digit Timers Variants Section

Field Name
SNMP Variable
Description
MFC Pulse Inter Digit

Timeout
r2DigitTimerVariantsMfcPulseInterDigitTimeout
The minimum delay between the end of transmission of the

last signal of the compelled cycle and the start of the next
one.
MFC Pulse Min On

Timeout
r2DigitTimerVariantsMfcPulseMinOnTimeout
The minimum time a backward tone can be on from the

backward perspective.
MFC Max Sequence

Timeout
r2DigitTimerVariantsMfcMaxSequenceTimeout
The maximum time for a complete compelled signaling cycle

from the forward perspective.
MFC Max On Timeout
r2DigitTimerVariantsMfcMaxOnTimeout
The maximum time a forward tone can be on from the

forward perspective.
MFC Max Off Timeout
r2DigitTimerVariantsMfcMaxOffTimeout
The maximum time a forward tone can be off from the

forward perspective.
Link Timers Sub-Page

Field Name
SNMP Variable
r2LinkTimerVariantsResetSpecific
Description
Country Section
Field Name
SNMP Variable
Description
Country Selection
r2CountrySelection

Override Default
Country Settings
r2LinkTimerVariantsOverrideDefault

R2 Timers Variants Section

Field Name
SNMP Variable
Description
r2LinkTimerVariantsLinkActivationTimeout
The maximum time the unit waits for an activation indication

coming from the physical link.

Timeout
r2LinkTimerVariantsLinkActivationRetryTimeout
The maximum time the unit waits before attempting to

reestablish the physical link.
Tones Sub-Page
Field Name
SNMP Variable
r2ToneVariantsResetSpecific
Description
Resets specific settings of R2 tones parameters with the
687
Country Section
Field Name
SNMP Variable
Description
Country Selection
r2CountrySelection

Override Default
Country Settings
r2ToneVariantsOverrideDefault
Overrides the default setting of R2 tones parameters for the

R2 Tones Forward Groups Section

Field Name
SNMP Variable
Description
End of DNIS Tone
r2ToneVariantsFwdGroup1EndOfDnis
Specifies the forward Group 1 tone used to send after the

DNIS digits.
End of ANI Tone
r2ToneVariantsFwdGroup1EndOfAni
Specifies the forward Group 1 tone used to send after the

ANI digits.
Restricted ANI Tone
r2ToneVariantsFwdGroup1RestrictedAni
Specifies the forward Group 1 tone used to reject a query.
R2 Tones Backward Groups Section

Field Name
688
SNMP Variable
Description

Tone
r2ToneVariantsBwdGroupASendNextDnisDigit
Specifies the backward Group A tone used to request the

next DNIS digit
Send Previous DNIS

Digit Tone
r2ToneVariantsBwdGroupASendPreviousDnisD
igit

previous DNIS digit.
Switch to Group II
Tone
r2ToneVariantsBwdGroupASwitchToGroupII
Specifies the backward Group A tone used to request to the

caller a switch of Group II signals.
Network Congestion
Tone
r2ToneVariantsBwdGroupANetworkCongestion
Specifies the backward Group A tone to be sent when a

congestion network is detected.
Send Calling Party

Category Tone
r2ToneVariantsBwdGroupASendCallingPartyCa
tegory
Specifies the backward Group A tone sent when the

backward group requests the calling party category.
Immediate Accept
Tone
r2ToneVariantsBwdGroupAImmediateAccept
Specifies the backward Group A tone sent when the

backward group accepts the call immediately.
Send DNIS Digit N-2

Tone
r2ToneVariantsBwdGroupASendDnisDigitNMin
us2

previous - 1 DNIS digit.
Send DNIS Digit N-3

Tone
r2ToneVariantsBwdGroupASendDnisDigitNMin
us3

previous - 2 DNIS digit.
Repeat all DNIS Tone
r2ToneVariantsBwdGroupARepeatAllDnis
Specifies the backward Group A tone used to request all

DNIS digits.
Send Next ANI Digit

Tone
r2ToneVariantsBwdGroupASendNextAniDigit

next ANI digit.
Send Special
Information Tone
r2ToneVariantsBwdGroupBSendSit
Specifies the backward Group B tone used to send a special

information tone.
User Busy Tone
r2ToneVariantsBwdGroupBUserBusy
Specifies the backward Group B tone used to signal a user

busy.
Network Congestion
Tone
r2ToneVariantsBwdGroupBNetworkCongestion
Specifies the backward Group B tone used to signal a

network congestion.
Unassigned Number
Tone
r2ToneVariantsBwdGroupBUnassignedNumber
Specifies the backward Group B tone used to signal an

unassigned number.
Line Free with Charge

Tone
r2ToneVariantsBwdGroupBLineFreeCharge
Specifies the backward Group B tone used to signal that the

line is free and charge must be applied.
Line Free without

Charge Tone
r2ToneVariantsBwdGroupBLineFreeNoCharge

line is free and no charges must be applied.
Line Out of Order Tone
r2ToneVariantsBwdGroupBLineOutOfOrder

line is out of order.
E&M Page (Mediatrix 3500/3600 Series only)
Field Name
Changed Number
Tone
SNMP Variable
r2ToneVariantsBwdGroupBChangedNumber
Description
subscriber has changed number.
IP Sync Section
Field Name
SNMP Variable
Description
Unit
unitStatus

Precision

Jitter
jitterStatus


Field Name
SNMP Variable
Description
Port
port

Static Master Host
staticMasterHost

Static Redundant
Master Host

is unreachable.

Field Name
Select Interface
SNMP Variable
Description
Status Sub-Page
Field Name
SNMP Variable
Description
Status
autoConfigureStatus
Indicates the current status of the E&M automatic

Last Result
Result of the last E&M automatic configuration.
Auto-detect and auto-configure E&M interfaces.
E&M Preset Configuration Section

Field Name
SNMP Variable
Description
Local Preset
Preset Name
689
E&M Channel Associated Signaling Config Sub-Page

Field Name
SNMP Variable
Description
Clock Mode
Port Pinout
physicalLinkPortPinout
Defines the port pinout
Monitor Link State
Monitors the physical link state of an E&M interface.
Line Coding
Line Framing
physicalLinkLineFraming
Channel Range
eamChannelRange
Defines the range of active bearer channels.
Channel Allocation
Strategy
eamChannelAllocationStrategy
Defines the strategy for selecting bearer channels.
eamMaxActiveCalls
Defines the maximum number of active calls on the

interface.
Encoding Scheme
eamEncodingScheme
Defines the voice encoding scheme in the bearer

capabilities.
E&M Signaling Type

Selection
eamSignalingType
Identifies the E&M signaling type.
Digit Attenuation
eamDigitAttenuation
Additional attenuation in dB for MFR2/DTMF digits

generation.
Signaling Sub-Page
Field Name
SNMP Variable
eamSignalingVariantsResetSpecific
Description
Resets specific settings of E&M Signaling parameters with
the default signaling settings.
E&M Signaling Type Section

Field Name
SNMP Variable
Description
E&M Signaling Type

Selection
Override Default
Signaling Settings
eamSignalingVariantsOverrideDefault
Overrides the default settings of E&M Signaling parameters

for the SignalingType and InterfaceName.
E&M Signaling Variants Section

Field Name
690
SNMP Variable
Description
Bits BCD
eamSignalingVariantsBitsBCD
Settings of the B, C and D bits when the device transmits

line signals.
ANI Length
eamSignalingVariantsAniLength
Defines the length of ANI (Automatic Number Identification)

to be requested or sent.
DNIS Length
eamSignalingVariantsDnisLength
Specifies the length of the DNIS expected.
Incoming Register
Signaling
eamSignalingVariantsIncomingRegisterSignalin
g
Defines the incoming register signaling method.
Outgoing Register
Signaling
eamSignalingVariantsOutgoingRegisterSignalin
g
Defines the outgoing register signaling method.
Incoming Dial Map
eamSignalingVariantsIncomingDialMap
Defines the dial map expression to match against on the

incoming side.
Outgoing Dial Map
eamSignalingVariantsOutgoingDialMap
Defines the dial map format for the outgoing dial string.
Wait Wink
eamSignalingVariantsWaitWink
Defines whether or not the outgoing register should wait for

a wink before proceeding with digit transmission.
Field Name
SNMP Variable
Description
Wait Wink Ack
eamSignalingVariantsWaitWinkAck
Defines whether or not the outgoing register should wait for

a wink acknowledge after all digit reception.
Send Wink
eamSignalingVariantsSendWink
Defines whether or not the incoming register should send a

wink to notify the remote side that digit information can be
sent.
Send Wink Ack
eamSignalingVariantsSendWinkAck
Defines whether or not the incoming register should send a

wink to acknowledge the receipt of all digits.
Timers Sub-Page
Field Name
SNMP Variable
eamTimerVariantsResetSpecific
Description
Resets specific settings of E&M Timers parameters with the
default signaling type settings.

Field Name
SNMP Variable
Description
E&M Signaling Type

Selection
Override Default
Signaling Settings
eamTimerVariantsOverrideDefault

E&M Timers Variants Section

Field Name
SNMP Variable
Description
Backward Wait Pre

Wink Timeout
eamTimerVariantsBwdWaitPreWinkTimeout
Amount of time an incoming register waits before sending

the wink that acknowledges the seizure.
Backward Send Wink

Timeout
eamTimerVariantsBwdSendWinkTimeout
Duration of the wink signal applied by the incoming register

to signal seizure acknowledgment.
Backward Wait First

Digit Timeout
eamTimerVariantsBwdWait1stDigitTimeout
Maximum time an incoming register waits for the first

incoming digit after the line seizure.
Backward Clear
Backward Timeout
eamTimerVariantsBwdClearBackwardTimeout
The maximum time an incoming register waits after sending

a clear backward line signal before transiting to the idle
state.
Backward Digit
Complete Timeout
eamTimerVariantsBwdDigitCompleteTimeout
The maximum time the incoming register waits for the next
digit before considering the digit sequence as completed.
Forward Wait Wink

Timeout
eamTimerVariantsFwdWaitWinkTimeout
Maximum time an outgoing register waits for seizure

acknowledgement after seizing the line.
Forward Wait Max

Wink on Timeout
eamTimerVariantsFwdWaitMaxWinkOnTimeout
The maximum time an outgoing register waits for the seizure

acknowledgment wink to complete.
Forward Wait Pre Dial

Timeout
eamTimerVariantsFwdWaitPreDialTimeout
The amount of time an outgoing register waits after the wink

to start dialing.
Forward Wait Answer

Timeout
eamTimerVariantsFwdWaitAnswerTimeout
The maximum time an outgoing register waits for the call to

be answered.
Forward Clear Forward

Timeout
eamTimerVariantsFwdClearForwardTimeout
The maximum time an outgoing register waits after sending

a clear forward line signal before transiting to the idle state.
Release Guard
Timeout
eamTimerVariantsReleaseGuardTimeout
Maximum time a register waits after sending an idle line

signal to prevent a new seizure of the line.
691
Digit Timers Sub-Page

Field Name
SNMP Variable
eamDigitTimerVariantsResetSpecific
Description
Resets specific settings of E&M Digit Timers parameters
with the default signaling type settings.

Field Name
SNMP Variable
Description
E&M Signaling Type

Selection
Override Default
Signaling Settings
eamDigitTimerVariantsOverrideDefault

E&M Digit Timers Variants Section

Field Name
SNMP Variable
Description
KP On Timeout
eamDigitTimerVariantsKPOnTimeout
The time during which the MF R1 KP tone is on.
KP Off Timeout
eamDigitTimerVariantsKPOffTimeout
The duration of the pause after the MF R1 KP tone.
Link Timers Sub-Page

Field Name
SNMP Variable
eamLinkTimerVariantsResetSpecific
Description
Resets specific settings of E&M Link Timers parameters
with the default signaling settings.

Field Name
SNMP Variable
Description
E&M Signaling Type

Selection
Override Default
Signaling Settings
eamLinkTimerVariantsOverrideDefault

E&M Link Timers Variants Section

Field Name
SNMP Variable
Description
eamLinkTimerVariantsLinkActivationTimeout
The maximum time the unit waits for an activation indication

coming from the physical link.

Timeout
eamLinkTimerVariantsLinkActivationRetryTime
out
The maximum time the unit waits before attempting to

reestablish the physical link.
Tones Sub-Page
Field Name
SNMP Variable
eamToneVariantsResetSpecific
692
Description
Resets specific settings of E&M Tones parameters with the
default signaling settings.

Field Name
SNMP Variable
Description
E&M Signaling Type

Selection
Override Default
Signaling Settings
eamToneVariantsOverrideDefault

E&M Tones Variants Section

Field Name
SNMP Variable
Description
KP Tone
eamToneVariantsKpTone
Specifies the KP (start-of-pulsing) signal.
ST Tone
eamToneVariantsStTone
Specifies the ST (end-of-pulsing) signal.
IP Sync Section
Field Name
SNMP Variable
Description
Unit
unitStatus

Precision

Jitter
jitterStatus


Field Name
SNMP Variable
Description
Port
port

Static Master Host
staticMasterHost

Static Redundant
Master Host

is unreachable.
693
POTS Page
POTS Page
Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
Status Sub-Page
Line Status Section
Field Name
SNMP Variable
Description
ID
lineId
String that identifies a line in other tables.
Type
lineTypeStatus
The status POTS type of the line.
State
lineState
The current call control state for this channel.
FXO Line Status Section

Field Name
SNMP Variable
Description
ID
fxoLinkStateId
String that identifies a FXO line.
Link State
fxoLinkStateLinkState
Link state as reported by the link state verification

mechanism.
Config Sub-Page
Field Name
SNMP Variable
Description
Caller ID
Customization
CallerIdCustomization
Allows selecting the detection/generation method of caller

ID.
Caller ID Transmission
CallerIdTransmission
Allows selecting the transmission type of the caller ID.
Vocal Unit Information
VocalUnitInformation
Determines whether or not the unit's IP or MAC address or

firmware version number can be acquired using the *#*0,
*#*1, and *#*8 digit maps respectively.
FXS Config Sub-Page
694
POTS Page
FXS Configuration Section

Field Name
SNMP Variable
Description
Line Supervision Mode
fxsLineSupervisionMode
Determines how the power drop and line polarity are used to
signal the state of a line.
Disconnect Delay
fxsDisconnectDelay
Determines whether or not call clearing occurs as soon as

the called user is the first to hang up a received call.
Auto Cancel Timeout
fxsDefaultAutoCancelTimeout
Time, in seconds, the endpoint rings before the call is

Inband Ringback
fxsInbandRingback
Determines whether or not the FXS endpoint needs to

generate a ringback for incoming ringing call.
Shutdown Behavior
fxsShutdownBehavior
Determines the FXS endpoint behavior when it becomes

shut down.
Power Drop on
Disconnect Duration
fxsPowerDropOnDisconnectDuration
Determines the power drop duration that is made at the end

of a call when the call is disconnected by the remote party.
Service Activation
FxsServiceActivation
Selects the method used by the user to activate

supplementary services like call hold, second call, call
waiting, call transfer and conference call.
FXS Country Configuration Section

Field Name
SNMP Variable
Description
Override Country
Customization
fxsCountryCustomizationOverride
Allows overriding FXS-related default country settings.
Country Override Loop

Current
fxsCountryCustomizationLoopCurrent
Loop current generated by the FXS port in ma.
Country Override Flash

Hook Detection Range
fxsCountryCustomizationFlashHookDetectionR
ange
The range in which the hook switch must remain pressed to

perform a flash hook.
FXS Bypass Section

Field Name
SNMP Variable
Description
Endpoint
fxsBypassId
String that identifies a line in other tables.
Activation
fxsBypassActivation
Specifies when the bypass needs to be activated.
Activation DTMF Map
fxsBypassActivationDtmfMap
Specifies the DTMFs to signal to enable the bypass.
Deactivation Timeout
fxsBypassDeactivationTimeout
Specifies the delay to wait before deactivating the bypass

after an on hook if the bypass is activated on demand.
FXO Config Sub-Page

Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C730 / C733 / C731
695
POTS Page
FXO Dialing Configuration Section

Field Name
SNMP Variable
Description
Pre Dial Delay
fxoPreDialDelay
The delay between the time the line is successfully seized,

or dial tone detected, and the moment the destination phone
number is dialed.
Dial Tone Detection

Mode
fxoDialToneDetectionMode
When dial tone detection is enabled, the unit waits for a dial
tone on the FXO line before initiating the dialling sequence.
Dial Tone Detection

Timeout
fxoDialToneDetectionTimeout
Indicates how long the unit waits for a dial tone before
considering the line is busy with an incoming FXO call.
FXO Answering Configuration Section

Field Name
SNMP Variable
Description
ID
fxoAnsweringDelayId
Wait Before Answering

Delay
fxoAnsweringDelayWaitBeforeAnsweringDelay
The waiting period before answering an incoming FXO call.
Answering on Caller Id
Detection
fxoAnsweringDelayAnsweringOnCallerIdDetecti
on
Enables answering upon caller ID detection instead of the

waiting delay configured by WaitBeforeAnsweringDelay.
Wait for Callee to

Answer
fxoAnsweringDelayWaitForCalleeToAnswer
When the endpoint is set up for automatic call, enabling this

variable makes the endpoint wait until the called the party
connection is established before answering the incoming
call.
FXO Incoming Call Behavior Section

Field Name
SNMP Variable
Description
ID
fxoIncomingCallBehaviorId
Not Allowed Behavior
fxoIncomingCallBehaviorNotAllowedBehavior
Under certain circumstances (locked port, configuration,

etc.), incoming FXO calls are not allowed.
FXO Line Verification Section

Field Name
SNMP Variable
Description
Link State Verification
fxoLinkStateVerification
The link state verification mechanism allows the detection of

defective or down lines based on the absence of current
when closing the loop.
Link State Verification

Timeout
fxoLinkStateVerificationTimeout
Indicates how long the unit waits to successfully take the

line before considering the line is defective or down.
FXO Force End of Call Section

Field Name
696
SNMP Variable
Description
Force End of Call on

Call Failure
fxoFeocOnCallFailureEnable
Enables forced-end-of-call on call failure.
Call Failure Timeout
FxoFeocOnCallFailureTimeout
The waiting period before terminating a call in an error state.

Silence Detection
Mode
FxoFeocOnSilenceDetectionMode
Enables forced-end-of-call on silence detection.
Silence Detection
Timeout
FxoFeocOnSilenceDetectionTimeout
Maximum amount of time that a call can remain silent before

it is terminated and the line is released.

Tone Detection Mode
FxoFeocOnToneDetectionMode
Enables forced-end-of-call upon tone detection.
Tone Detection
Custom Frequency
FxoFeocToneCustomFrequency
Frequency to detect in the custom cadence.
SIP Page
Field Name
SNMP Variable
Description
Tone Detection
Custom Cadence
FxoFeocToneCustomCadence
A cadence is a series of frequencies that are played for a

specified time, making up a tone.
Detection Custom
Repetition
FxoFeocToneCustomRepetition
Repetition of the cadence.
SIP Page
Gateways Sub-Page
SIP Gateway Status Section
Field Name
SNMP Variable
Description
Name
gatewayStatusName
Name of the SIP gateway.
Network Interface
gatewayStatusNetworkInterface
Network on which the gateway listens for incoming SIP

traffic.
Port
gatewayStatusPort
Port on which the gateway listens for incoming unsecure

SIP traffic.
Secure Port
gatewayStatusSecurePort
Port on which the gateway listens for incoming secure SIP

traffic.
State
gatewayStatusState
Current state of the gateway.
SIP Gateway Configuration Section

Field Name
SNMP Variable
Description
Name
gatewayName
Name of the SIP gateway. It identifies the gateway in other

tables.
Type
gatewayType
Type of gateway: Trunk or Endpoint
Network Interface
gatewayNetworkInterface
Network on which the gateway listens for incoming SIP

traffic.
Port
gatewayPort
Port on which the gateway listens for incoming unsecure

SIP traffic.
Secure Port
gatewayStatusSecurePort
Port on which the gateway listens for incoming secure SIP

traffic.
InsertGateway (command)
Adds a row.
gatewayDelete
Deletes this row.
Servers Sub-Page
Field Name
SNMP Variable
RegistrationRefresh (command)
Description
Command to refresh the registrations.
TLS Persistent Connections Status Section

Field Name
SNMP Variable
Description
Gateway
tlsPersistentConnectionStatusGateway
Local Port
tlsPersistentConnectionStatusLocalPort
Local port used by the TLS persistent connection.
697
Field Name
SIP Page
SNMP Variable
Description
Configured Remote
Host
tlsPersistentConnectionStatusRemoteHost
The remote host used to establish the TLS persistent

connection.
Remote IP Address
tlsPersistentConnectionStatusRemoteAddress
The resolved IP address of the remote host used to

establish the TLS persistent connection.
State
tlsPersistentConnectionStatusState
The current state of the TLS persistent connection.
SIP Default Servers Section

Field Name
SNMP Variable
Description
Registrar Host
defaultStaticRegistrarServerHost
SIP registrar server FQDN and port.
Proxy Host
defaultStaticProxyHomeDomainHost
SIP proxy server FQDN and port.
Outbound Proxy Host
defaultStaticProxyOutboundHost
SIP outbound proxy server FQDN and port.
Messaging Server
Host
defaultStaticMessagingHost
Messaging server FQDN and port.
SIP Gateway Specific Registrar Servers Section

Field Name
SNMP Variable
Description
Gateway Name
gwSpecificRegistrationGatewayName
String that identifies a SIP gateway in other tables.
Gateway Specific
gwSpecificRegistrationEnableConfig
Defines the configuration to use for a specific SIP gateway.
Registrar Host
gwSpecificRegistrationServerHost
SIP registrar server FQDN and port for a specific SIP

gateway.
SIP Gateway Specific Proxy Servers Section

Field Name
SNMP Variable
Description
Gateway Name
gwSpecificProxyGatewayName
Gateway Specific
gwSpecificProxyEnableConfig
Defines the configuration to use for a specific SIP gateway.
Proxy Host
gwSpecificProxyHomeDomainHost
SIP proxy server FQDN and port for a specific SIP gateway.
Outbound Proxy Host
gwSpecificProxyOutboundHost
SIP outbound proxy server FQDN and port for a specific SIP
gateway.
Keep Alive Section

Field Name
SNMP Variable
Description
Keep Alive Method
sipKeepAliveMethod
Method used to perform the SIP keep alive.
Keep Alive Interval
sipKeepAliveInterval
Defines the interval, in seconds, at which SIP OPTIONS are

sent to verify the server status.
Keep Alive Destination
sipKeepAliveDestination
Determines the behaviour of the device when performing

the keep alive action.
SIP Gateway Specific Keep Alive Targets Section

Field Name
698
SNMP Variable
Description
Gateway Name
gwKeepAliveAlternateDestinationGatewayNam
e
Alternate Target
gwKeepAliveAlternateDestinationAlternateDesti
nation
Alternate destination target server FQDN and port for a

SIP Page
Registrations Sub-Page
Endpoints Registration Status Section
Field Name
SNMP Variable
Description
Endpoint
registrationStatusEndpoint
The endpoint related to this registration.
User Name
registrationStatusUsername
The username currently used by the registration.
Gateway Name
registrationStatusGateway
Registrar
registrationStatusRegistrar
The host of the registrar currently used by the registration.
Status
registrationStatusState
The current state of the registration.
Endpoints Messaging Subscription Status Section

Field Name
SNMP Variable
Description
Endpoint
mwiStatusEndpoint
The endpoint related to this subscription.
User Name
mwiStatusUsername
The username currently used by the subscription.
Gateway Name
mwiStatusGatewayName
The SIP gateway used for this subscription.
Messaging Host
mwiStatusMessagingHost
Messaging server FQDN and port used to subscribe the

event state.
MWI Status
mwiStatusSubscriptionState
The current state of the subscription.
Unit Registration Status Section

Field Name
SNMP Variable
Description
User Name
registrationStatusUsername
The username currently used by the registration.
Gateway Name
registrationStatusGateway
Registrar
registrationStatusRegistrar
The host of the registrar currently used by the registration.
Status
registrationStatusState
The current state of the registration.
Endpoints Registration Section

Field Name
SNMP Variable
Description
Endpoint
userAgentEpId
User Name
userAgentUserName
String that uniquely identifies this endpoint in the domain.
Friendly Name
userAgentFriendlyName
Friendly name for SIP User Agent.
Register
userAgentRegister
Indicate whether the endpoint needs to register to the

registrar.
Gateway Name
userAgentGatewayName
Selects on which SIP gateway the user configuration is

applied.
Unit Registration Section

Field Name
SNMP Variable
Description
Index
UnitRegistrationsIndex
Unique identifier of the row.
User Name
UnitRegistrationsUsername
String that uniquely identifies this unit in the domain.
Gateway Name
UnitRegistrationsGatewayName
Selects on which SIP gateway the unit configuration is

applied.
699
Field Name
SIP Page
SNMP Variable
Description
InsertUnitRegistration (command)
Adds a row.
UnitRegistrationsDelete
Delete this row.
Registration Configuration Section

Field Name
SNMP Variable
Description
Default Registration
Refresh Time
defaultRegistrationRefreshTime
Defines the time, relative to the end of the registration, at

which a registered unit will begin updating its registration.
Proposed Expiration
Value In Registration
defaultRegistrationProposedExpirationValue
Configures the suggested expiration delay of a contact in

the SIP REGISTER.
Default Expiration
Value In Registration
defaultRegistrationExpirationValue
Configures the default registration expiration.
Authentication Sub-Page
Field Name
700
SNMP Variable
Description
Index
authenticationIndex
Authentication index for this row.
Criteria Selection
authenticationCriteriaSelection
Selects which criterion to use for matching an authentication

request with an authentication entry.
Unit: The authentication entry is used on all challenges that
were not matched on a previous entry.
Endpoint: The authentication entry is used for all
challenges related to the endpoint specified in
'Authentication.EpId'.
Gateway: The authentication entry is used for all
challenges related to the gateway specified in
'Authentication.GatewayName'.
Username: The authentication entry is used for all
challenges related to the username specified in
'Authentication.UsernameCriteria'..
Endpoint
authenticationEpId
String that identifies an endpoint in the UserAgent. Used

only if the variable CriteriaSelection is set to EndPoint.
Gateway
authenticationGatewayName
String that identifies a SIP gateway in the GatewayStatus

table. It is only used if the variable CriteriaSelection is set to
Gateway.
Username Criteria
AuthenticationUsernameCriteria
String that identifies a username in the SIP request to

authenticate. It is used only if the variable 'CriteriaSelection'
is set to 'Username'
Validate Realm
authenticationValidateRealm
Defines whether or not the current credentials are valid for

any realm.
Realm
authenticationRealm
Authentication Realm.
User Name
authenticationUserName
String that uniquely identifies this entity in the realm.
Password
authenticationPassword
User password.
authenticationInsert
authenticationDelete
Deletes this row.
authenticationDown
authenticationUp
SIP Page
Transport Sub-Page
Field Name
SNMP Variable
Description
Add SIP Transport in

Registration
transportConfigRegistrationEnable
Indicates whether or not the SIP Gateway must include its

supported transports in its registrations.
Add SIP Transport in

Contact Header
transportConfigContactEnable
Indicates whether or not the SIP Gateway must include its

supported transport in all SIP messages that have the
contact header, except for the REGISTER message.
Persistent Base Port
transportPersistentBasePort
Base port used to establish persistent connections with SIP

servers.
Persistent Retry
Interval
transportFailbackInterval
Time interval before retrying the establishment of a

persistent connection.
TLS Trusted Certificate

Level
transportTlsCertificateTrustLevel
Defines how a peer certificate is considered trusted for a

TLS connection.
TCP Connect Timeout
interopTcpConnectTimeout
Defines the maximum time, in seconds, the unit should try to

establish a TCP or TLS connection to SIP hosts.
Protocol Configuration Section

Field Name
SNMP Variable
Description
UDP
transportConfigUdpEnable
Enables or disables the UDP transport.
UDP QValue
transportConfigUdpQValue
Indicates the priority of the UDP transport.
TCP
transportConfigTcpEnable
Enables or disables the TCP transport.
TCP QValue
transportConfigTcpQValue
Indicates the priority of the TCP transport.
TLS
transportConfigTlsEnable
Enables or disables the TLS transport.
TLS QValue
transportConfigTlsQValue
Indicates the priority of the TLS transport.
Interop Sub-Page
Behavior on T.38 INVITE Not Accepted Section
Field Name
SNMP Variable
Description
SIP Error Code
behaviorOnT38InviteNotAcceptedSipErrorCode
SIP code in the error response to an INVITE for T.38 fax.
Behavior
behaviorOnT38InviteNotAcceptedBehavior
Behavior of the device when receiving a SIP error response

to an INVITE for T.38 fax.
SIP Interop Section

Field Name
SNMP Variable
Description
Secure Header
interopSiemensTransportHeaderEnable
Add the 'x-Siemens-Call-Type' header to the SIP packets.
Default Username
Value
interopDefaultUsernameValue
Username to use when the username is empty or undefined.
OPTIONS Method
Support
interopSipOptionsMethodSupport
Determines the behaviour of the device when answering a

SIP OPTIONS request.
Ignore OPTONS on no
usable endpoints
InteropIgnoreSipOptionsOnNoUsableEndpoints
Determines whether or not the SIP OPTIONS requests

should be ignored when all endpoints are unusable.
Behavior On Machine
Detection
InteropBehaviorOnMachineDetection
Specifies the SIP device behavior when a machine is

detected during a call.
701
Field Name
SIP Page
SNMP Variable
Description
Registration Contact
Matching
InteropRegistrationContactMatching
Specifies the matching behaviour for the contact header

received in positive responses to REGISTER requests sent
by the unit.
Transmission Timeout
interopTransmissionTimeout
Changes the time to wait for a response or an ACK before

considering a transaction timed out.
SDP Interop Section

Field Name
SNMP Variable
Description
Offer Answer Model

Answer Codec
Negotiation
answerCodecNegotiation
Defines the codec negotiation rule when generating a SDP

answer.
Enforce Offer Answer

Model
interopEnforceOfferAnswerModel
Determines whether or not the unit requires strict adherence

to RFC 3264 from the peer when negotiating capabilities for
the establishment of a media session.
Allow Less Media in

Response
interopAllowLessMediaInResponse
Selects whether or not the unit enables the mapping

between the "+" prefix of the username and the "type of
number" property.
Allow Media
Reactivation in Answer
interopAllowMediaReactivationInAnswer
Determines the unit behaviour when receiving a SDP

answer activating a media that had been previously
deactivated in the offer.
Multiple Active Media

Allow Audio and Image
Negotiation
interopAllowAudioAndImageNegotiation
Determines the unit behaviour when offering media or

answering to a media offer with audio and image
negotiation.
Allow Multiple Active

Media in Answer
interopAllowMultipleActiveMediaInAnswer
Determines the behaviour of the device when answering a

request offering more than one active media.
Other
On Hold SDP Stream
Direction in Answer
interopOnHoldSdpStreamDirection
Define how to set the direction attribute and the connection

address in the SDP when answering a hold offer with the
direction attribute sendonly.
Codec Vs Bearer
Capabilities Mapping
Preferred Codec
Choice
interopCodecVsBearerCapabilitiesMappingPref
erredCodecChoice
Configures the behavior of the

CodecVsBearerCapabilitiesMapping table by modifying the
selection of the preferred codec in the incoming SDP.
TLS Interop Section

Field Name
SNMP Variable
InteropTlsCertificateValidation
Description
Specifies which level of security is used to validate the peer
certificate.
Misc Interop Section

Field Name
702
SNMP Variable
Description
Map Plus to TON

International
interopMapPlusToTonInternational
Defines the behaviour of the unit when receiving less media

announcements in the response than in the offer.
Ignore Plus in
Username
interopIgnorePlusInUsername
Determines whether or not the plus character (+) is ignored

when attempting to match a challenge username with
usernames in the Authentication table.
Escape Pound (#) in

SIP URI Username
interopEscapePoundInSipUriUsername
Determines whether or not the pound character (#) must be

escaped in the username part of a SIP URI.
Escape Format
interopEscapeFormat
Configures the escaped characters to lower or uppercase

hexadecimal format in SIP headers.
SIP Page
Misc Sub-Page
Penalty Box Section
Field Name
SNMP Variable
Description
Penalty Box Activation
penaltyBoxEnable
Indicates whether the unit uses the penalty box feature.
Penalty Box Time
penaltyBoxTime
Amount of time that a host spends in the penalty box.
SIP to Cause Error Mapping Section

Field Name
SNMP Variable
Description
SIP Code
errorMappingSipToCauseSipCode
SIP code to map to a cause.
Cause
errorMappingSipToCauseCause
Cause to map to the SIP code.
ErrorMappingInsertSipToCause (command)
erromappingSipToCauseDelete
Deletes this row.
Configure New SIP To Cause Error Mapping Panel

Field Name
SNMP Variable
Description
SIP Code
errorMappingSipToCauseSipCode
Cause
errorMappingSipToCauseCause
Cause to SIP Error Mapping

Field Name
SNMP Variable
Description
Cause
errorMappingCauseToSipCause
SIP Code
errorMappingCauseToSipSipCode
ErrorMappingInsertCauseToSip (command)
erromappingCauseToSipDelete
Deletes this row.
Cause To SIP Error Mapping Panel

Field Name
SNMP Variable
Description
Cause
errorMappingCauseToSipCause
SIP Code
errorMappingCauseToSipSipCode
Additional Headers Section

Field Name
SNMP Variable
Description
Reason Header
Support
ReasonHeaderSupport
Indicates whether or not the unit uses the SIP reason

header.
Referred-By Support
ReferredByHeader
Indicates how the Referred-By header is used when

participating in a transfer.
703
SIP Page
PRACK Section
Field Name
SNMP Variable
Description
UAS PRACK Support
uasPrackSupport
Determines the support of RFC 3262 (PRACK) when acting

as as user agent server.
UAC PRACK Support
uacPrackSupport
Determines the support of RFC 3262 (PRACK) when acting

as as user agent client.
Session Refresh Section

Field Name
SNMP Variable
Description
Session Refresh Timer

Enable
defaultSessionTimerEnable
Enables/Disables the session expiration services.
Minimum Expiration
Delay (s)
defaultSessionTimerMinimumExpirationDelay
Minimum value for the periodical session refreshes.
Maximum Expiration
Delay (s):
defaultSessionTimerMaximumExpirationDelay
Suggested maximum time for the periodical session

refreshes.
Session Refresh
Request Method
sessionRefreshRequestMethod
Selects the method used for sending Session Refresh

Requests.
SIP Gateway Configuration Section

Field Name
SNMP Variable
Description
Gateway Name
gatewayName
Name of the SIP gateway. It identifies the gateway in other

tables.
SIP Domain Override
gatewayDomain
Controls whether or not to override the SIP domain used.
SIP Transfer Section

Field Name
Blind Transfer Method
SNMP Variable
BlindTransferMethod
Description
Selects the SIP method to use in a blind transfer scenario.
Diversion Section
Field Name
Method
SNMP Variable
diversionConfigMethod
Description
Selects the SIP method used to receive/send call diversion
information in an INVITE.
Event Handling Section

Field Name
704
SNMP Variable
Description
Gateway Name
gwEventHandlingGatewayName
Reboot
gwEventHandlingReboot
Specifies whether a remote reboot via a SIP NOTIFY

message event is supported or not for a specific SIP
gateway.
CheckSync
gwEventHandlingCheckSync
Specifies whether a transfer script via a SIP NOTIFY

message event is supported or not for a specific SIP
gateway.
Media Page
Messaging Subscription Section

Field Name
Username in RequestURI
SNMP Variable
defaultUsernameInRequestUriEnable
gwSpecificMwiUsernameInRequestUriEnable
Description
Indicates whether or not the unit adds the username in the
request URI of MWI SUBSCRIBE requests.
Advice Of Charge (AOC) Section

Field Name
SNMP Variable
Description
Gateway Name
GatewayName
AOC-D Support
AocDSupport
Specifies whether AOC (D)uring a call is supported for a

AOC-E Support
AocESupport
Specifies whether AOC at the (E)nd of a call is supported for

a specific SIP gateway.
Media Page
CODECS Sub-Page
Field Name
Select Endpoint
SNMP Variable
endpointEpId
Description
CODEC Section
Field Name
SNMP Variable
Description
epSpecificCodecG711AlawEnableConfig
epSpecificCodecG711MulawEnableConfig
epSpecificCodecG723EnableConfig
Endpoint Specific
epSpecificCodecG726r16kbpsEnableConfig
Configuration to use for a specific endpoint.
epSpecificCodecT38EnableConfig
epSpecificCodecClearModeEnableConfig
epSpecificCodecClearChannelEnableConfig
epSpecificCodecXCCDEnableConfig
705
Field Name
Media Page
SNMP Variable
Description
defaultCodecG711AlawVoiceEnable
epSpecificCodecG711AlawVoiceEnable
defaultCodecG711MulawVoiceEnable
epSpecificCodecG711MulawVoiceEnable
defaultCodecG723VoiceEnable
epSpecificCodecG723VoiceEnable
defaultCodecG726r16kbpsVoiceEnable
epSpecificCodecG726r16kbpsVoiceEnable
Voice
Indicates whether the codec can be selected for voice

transmission.
defaultCodecClearModeVoiceEnable
epSpecificCodecClearModeVoiceEnable
defaultCodecClearChannelVoiceEnable
epSpecificCodecClearChannelVoiceEnable
defaultCodecXCCDVoiceEnable
epSpecificCodecXCCDVoiceEnable
defaultCodecG711AlawDataEnable
epSpecificCodecG711AlawDataEnable
defaultCodecG711MulawDataEnable
epSpecificCodecG711MulawDataEnable
defaultCodecG726r32kbpsDataEnable
epSpecificCodecG726r32kbpsDataEnable
Data
defaultCodecT38DataEnable
epSpecificCodecT38DataEnable
Indicates whether the codec can be selected for data

transmission.
defaultCodecClearModeDataEnable
epSpecificCodecClearModeDataEnable
defaultCodecClearChannelDataEnable
epSpecificCodecClearChannelDataEnable
defaultCodecXCCDDataEnable
epSpecificCodecXCCDDataEnable
CODEC vs. Bearer Capabilities Mapping Section

Field Name
706
SNMP Variable
Description
Index
defaultCodecVsBearerCapabilitiesMappingInde
x
Index of the current Codec vs. Bearer match.
Enable
defaultCodecVsBearerCapabilitiesMappingEna
bleMap
Defines if the outgoing codecs priority or selection should

reflect the incoming ITC and vice versa.
CODEC
defaultCodecVsBearerCapabilitiesMappingCod
ec
The codec to be prioritized or selected in an outgoing

INVITE when the incoming SETUP's ITC matches
defaultCodecVsBearerCapabilitiesMappingInformationTran
sferCap.
Mapping Type
defaultCodecVsBearerCapabilitiesMappingMap
pingType
The ITC value to be set in the outgoing SETUP when the

incoming INVITE's priority codec matches
defaultCodecVsBearerCapabilitiesMappingCodec.
Media Page
Field Name
ITC
SNMP Variable
defaultCodecVsBearerCapabilitiesMappingInfor
mationTransferCap
Description
Mapping Type

Field Name
SNMP Variable
Description
Endpoint Specific
epSpecificCodecEnableConfig
Enable (G.711 and

G.726)
defaultCodecGenericVoiceActivityDetection
epSpecificCodecGenericVoiceActivityDetection
Generic VAD configuration.
G.711 a-law Section

Field Name
SNMP Variable
Description
Use Endpoint Specific
epSpecificCodecG711AlawEnableConfig
Voice Transmission
defaultCodecG711AlawVoiceEnable
epSpecificCodecG711AlawVoiceEnable

transmission.
Voice Priority
defaultCodecG711AlawVoicePriority
epSpecificCodecG711AlawVoicePriority
Priority of this voice codec versus the other voice codecs.
Data Transmission
defaultCodecG711AlawDataEnable
epSpecificCodecG711AlawDataEnable

transmission.
Data Priority
defaultCodecG711AlawDataPriority
epSpecificCodecG711AlawDataPriority
Priority of this data codec versus the other data codecs.
Minimum Packetization
Time
defaultCodecG711AlawMinPTime
epSpecificCodecG711AlawMinPTime
Lower boundary for the packetization period.
Maximum
Packetization Time
defaultCodecG711AlawMaxPTime
epSpecificCodecG711AlawMaxPTime
Upper boundary for the packetization period.
G.711 u-law Section

Field Name
SNMP Variable
Description
epSpecificCodecG711MulawEnableConfig
Voice Transmission
defaultCodecG711MulawVoiceEnable
epSpecificCodecG711MulawVoiceEnable

transmission.
Voice Priority
defaultCodecG711MulawVoicePriority
epSpecificCodecG711MulawVoicePriority
Data Transmission
defaultCodecG711MulawDataEnable
epSpecificCodecG711MulawDataEnable

transmission.
Data Priority
defaultCodecG711MulawDataPriority
epSpecificCodecG711MulawDataPriority
Time
defaultCodecG711MulawMinPTime
epSpecificCodecG711MulawMinPTime
Maximum
Packetization Time
defaultCodecG711MulawMaxPTime
epSpecificCodecG711MulawMaxPTime
G.723 Section
Field Name
SNMP Variable
Description
Voice Transmission

transmission.
707
Field Name
Media Page
SNMP Variable
Description
Voice Priority
defaultCodecG723VoicePriority
epSpecificCodecG723VoicePriority
Bit Rate
defaultCodecG723Bitrate
epSpecificCodecG723Bitrate
G.723.1 bit rate to use.
Time
defaultCodecG723MinPTime
epSpecificCodecG723MinPTime
Maximum
Packetization Time
defaultCodecG723MaxPTime
epSpecificCodecG723MaxPTime
G.726 16Kbps Section

Field Name
SNMP Variable
Description
Voice Transmission

transmission.
Voice Priority
defaultCodecG726r16kbpsVoicePriority
epSpecificCodecG726r16kbpsVoicePriority
Payload Type
defaultCodecG726r16kbpsPayloadType
epSpecificCodecG726r16kbpsPayloadType
RTP dynamic payload type used in an initial offer.
Time
defaultCodecG726r16kbpsMinPTime
epSpecificCodecG726r16kbpsMinPTime
Maximum
Packetization Time
defaultCodecG726r16kbpsMaxPTime
epSpecificCodecG726r16kbpsMaxPTime

Field Name
SNMP Variable
Description
Voice Transmission

transmission.
Voice Priority
Payload Type
Time
Maximum
Packetization Time

Field Name
708
SNMP Variable
Description
Voice Transmission

transmission.
Voice Priority
Data Transmission

transmission.
Data Priority
defaultCodecG726r32kbpsDataPriority
epSpecificCodecG726r32kbpsDataPriority
Media Page
Field Name
SNMP Variable
Description
Payload Type
Time
Maximum
Packetization Time

Field Name
SNMP Variable
Description
Voice Transmission

transmission.
Voice Priority
Data Transmission

transmission.
Data Priority
defaultCodecG726r40kbpsDataPriority
epSpecificCodecG726r40kbpsDataPriority
Payload Type
Time
Maximum
Packetization Time
G.729 Section
Field Name
SNMP Variable
Description
Voice Transmission

transmission.
Voice Priority
defaultCodecG729VoicePriority
epSpecificCodecG729VoicePriority
Time
defaultCodecG729MinPTime
epSpecificCodecG729MinPTime
Maximum
Packetization Time
defaultCodecG729MaxPTime
epSpecificCodecG729MaxPTime
Built-In VAD
defaultCodecG729VoiceActivityDetection
epSpecificCodecG729VoiceActivityDetection
G.729 VAD configuration.
T.38 Section
Field Name
SNMP Variable
Description
epSpecificCodecT38EnableConfig
Enable
defaultCodecT38DataEnable
epSpecificCodecT38DataEnable
If enabled, the T.38 protocol is used for fax transmission.
Priority
defaultCodecT38DataPriority
epSpecificCodecT38DataPriority
Redundancy Level
defaultCodecT38RedundancyLevel
epSpecificCodecT38RedundancyLevel
Number of redundancy packets.
709
Field Name
Media Page
SNMP Variable
Description
Detection Threshold
defaultCodecT38DetectionThreshold
epSpecificCodecT38DetectionThreshold
Sets the T.38 input signal detection threshold.
Frame Redundancy
Level
defaultCodecT38FinalFramesRedundancy
Defines the number of times T.38 packets will be

retransmitted.
No Signal
defaultCodecT38NoSignalEnable
Enables/disables the sending of T.38 no-signal packets.
No Signal Timeout
defaultCodecT38NoSignalTimeout
The period, in seconds, at which no-signal packets are sent

during a T.38 transmission, in the absence of valid data.
Clear Mode Section

Field Name
SNMP Variable
Description
epSpecificCodecClearModeEnableConfig
Voice Transmission
defaultCodecClearModeVoiceEnable
epSpecificCodecClearModeVoiceEnable

transmission.
Voice Priority
defaultCodecClearModeVoicePriority
epSpecificCodecClearModeVoicePriority
Data Transmission
defaultCodecClearModeDataEnable
epSpecificCodecClearModeDataEnable

transmission.
Data Priority
defaultCodecClearModeDataPriority
epSpecificCodecClearModeDataPriority
Payload Type
defaultCodecClearModePayloadType
epSpecificCodecClearModePayloadType
Time
defaultCodecClearModeMinPTime
epSpecificCodecClearModeMinPTime
Maximum
Packetization Time
defaultCodecClearModeMaxPTime
epSpecificCodecClearModeMaxPTime
Clear Channel Section

Field Name
SNMP Variable
Description
epSpecificCodecClearChannelEnableConfig
Voice Transmission
defaultCodecClearChannelVoiceEnable
epSpecificCodecClearChannelVoiceEnable

transmission.
Voice Priority
defaultCodecClearChannelVoicePriority
epSpecificCodecClearChannelVoicePriority
Data Transmission
defaultCodecClearChannelDataEnable
epSpecificCodecClearChannelDataEnable

transmission.
Data Priority
defaultCodecClearChannelDataPriority
epSpecificCodecClearChannelDataPriority
Payload Type
defaultCodecClearChannelPayloadType
epSpecificCodecClearChannelPayloadType
Time
defaultCodecClearChannelMinPTime
epSpecificCodecClearChannelMinPTime
Maximum
Packetization Time
defaultCodecClearChannelMaxPTime
epSpecificCodecClearChannelMaxPTime
X CCD Section
Field Name
Endpoint Specific
710
SNMP Variable
epSpecificCodecXCCDEnableConfig
Description
Media Page
Field Name
SNMP Variable
Description
Voice Transmission
defaultCodecXCCDVoiceEnable
epSpecificCodecXCCDVoiceEnable

transmission.
Voice Priority
defaultCodecXCCDVoicePriority
epSpecificCodecXCCDVoicePriority
Data Transmission
defaultCodecXCCDDataEnable
epSpecificCodecXCCDDataEnable

transmission.
Data Priority
defaultCodecXCCDDataPriority
epSpecificCodecXCCDDataPriority
Payload Type
defaultCodecXCCDPayloadType
epSpecificCodecXCCDPayloadType
Time
defaultCodecXCCDMinPTime
epSpecificCodecXCCDMinPTime
Maximum
Packetization Time
defaultCodecXCCDMaxPTime
epSpecificCodecXccdaxPTime
Security Sub-Page
Security Section
Field Name
SNMP Variable
Description
RTP
Mode
defaultSecurityRtpMode
epSpecificSecurityRtpMode
Defines the RTP payload mode (secure or not secure).
Key Management
defaultSecurityKeyManagement
epSpecificSecurityKeyManagement
Defines the key management protocol for SRTP.
Encryption
defaultSecurityRtpEncryption
epSpecificSecurityRtpEncryption
Defines the encryption type to be used with SRTP.

T.38
Allow unsecure T.38

with secure RTP
allowUnsecureT38WithSrtp
Enables T38 even if the call has been established previously

in SRTP.
RTP Stats
Field Name
SNMP Variable
Description
Period Beginning
lastPeriodsStatsPeriodBeginning
Date and time of the collection period beginning.
Period End
lastPeriodsStatsPeriodEnd
Date and time of the collection period end.
Octets Tx
lastPeriodsStatsOctetsTransmitted
Number of octets transmitted during the collection period.
Octets Rx
lastPeriodsStatsOctetsReceived
Number of octets received during the collection period.
Packets Tx
lastPeriodsStatsPacketsTransmitted
Number of packets transmitted during the collection period.
Packets Rx
lastPeriodsStatsPacketsReceived
Number of packets received during the collection period.
Packets Lost
lastPeriodsStatsPacketsLost
Number of packets lost during the collection period.
Min. Jitter
lastPeriodsStatsMinimumInterarrivalJitter
Minimum interarrival time, in milliseconds, during the

collection period.
Max. Jitter
lastPeriodsStatsMaximumInterarrivalJitter
Maximum interarrival time, in milliseconds, during the

collection period.
Avg. Jitter
lastPeriodsStatsAverageInterarrivalJitter
Average interarrival time, in milliseconds, during the

collection period.
711
Field Name
Media Page
SNMP Variable
Description
Min. Latency
lastPeriodsStatsMinimumLatency
Minimum latency, in milliseconds, during the collection

period.
Max. Latency
lastPeriodsStatsMaximumLatency
Maximum latency, in milliseconds, during the collection

period.
Avg. Latency
lastPeriodsStatsAverageLatency
Average latency, in milliseconds, during the collection

period.
Field Name
SNMP Variable
Description
Octets Tx
lastConnectionsStatsOctetsTransmitted
Number of octets transmitted during the connection.
Octets Rx
lastConnectionsStatsOctetsReceived
Number of octets received during the connection.
Packets Tx
lastConnectionsStatsPacketsTransmitted
Number of packets transmitted during the connection.
Packets Rx
lastConnectionsStatsPacketsReceived
Number of packets received during the connection.
Packets Lost
lastConnectionsStatsPacketsLost
Number of packets lost during the connection.
Min. Jitter
lastConnectionsStatsMinimumInterarrivalJitter
Minimum interarrival time, in milliseconds, during the

connection.
Max. Jitter
lastConnectionsStatsMaximumInterarrivalJitter
Maximum interarrival time, in milliseconds, during the

connection.
Avg. Jitter
lastConnectionsStatsAverageInterarrivalJitter
Average interarrival time, in milliseconds, during the

connection.
Min. Latency
lastConnectionsStatsMinimumLatency
Minimum latency, in milliseconds, during the connection.
Max. Latency
lastConnectionsStatsMaximumLatency
Maximum latency, in milliseconds, during the connection.
Avg. Latency
lastConnectionsStatsAverageLatency
Average latency, in milliseconds, during the connection.
Statistics Configuration Section

Field Name
SNMP Variable
Description
Collection Period
(minutes)
statsCollectionPeriodDuration
Specifies the collection period duration (in minutes).
Generate Connection
End Notification
statsPerConnectionNotificationEnable
Enables the generation of connection end statistics

notification.
Generate Collection
Period End Notification
statsPerPeriodNotificationEnable
Enables the generation of period statistics notification.
Misc Sub-Page
Jitter Buffer Section
Field Name
712
SNMP Variable
Description
Endpoint Specific
epSpecificJitterBufferEnableConfig
Level
defaultJitterBufferLevel
epSpecificJitterBufferLevel
Jitter bufffer level.
Voice Call
Minimum
defaultJitterBufferCustomMinLength
epSpecificJitterBufferCustomMinLength
Jitter buffer minimum length.
Voice Call
Maximum
defaultJitterBufferCustomMaxLength
epSpecificJitterBufferCustomMaxLength
Jitter buffer maximum length.
Data Call
Playout Type
defaultVbdJitterBufferType
epSpecificCustomVbdJitterBufferType
Algorithm to use for managing the jitter buffer during a call.
Telephony Page
Field Name
SNMP Variable
Description
Data Call
Minimum
defaultVbdJitterBufferCustomMinLength
epSpecificCustomVbdMinLength
The delay the jitter buffer tries to maintain.
Data Call
Nominal
defaultVbdJitterBufferCustomNomLength
epSpecificCustomVbdNomLength
The delay the jitter buffer uses when a call begins.
Data Call
Maximum
defaultVbdJitterBufferCustomMaxLength
epSpecificCustomVbdMaxLength
The highest delay the jitter buffer is allowed to introduce.
DTMF Transport Section

Field Name
SNMP Variable
Description
Endpoint Specific
epSpecificDtmfTransportEnableConfig
Transport Method
defaultDtmfTransportMethod
epSpecificDtmfTransportMethod
Type of DTMF transport.
SIP Transport Method
interopDtmfTransportMethod
Defines the method used to transport DTMFs out-of-band

over the SIP protocol
Payload Type
defaultDtmfTransportPayloadType
epSpecificDtmfTransportPayloadType
RTP dynamic payload type used for telephone-event in an

initial offer.
Machine Detection Section

Field Name
SNMP Variable
Description
Endpoint Specific
specificMachineDetectionEnableConfig
Configuration to use for a specific interface.
CNG Tone Detection
defaultMachineDetectionCngToneDetection
specificMachineDetectionCngToneDetection
Enables fax calling tone (CNG tone) detection.
CED Tone Detection
defaultMachineDetectionCedToneDetection
specificMachineDetectionCedToneDetection
Enables CED tone detection.
V.21 Modulation
Detection
defaultMachineDetectionV21ModulationDetecti
on
specificMachineDetectionV21ModulationDetecti
on
Enables fax V.21 modulation detection.
Behavior on CED Tone

Detection
defaultMachineDetectionBehaviorOnCedToneD
etection
specificMachineDetectionBehaviorOnCedTone
Detection
Defines the behavior of the unit upon detection of a CED

tone.
Base Ports Section

Field Name
SNMP Variable
Description
RTP
ipTransportRtpBasePort
UDP base port for the RTP/RTCP protocols.
SRTP
ipTransportSrtpBasePort
UDP base port for the SRTP/SRTCP protocols.
T.38
ipTransportT38BasePort
T.38 base port.
Telephony Page
DTMF Maps Sub-Page
713
Telephony Page

Field Name
First DTMF Timeout
SNMP Variable
dtmfMapTimeoutFirstDtmf
Description
Time the user has to enter the first DTMF after the dial tone.
Inter DTMF Timeout
dtmfMapTimeoutInterDtmf
Value of the T DTMF in the DTMF map strings.
Completion Timeout
dtmfMapTimeoutCompletion
Total time the user has to dial the DTMF sequence.
DTMF Maps Digit

Detection (FXO/FXS)
DtmfMapDigitDetection
Determines when a digit is processed through the DTMF

maps.
Allowed DTMF Map Section

Field Name
SNMP Variable
Description
Index
callDtmfMapAllowedIndex
Accepted DTMF map index for this row.
Enable
callDtmfMapAllowedEnable
Enables/Disables the row.
Apply to
callDtmfMapAllowedApplyTo
Entity to which apply the DTMF map.
Endpoint
callDtmfMapAllowedEpId
DTMF Map
callDtmfMapAllowedDtmfMap
DTMF map that is considered valid when dialed.
Transformation
callDtmfMapAllowedDtmfTransformation
Transformation to apply to the signalled DTMF before using

it as call destination.
Target
callDtmfMapAllowedTargetHost
Target to use when the DTMF map matches.
Emergency
callDtmfMapAllowedEmergency
Enables/Disables the emergency process of the call.
Refused DTMF Map Section

Field Name
SNMP Variable
Description
Index
callDtmfMapRefuseIndex
Refused DTMF map index for this row.
Enable
callDtmfMapRefuseEnable
If enabled, this DTMF map is recognised and refused only if

it is also valid.
Apply to
callDtmfMapRefuseApplyTo
Sets the entity to which apply the DTMF map.
Endpoint
callDtmfMapRefuseEpId
DTMF Map
callDtmfMapRefuseDtmfMap
DTMF map that is considered invalid when dialed.
DTMF Map Timeout Section

Field Name
714
SNMP Variable
Description
Endpoint
EpSpecificDtmfMapTimeoutEpId
Override
EpSpecificDtmfMapTimeoutEnableConfig
Defines the configuration to use for a specific endpoint.
First DTMF Timeout
EpSpecificDtmfMapTimeoutFirstDtmf
Time the user has to enter the first DTMF after the dial tone.
Inter DTMF Timeout
EpSpecificDtmfMapTimeoutInterDtmf
Value of the 'T' DTMF in the DTMF map strings.
Completion Timeout
EpSpecificDtmfMapTimeoutCompletion
Total time the user has to dial the DTMF sequence.
Telephony Page
Call Forward Sub-Page

Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
Field Name
Select Endpoint
SNMP Variable
endpointEpId
Description
Call Forward On Busy Section

Field Name
SNMP Variable
Description
Endpoint Specific
epSpecificForwardOnBusyEnableConfig
Allow Activation via

Handset
defaultForwardOnBusyEnable
epSpecificForwardOnBusyEnable
Enables/Disables the call forward on busy service.
DTMF Map Activation
defaultForwardOnBusyDtmfMapActivation
DTMF map the user can dial to enable the application of the
service.
DTMF Map
Deactivation
defaultForwardOnBusyDtmfMapDeactivation
DTMF map the user can dial to disable the application of the
service.
Activation
forwardOnBusyConfigActivation
Activation status of the call forward on busy service.
Forwarding Address:
forwardOnBusyConfigForwardingAddress
Address or telephone number to which the user wants to

forward calls.
Call Forward On No Answer Section

Field Name
SNMP Variable
Description
Endpoint Specific
epSpecificForwardNoAnswerEnableConfig

Handset
defaultForwardNoAnswerEnable
epSpecificForwardNoAnswerEnable
Enables/Disables the call forward on no answer service.
DTMF Map Activation
defaultForwardNoAnswerDtmfMapActivation
service.
DTMF Map
Deactivation
defaultForwardNoAnswerDtmfMapDeactivation
service.
Timeout
defaultForwardNoAnswerTimeout
epSpecificForwardNoAnswerTimeout
Time, in milliseconds, the telephone keeps ringing before

the call forwarding activates.
Activation
forwardNoAnswerConfigActivation
Activation status of the call forward on no answer service.
Forwarding Address:
forwardNoAnswerConfigForwardingAddress

forward calls.
715
Telephony Page
Call Forward Unconditional Section

Field Name
SNMP Variable
Description
Endpoint Specific
epSpecificForwardUnconditionalEnableConfig

Handset
defaultForwardUnconditionalEnable
epSpecificForwardUnconditionalEnable
Enables/Disables the unconditional call forward service.
DTMF Map Activation
defaultForwardUnconditionalDtmfMapActivation
service.
DTMF Map
Deactivation
defaultForwardUnconditionalDtmfMapDeactivati
on
service.
Activation
forwardUnconditionalConfigActivation
Activation state of the unconditional call forward service.
Forwarding Address:
forwardUnconditionalConfigForwardingAddress

forward calls.
Services Sub-Page
Field Name
Select Endpoint
SNMP Variable
endpointEpId
Description
Service Section
Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
Field Name
716
SNMP Variable
Description
Blind Transfer
transferStatusBlindState
Status of the blind transfer service.
Attended Transfer
transferStatusAttendedState
Status of the attended transfer service.
Call Waiting
callWaitingStatusState
Status of the call waiting service.
Conference
conferenceStatusState
Status of the conference service.
Hold
holdStatusState
Status of the holding service.
Second Call
secondCallStatusState
Status of the second call service.
Telephony Page
Active Call Completion Section

Mediatrix 3716
Mediatrix 3731
Mediatrix 3732
Mediatrix 3741
Mediatrix 3742
Mediatrix LP Series
Mediatrix C7 Series
Field Name
SNMP Variable
Description
Endpoint
callCompletionConfigEpId
Identification of the endpoint using this call completion

service.
Type
callCompletionConfigType
The type of the call completion.
Target Address
callCompletionConfigTargetAddress
The target address of the call completion.
Target State
callCompletionConfigTargetState
The state of the call completion target.
Services Configuration Section

Field Name
SNMP Variable
Description
Endpoint Specific
epSpecificCallEnableConfig
Hook Flash Processing
defaultCallHookFlashProcessing
epSpecificCallHookFlashProcessing
Selects how to process hook-flash detection.
Endpoint Specific
epSpecificAutoCallEnableConfig
Automatic Call
Activation
defaultAutoCallEnable
epSpecificAutoCallEnable
Enables/Disables the automatic call service. This service

provides a 'redphone'-like experience.
Automatic Call Target
defaultAutoCallTargetAddress
epSpecificAutoCallTargetAddress
Address or telephone number that the user wants to

automatically call.
Automatic Call
Call Completiona
Endpoint Specific
epSpecificCallCompletionEnableConfig
Allow CCBS Activation

Via Handset
defaultCallCompletionBusySubscriberEnable
epSpecificCallCompletionBusySubscriberEnabl
e
Enables/Disables the call completion busy subscriber

(CCBS) service.
Allow CCNR Activation

Via Handset
defaultCallCompletionNoResponseEnable
epSpecificCallCompletionNoResponseEnable
Enables/Disables the call completion no response (CCNR)

service.
CCBS DTMF Map

Activation
defaultCallCompletionBusySubscriberDtmfMap
Activation
call completion busy subscriber (CCBS) service.
CCNR DTMF Map

Activation
defaultCallCompletionNoResponseDtmfMapAct
ivation
call completion no response (CCNR) service.
DTMF Map
Deactivation
defaultCallCompletionDtmfMapDeactivation
call completion busy subscriber (CCBS) and call completion
no response (CCNR) services.
Expiration Timeout
defaultCallCompletionExpirationTimeout
Defines the delay after the call completion activation to

automatically deactivate the call completion if the call is not
completed.
717
Field Name
Method
Telephony Page
SNMP Variable
Description
defaultCallCompletionMethod
Selects the call completion method to detect that the call

completion destination is ready to complete the call.
Auto Reactivate
defaultCallCompletionAutoReactivateEnable
Enables/Disables the call compleion auto reactivation.
Auto Reactivate Delay
defaultCallCompletionAutoReactivateDelay
Defines the minimal delay to wait before executing a call

completion after its activation.
Early-Media Behaviour
defaultCallCompletionEarlyMediaBehaviour
Defines how the call completion service needs to interpret

the reception of a progress message with early media.
Polling Interval
defaultCallCompletionPollingInterval
Defines the delay between the calls to the call completion

target used for the polling mechanism.
Call Transferb
Endpoint Specific
epSpecificTransferEnableConfig
Blind Transfer
Activation
defaultTransferBlindEnable
epSpecificTransferBlindEnable
Enables/Disables the blind call transfer service.
Attended Transfer
Activation
defaultTransferAttendedEnable
epSpecificTransferAttendedEnable
Enables/Disables the attended call transfer service.

Call Waitingc
Endpoint Specific
epSpecificCallWaitingEnableConfig
Call Waiting Activation
defaultCallWaitingEnable
epSpecificCallWaitingEnable
Enables/Disables the call waiting service.
Cancel DTMF Map
defaultCallWaitingCancelDtmfMap
Default DTMF Map to Cancel the Call Waiting Service

Conferenced
Endpoint Specific
epSpecificConferenceEnableConfig
Conference Activation
defaultConferenceEnable
epSpecificConferenceEnable
Enables/Disables the call conference service.

Delayed Hotlinee
Endpoint Specific
epSpecificDelayedHotlineEnableConfig
Delayed Hotline
Activation
defaultDelayedHotlineEnable
epSpecificDelayedHotlineEnable
Enables/Disables the delayed hotline service.
Delayed Hotline
Condition
defaultDelayedHotlineCondition
epSpecificDelayedHotlineCondition
Selects the condition(s) that activate the delayed hotline.
Delayed Hotline Target
defaultDelayedHotlineTargetAddress
epSpecificDelayedHotlineTargetAddress
Address or telephone number of the target of the delayed

hotline.

Activation
defaultCallAllowDirectIp
Enables/Disables the direct IP address call service.

Holdf
Endpoint Specific
epSpecificHoldEnableConfig
Hold Activation
defaultHoldEnable
epSpecificHoldEnable
Enables/Disables the holding service.

Second Callg
Endpoint Specific
epSpecificSecondCallEnableConfig
Second Call Activation
defaultSecondCallEnable
epSpecificSecondCallEnable
Enables/Disables the second call service.
a. Mediatrix 3200 Series, 3300 Series, 3716, 3731, 3732, 3741, 3742, LP/4100/C7 Series only
b. Mediatrix 3200 Series, 3300 Series, 3716, 3731, 3732, 3741, 3742, LP/4100/C7 Series only
c. Mediatrix 3200 Series, 3300 Series, 3716, 3731, 3732, 3741, 3742, LP/4100/C7 Series only
d. Mediatrix 3200 Series, 3300 Series, 3716, 3731, 3732, 3741, 3742, LP/4100/C7 Series only
e. Mediatrix 3200 Series, 3300 Series, 3716, 3731, 3732, 3741, 3742, LP/4100/C7 Series only
f. Mediatrix 3200 Series, 3300 Series, 3716, 3731, 3732, 3741, 3742, LP/4100/C7 Series only
718
Telephony Page
g. Mediatrix 3200 Series, 3300 Series, 3716, 3731, 3732, 3741, 3742, LP/4100/C7 Series only
Tone Customization Sub-Page

Field Name
SNMP Variable
Description
Select Tone
countryCustomizationToneTone
Tone to customize.
Override Current Tone

Values
countryCustomizationToneOverride
Allows overriding the default country tone setting.
Current Tone Definition section

Field Name
SNMP Variable
Description
Frequencies
Value
Power
countryToneStatusPattern
Pattern description of the currently used tone for the country
Loop Count
Current Tone States section

Field Name
SNMP Variable
Description
State
On/Off
Frequencies
Duration
countryToneStatusPattern
Pattern description of the currently used tone for the country
Loop
Next State
Overriden Tone Definition section

Field Name
SNMP Variable
Description
Frequencies
Value
Power
countryCustomizationTonePattern
Pattern description of the custom tone.
Loop Count
Overriden Tone States section

Field Name
SNMP Variable
Description
State
On/Off
Frequencies
Duration
Loop
Next State
719
Telephony Page
Music on Hold Sub-Page

Field Name
SNMP Variable
Description
Transfer (command)
Saves the settings and transfers the MP3 file now.
CancelTransfer (command)
Saves the settings and stops a file transfer in progress.
Status Section
Field Name
SNMP Variable
Description
File Status
fileStatus
Status of the MP3 file in the unit.
Last Transfer Result
lastTransferStatus
Status of the last file transfer attempt.
Last Successful
Transfer
lastTransferDateTime
Date and time of the last successful music file transfer.
Music On Hold Configuration Section

Field Name
Streaming
SNMP Variable
musicOnHoldStreamingEnable
Description
Indicate whether or not the unit should play music when
being put on hold.
Transfer Configuration Section

Field Name
SNMP Variable
Description
URL
fileUrl
URL to a MP3 file which will be loaded at unit startup and

reloaded every time the ReloadInterval elapsed.
User Name
username
When authentication is required by the remote file server,

this variable will be used as the username.
Password
password
When authentication is required by the remote file server,

this variable will be used as the password.
Reload Interval
reloadInterval
Time, in hours, between attempts to load the MP3 file.
Misc Sub-Page
Country Section
Field Name
Country Selection
SNMP Variable
countrySelection
Description
List of predefined country settings.
Custom Tone Section

Field Name
SNMP Variable
Description
Override
countryCustomizationToneOverride
Overrides the default country tone setting.
Pattern
Call Detail Record Section

Field Name
Syslog Remote Host
720
SNMP Variable
syslogRemoteHost
Description
Host name and port number of the device that archives CDR
log entries.
Call Router Page
Field Name
SNMP Variable
Description
Syslog Format
syslogFormat
Specifies the format of the syslog Call Detail Record.
Syslog Facility
syslogFacility
Syslog facility used by the unit to route the Call Detail

Record messages.
Call Router Page

Status Sub-Page
Field Name
Config Modified
SNMP Variable
Description
Shows whether the configuration of the call routing was
Route Section
Field Name
Type
SNMP Variable
routeStatusType
Description
Displays the associated route type.
Source
routeStatusSourceCriteria
Source criteria to match to apply the route.
Properties Criteria
routeStatusPropertiesCriteria
Call properties criteria to match to apply the route.
Expression Criteria
routeStatusExpressionCriteria
Expression criteria to match to apply the route.
Mappings
routeStatusMapping
Name of the properties manipulation to apply to the call if

the criteria match.
Signaling Properties
routeStatusSignalingProperties
Name of the signaling properties to apply to the call.
Destination
routeStatusDestination
Destination to apply to the call if it matches the criteria.
Signaling Properties Section

Field Name
SNMP Variable
Description
Name
signalingPropertiesStatusName
Name of the Signaling properties defined by this row.
Early Connect
signalingPropertiesStatusEarlyConnect
Enables/Disables the early connect feature.
Early Disconnect
signalingPropertiesStatusEarlyDisconnect
Enables/Disables the early disconnect feature.
Destination Host
signalingPropertiesStatusDestinationHost
SIP messages destination.
Mapping Section
Field Name
SNMP Variable
Description
Criteria
mappingTypeStatusCriteria
mappingExpressionStatusCriteria
Expression or call property to compare with the call and

match in order to apply the properties manipulation.
Transformation
mappingTypeStatusTransformation
mappingExpressionStatusTransformation
Call properties to transform and transformation to apply to

the call properties.
Sub Mapping
mappingExpressionStatusSubMappings
Name of a subsequent properties manipulation to execute.
Hunt Section
Field Name
Name
SNMP Variable
huntStatusName
Description
Name of the hunt defined by this row.
721
Field Name
Destinations
Call Router Page
SNMP Variable
Description
huntStatusDestinations
List of hunt destinations separated by comma.
Selection Algorithm
huntStatusSelectionAlgorithm
Destination selection algorithm.
Timeout
huntStatusTimeout
Maximal time allowed to the destination to handle the call.
Causes
huntStatusCauses
List of call rejection causes to continue the hunt.
SIP Redirects Section

Field Name
SNMP Variable
Description
Name
sipRedirectStatusName
Name of the SIP Redirect defined by this row.
Destination Host
sipRedirectStatusDestination
Host address inserted in the Moved Temporarily response.
Available Interfaces Section

Field Name
SNMP Variable
Description
Index
interfaceStatusIndex
Name
interfaceStatusName
Name of the interface.
Route Configuration Sub-Page

Field Name
SNMP Variable
Description
ApplyConfig (command)
Applies the call routing configuration.
RollbackConfig (command)

configuration as showed in the status. The current
configuration will be lost.
Route Section
Field Name
SNMP Variable
Description
routeUp
routeDown
routeInsert
routeInsertRoute

Inserts a new row at the end of the Route table.
routeDelete
Deletes this row.
Configure Route Panel

Field Name
722
SNMP Variable
Description
Source
routeSourceCriteria
Source criteria to match to apply the route.
Properties Criteria
routePropertiesCriteria
Call properties criteria to match to apply the route.
Expression Criteria
routeExpressionCriteria
Expression criteria to match to apply the route.
Mappings
routeMapping
Name of the properties manipulation to apply to the call if

the criteria match.
Signaling Properties
routeSignalingProperties
Name of the signaling properties to apply to the call.
Destination
routeDestination
Destination to apply to the call if the call matches the criteria.
Config Status
routeConfigStatus
Call Router Page
Mapping Type Section

Field Name
SNMP Variable
Description
mappingTypeUp
mappingTypeDown
mappingTypeInsert
mappingTypeInsertMappingType

Inserts a new row at the end of the MappingType table.
mappingTypeDelete
Deletes this row.
Configure Mapping Type Panel

Field Name
SNMP Variable
Description
Name
mappingTypeName
Name of the the properties manipulation.
Criteria
mappingTypeCriteria
Call properties that the service must compare with the call
and match in order to apply the properties manipulation.
Transformation
mappingTypeTransformation
Call properties to transform.
Config Status
mappingTypeConfigStatus
It indicates whether the configuration of the row is valid.
Mapping Expression Section

Field Name
SNMP Variable
Description
mappingExpressionUp
mappingExpressionDown
mappingExpressionInsert
mappingExpressionInsertMappingExpression

Inserts a new row at the end of the MappingExpression
table.
mappingExpressionDelete
Deletes this row.
Configure Mapping Expression Panel

Field Name
SNMP Variable
Description
Name
mappingExpressionName
Name of the properties manipulation.
Criteria
mappingExpressionExpressionCriteria
Expression to compare with the call and match in order to

apply the properties manipulation.
Transformation
mappingExpressionTransformation
Transformation to apply to the call properties.
Sub Mappings
mappingExpressionSubMapping
Name of a subsequent properties manipulation to execute.
Config Status
mappingExpressionConfigStatus
It indicates whether the configuration of the row is valid.
Signaling Properties Section

Field Name
SNMP Variable
Description
signalingPropertiesUp
signalingPropertiesDown
signalingPropertiesInsert
signalingPropertiesInsertSignalingProperties

Inserts a new row at the end of the Signaling Properties
table.
723
Field Name
Call Router Page
SNMP Variable
signalingPropertiesDelete
Description
Deletes this row.
Configure Signaling Properties Panel

Field Name
SNMP Variable
Description
Name
signalingPropertiesName
Name of the Signaling properties defined by this row.
Early Connect
signalingPropertiesEarlyConnect
Enables/Disables the early connect feature.
Early Disconnect
signalingPropertiesEarlyDisconnect
Enables/Disables the early disconnect feature.
Destination Host
signalingPropertiesDestinationHost
SIP messages destination.
Allow 180 with SDP
signalingPropertiesAllow180Sdp
Enables/Disables the 180 with SDP allowed.
Allow 183 without SDP
signalingPropertiesAllow183NoSdp
Enables/Disables the 183 without SDP allowed.
Privacy
signalingPropertiesPrivacy
Sets the privacy level of the call.
SIP Headers
Translations
signalingPropertiesSipHeadersTranslation
Name of the SIP headers translation to apply to the call.
Call Properties
Translations
signalingPropertiesCallPropertiesTranslation
Name of the call properties translation to apply to the call.
Config Status
signalingPropertiesConfigStatus
SIP Headers Translations Section

Field Name
SNMP Variable
Description
sipHeadersTranslationUp
sipHeadersTranslationDown
sipHeadersTranslationInsert
sipHeadersTranslation

Inserts a new row at the end of the SIP Headers Translation
table.
sipHeadersTranslationDelete
Deletes this row.
Configure SIP Headers Translation Panel

Field Name
SNMP Variable
Description
Name
sipHeadersTranslationName
Name of the SIP headers translation defined by this row.
SIP Header
sipHeadersTranslationSipHeader
Sets which SIP header is modified by this translation.
Built From
sipHeadersTranslationBuiltFrom
Sets what information is used to build the selected SIP

header.
Fix Value
sipHeadersTranslationFixValue
Fix value to be inserted in the SIP header.
Config Status
sipHeadersTranslationConfigStatus
Call Properties Translations Section

Field Name
724
SNMP Variable
Description
callPropertiesTranslationUp
callPropertiesTranslationDown
callPropertiesTranslationInsert
callPropertiesTranslationInsertCallPropertiesTr
anslation

Inserts a new row at the end of the SIP Headers Translation
table.
Call Router Page
Field Name
SNMP Variable
callPropertiesTranslationDelete
Description
Deletes this row.
Configure Call Properties Translations Panel

Field Name
SNMP Variable
Description
Name
callPropertiesTranslationName
Name of the call properties translation defined by this row.
Call Property
callPropertiesTranslationCallProperty
Sets which call property is modified by this translation.
Built From
callPropertiesTranslationBuiltFrom
Sets what information is used to build the selected call

property.
Fix Value
callPropertiesTranslationFixValue
Fix value to be inserted in the call property.
Config Status
callPropertiesTranslationConfigStatus
Hunt Section
Field Name
SNMP Variable
Description
huntUp
huntDown
huntInsert
huntInsertHunt

Inserts a new row at the end of the Hunt table.
huntDelete
Deletes this row.
Configure Hunt Panel

Field Name
SNMP Variable
Description
Name
huntName
Name of the hunt defined by this row.
Destinations
huntDestinations
List of hunt destinations separated by comma.
Selection Algorithm
huntSelectionAlgorithm
Destination selection algorithm.
Timeout
huntTimeout
Maximal time allowed to the destination to handle the call.
Causes
huntCauses
List of call rejection causes to continue the hunt.
Config Status
huntConfigStatus
SIP Redirects Section

Field Name
SNMP Variable
Description
sipRedirectUp
sipRedirectDown
sipRedirectInsert
sipRedirectInsertSipRedirect

Inserts a new row at the end of the SIP Redirects table.
sipRedirectDelete
Deletes this row.
Configure SIP Redirects Panel

Field Name
Name
SNMP Variable
sipRedirectName
Description
Name of the SIP Redirect defined by this row.
725
Field Name
Management Page
SNMP Variable
Description
Destination Host
sipRedirectDestinations
Host address to be inserted in the Moved Temporarily

response.
Config Status
sipRedirectConfigStatus
Configuration status of the row. It indicates whether the

configuration of the row is valid.
Auto-Routing Sub-Page
Field Name
SNMP Variable
Description
Auto-routing
autoRoutingEnable
Enables/Disables the automatic insertion of default routes

for selected endpoints.
Criteria Type
autoRoutingCriteriaType
Determines the type of criteria to use to create automatic

rule from SIP to the telephony endpoints.
Incoming Mappings
autoRoutingIncomingMappings
Name of the properties manipulations associated with the

route from the SIP gateway to the endpoint.
Outgoing Mappings
autoRoutingOutgoingMappings
Name of the properties manipulations associated with the

route from the endpoint to the SIP gateway.
Incoming Signaling
Properties
autoRoutingIncomingSignalingProperties
Name of the signaling properties associated with the route

from the SIP gateway to the endpoint.
Outgoing Signaling
Properties
autoRoutingOutgoingSignalingProperties
Name of the signaling properties associated with the route

from the endpoint to the SIP gateway.
Endpoints auto-routing section

Field Name
SNMP Variable
Description
Endpoint
autoRoutingEpId
Character string that identifies an endpoint in other tables.
Auto-routable
autoRoutingAutoroutable
Determines whether or not automatic routes are generated

for the endpoint when auto-routing is enabled.
Auto-routing Gateway
autoRoutingAutoRoutingGateway
Name of the SIP gateway to use as the destination of

outgoing calls and the source of incoming calls when
generating auto-routing rules.
E164
autoRoutingE164
The telephone number associated with this endpoint, if any.
SIP Username
autoRoutingSipUsername
The SIP username associated with this endpoint, if any.
Name
autoRoutingName
The FriendlyName associated with this endpoint, if any.
Management Page
Configuration Scripts Sub-Page
Field Name
SNMP Variable
Description
ConfiguredScriptExport (command)
Command to export the configuration script.
ConfiguredScriptsTransferAndRun (command)
Command to launch the configuration scripts download.
Scripts Status Section

Field Name
Current State (Export)
726
SNMP Variable
scriptsStatsCurrentTransferState
Description
The current state of the configuration script transfer and
execution.
Management Page
Field Name
SNMP Variable
Description
Current State
(Execute)
ScriptsStatsCurrentExportState
The current state of the configuration script exportation.
Last Result (Export)
scriptsStatsLastTransferResult
Result of the last configuration scripts transfer command.
Last Result (Execute)
ScriptsStatsLastExportResult
Result of the last configuration script exportation command.
Last Successful
(Export)
scriptsStatsLastTransferDateTime
Date and time of the last successful configuration script

transfer command.
Last Successful
(Execute)
ScriptsStatsLastExportDateTime
Date and time of the last successful configuration script

exportation and transfer command since the last reset to
default settings.
Export Script Section

Field Name
SNMP Variable
Description
Content
scriptExportContent
Content to export in the generated configuration script.
Service Name
ScriptExportServiceName
Name of the service from which to export configuration.
Send To URL
scriptExportUrl
URL where to send the configuration script exported.
Privacy Key
scriptExportSecretKey
Key used to encrypt the configuration script to export.
Execute Scripts Section

Field Name
SNMP Variable
Description
Generic File Name
scriptGenericFileName
Name of the generic configuration script.
Specific File Name
scriptSpecificFileName
Name of the specific configuration script.
Transfer Protocol
scriptsTransferProtocol
Protocol used to transfer the configuration script files.
Host Name
scriptsTransferSrvHostname
Configuration scripts server hostname and port.
Location
scriptsLocation
Path to the location of the configuration scripts.
User Name
scriptsTransferUsername
User name used to transfer the configuration script.
Password
scriptsTransferPassword
Password used to transfer the configuration script.
Privacy Key
scriptsSecretKey
Key used to decrypt encrypted configuration scripts.
Allow Repeated
Execution
scriptsAllowRepeatedExecution
Allows the execution of a script even if it is identical to the

last executed script.
Automatically Update Scripts Section

Field Name
SNMP Variable
Description
Update On Restart
scriptsTransferOnRestartEnable
Enables automatic configuration scripts transfer on restart.
Update Periodically
scriptsTransferPeriodicEnable
Enables automatic periodic configuration scripts transfer.
Time Unit
scriptsTransferPeriodicTimeUnit
Time unit for the variable scriptsTransferInterval.
Period
scriptsTransferInterval
Time interval between automatic configuration scripts

transfer.
Time Range
scriptsTransferTimeRange
Time range when the automatic configuration scripts

transfer occurs.
DHCP Download
Enable
scriptsDhcpDownloadEnable
DHCP Triggered Script Support.
727
Management Page
Transfer Scripts Through Web Browser Section

Field Name
SNMP Variable
Description
Upload Parameters
N/A
N/A
Content
scriptExportContent
Content to export in the generated configuration script.
Privacy Key
scriptsSecretKey
Key used to decrypt encrypted configuration scripts.
Backup / Restore Sub-Page

Field Name
SNMP Variable
Description
ConfiguredBackupImage (command)
Command to launch the configuration backup.
ConfiguredRestoreImage (command)
Command to launch the configuration restore.
Status Section
Field Name
SNMP Variable
Description
Last Backup Result
imageBackupStatus
Result of the last configuration backup command.
Last Restore Result
imageRestoreStatus
Result of the last configuration restore command.
Image Configuration Section

Field Name
SNMP Variable
Description
File Name
imageFileName
Name of the file used to backup (save) and restore (load)

the units configuration.
Transfer Protocol
imageTransferProtocol
Protocol used to upload a configuration image during

backup and transfer during restore.
Host Name
imageTransferSrvHostname
Configuration backup/restore server hostname and port.
Location
imageLocation
Path to the location of the configuration image file.
User Name
imageTransferUsername
User name used to transfer the configuration image.
Password
imageTransferPassword
Password used to transfer the configuration image.
Content
imageBackupContent
Defines what to include in the backup.
Privacy Algorithm
imagePrivacyAlgo
Enables/disables decryption of the configuration image.
Privacy Key
imageSecretKey
Key used to decrypt or encrypt a configuration image.
Firmware Upgrade Sub-Page

Field Name
SNMP Variable
Install (command)
Description
Command to launch the firmware download.
Status Section
Field Name
728
SNMP Variable
Description
Firmware Pack
Updater Status
status
Indicates the current status of the Firmware Pack Updater.
Last Installation Result
mfpLastInstallationResult
Result of the last install command.
Last Successful
Installation
mfpLastInstallationDateTime
Date and time of the last successful install command.
Management Page
Firmware Packs Installed

Field Name
SNMP Variable
Description
Name
mfpSelectionMfpname
Name of the Firmware Pack to install.
Version
mfpVersion
Version of the MFP to install.
Profile
mfpProfileName
Name of the profile.
Bank
mfpInstalledInfoMfpBank
Bank where the MFP is installed.
Rollback (command)
Launches the rollback of the previously installed MFP found

in recovery bank.

Field Name
SNMP Variable
Description
Language
languageSelection
Language.
Version
mfpVersion
Version of the MFP to install.
Automatic Restart
Enable
automaticRestartEnable
Enables the firmware pack updater to automatically restart

the system when needed for completing a firmware update
operation.
Automatic Restart
Grace Delay
automaticRestartGraceDelay
Configures the grace delay in minutes that the unit waits for
all telephony calls to be terminated before the automatic
restart can occur.
Firmware Pack 1
mfpSelectionMfpName
Firmware Pack 2
mfpSelectionMfpName
Firmware Pack 3
mfpSelectionMfpName
Firmware Pack 4
mfpSelectionMfpName
Firmware Pack 5
mfpSelectionMfpName
Field Name
SNMP Variable
Description
Location
mfpLocation
Path to the directory containing MFPs.
Transfer Protocol
mfpTransferProtocol
Protocol to use to access the update tree.
User Name
mfpTransferUsername
User name to use to access the update tree.
Password
mfpTransferPassword
Password to use to access the update tree.
Host Name
mfpTransferSrvHostname
Name or IP address and port of the Update Files server.
Certificates Sub-Page
Host Certificates Section
Field Name
SNMP Variable
Description
File Name
hostCertificatesInfoFileName
Name of the certificate file.
Issued To
hostCertificatesInfoIssuedTo
Certificate subject name. This is the common name that

must match the host being authenticated.
Issued By
hostCertificatesInfoIssuedBy
Certificate issuer name. This is the certificate authority that

signed this certificate.
Valid From
hostCertificatesInfoValidFrom
Certificate lower bound validity duration range.
Valid To
hostCertificatesInfoValidTo
Certificate higher bound validity duration range.
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Field Name
Usage
Management Page
SNMP Variable
Description
hostCertificatesAuthenticationUsage
Identifies in which role or context a certificate can be used

by the host it authenticates.
hostCertificatesInfoDelete
Removes the certificate from the unit.
Others Certificates Section

Field Name
SNMP Variable
Description
File Name
othersCertificatesInfoFileName
Name of the certificate file.
Issued To
othersCertificatesInfoIssuedTo
Certificate subject name. This is the common name that

must match the host being authenticated.
Issued By
othersCertificatesInfoIssuedBy
Certificate issuer name. This is the certificate authority that

signed this certificate.
Valid From
othersCertificatesInfoValidFrom
Certificate lower bound validity duration range.
Valid To
othersCertificatesInfoValidTo
Certificate higher bound validity duration range.
Usage
othersCertificatesAuthenticationUsage
Identifies in which role or context a certificate can be used

by the host it authenticates.
CA
othersCertificatesInfoCertificateAuthority
Indicates if the certificate is a CA certificate.
othersCertificatesInfoDelete
Removes the certificate from the unit.
Host Certificate Associations Section

Field Name
SNMP Variable
Description
File Name
hostCertificateAssociationFileName
Certificate file name.
SIP
hostCertificateAssociationSip
Specifies if this certificate can be used for SIP security.The

default value is Enabled.
Web
hostCertificateAssociationWeb
Specifies if this certificate can be used for Web security.The

EAP
hostCertificateAssociationEap
Specifies if this certificate can be used for EAP security. The

Conf
hostCertificateAssociationConf
Specifies if this certificate can be used for Conf security. The

Fpu
hostCertificateAssociationFpu
Specifies if this certificate can be used for Fpu security. The

File
hostCertificateAssociationFile
Specifies if this certificate can be used for File security. The

Cert
hostCertificateAssociationCert
Specifies if this certificate can be used for Cert security. The

Certificate Authorities Section

Field Name
SNMP Variable
Description
File Name
certificateAuthoritiesFileName
Certificate authority (CA) file name.
Override OCSP URL
certificateAuthoritiesOverrideIssuedCertificateO
cspUrl
Defines a specific OCSP URL to use for certificate

revocation status of certificates issued by this certificate
authority (CA).
SNMP Sub-Page
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Management Page

Field Name
SNMP Variable
Description
SNMP Listening Port
port
Port on which the SNMP service should listen for incoming

SNMP requests.
Enable SNMP V1
enableSnmpV1
Specifies if a user can connect to the system by using

SNMPv1.
Enable SNMP V2
enableSnmpV2

SNMPv2.
Enable SNMP V3
enableSnmpV3

SNMPv3.
Authentication Protocol
authProtocol
Protocol to use with SNMPv3.
Privacy Protocol
privProtocol
Protocol to use with SNMPv3.
Privacy Password
privPassword
Password to use with SNMPv3 when using DES privacy.
Community
community
String to use for the community field of SNMPv1 and

SNMPv2 read-write commands and traps.
Enable SNMP Trap
enableTrap
Specifies if traps can be sent.
Trap Destination
trapDest
Addresses/FQDNs and ports where to send traps.Up to 5

destinations can be specified by using a comma between
them. The port numbers are optional.
CWMP Sub-Page
Field Name
SNMP Variable
Description
Root Element
RootElement
Determines the data model used for the configuration.
Listening Port
ListeningPort
Port on which the unit listens for incoming TR-069

connections.
ACS Status
AcsStatus
Indicates the status of the connection with the ACS.
ACS URL Config

Source
AcsUrlConfigSource
Determines the method to obtain the URL of the ACS.
ACS Static URL
AcsStaticUrl
URL used by the unit to connect to the ACS using the CPE
WAN Management Protocol if the AcsUrlConfigSource
variable is set to 'Static'.
Username
Username
Username used to authenticate the unit when making a

connection to the ACS using the CPE WAN Management
Protocol.
Password
Password
Password used to authenticate the unit when making a

connection to the ACS the CPE WAN Management
Protocol.
This variable contains a secret value.
Periodic Inform Enable
Defines whether or not the unit needs to periodically send

CPE information to the ACS using the Inform method call.
Periodic Inform Interval
The duration, in seconds, of the interval for which the unit

needs to attempt to connect with the ACS and call the
Inform method.
Periodic Inform Time
PeriodicInformTime
An absolute time reference in UTC format to determine

when the unit initiates the periodic Inform method calls.
Annex-F
Tr069AnnexFEnable
Enables/disables the support of variables under the TR-069

Device.GatewayInfo tree.
TR-104 Enable
Tr104Enable
Enables/disables the support of variables under the TR-069

Device.Services.VoiceService tree.
Network Interface
Tr106LanNetworkInterface
Network interface referred by the TR-106 LAN profile.
STUN Status
Tr111StunStatus
Indicates the status of the connection with the STUN server.
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Field Name
Management Page
SNMP Variable
Description
STUN Enable
Tr111StunEnable
Enables/disables the use of STUN binding requests to

discover the units external IP address and port and to set
the ManagementServer.UDPConnectionRequestAddress
value accordingly.
NAT Detected
Tr111NatDetected
When Tr111StunEnable is enabled, this parameter indicates

whether or not the unit has detected the address and/or port
mapping in use.
STUN Server
Tr111StunServerHost
Host name or IP address of the STUN server for the unit to

send Binding Requests if Tr111StunEnable is enabled.
Access Control Sub-Page

Users Section
Field Name
SNMP Variable
Description
User Name
usersUserName
Contains the user name.
Password
usersPassword
Contains the user's password.
Access Rights
usersAccessRights
Defines the access rights template applying to a user.
InsertUser
Inserts a new user in the Users table.
Delete (Row Command)
Deletes this row.
Services Access Control Type Section

Field Name
SNMP Variable
Description
Service
servicesAaaTypeService
Service name for which the Aaa types are configured.
Authentication Type
servicesAaaTypeAuthenticationType
Authentication type a service uses for incoming

authentication requests.
Accounting Type
servicesAaaTypeAccountingType
Accounting type a service uses once a user is successfully

authenticated on the unit.
Radius Servers Section

Field Name
SNMP Variable
Description
Authentication
Host
radiusServersAuthenticationHost
Hostname and port of a Radius server used for

authentication requests.
Server Secret
radiusServersAuthenticationSecret
Secret key shared between the Radius server and the unit.
Accounting
Host
radiusServersAccountingHost
Hostname and port of a Radius server used for accounting

requests.
Server Secret
radiusServersAccountingSecret
Secret key shared between the Radius server and the unit.
General
Server Request
Timeout
radiusServersTimeoutS
Maximum time, in seconds, the unit waits for a reply from a

Radius server.
Radius Users Access

Rights
radiusUserAccessRights
Defines the access rights template applying to all Radius

users.
File Sub-Page
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Management Page
Internal Files Section

Field Name
SNMP Variable
Description
Name
filesFilename
Relative path and name of the file.
Description
filesFileDescription
Textual description describing the content of the file.
Size
filesFileSize
File size of the associated file.
filesDelete
Deletes this row.
Misc Sub-Page
System Management Section
Field Name
Network Interface
SNMP Variable
managementInterface
Description
Specifies to which network interface system management
services are bound.
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Management Page
P P E N D I X
Glossary
10 BaseT
An Ethernet local area network that works on twisted pair wiring.
100 BaseT
A newer version of Ethernet that operates at 10 times the speed of a 10 BaseT Ethernet.
Access Device
Device capable of sending or receiving data over a data communications channel.
Accounting
Accounting measures the resources a user consumes during access. This can include the amount of system
time or the amount of data a user has sent and/or received during a session. Accounting is carried out by
logging of session statistics and usage information and is used for authorization control, billing, trend analysis,
resource utilization, and capacity planning activities.
A-Law
The ITU-T companding standard used in the conversion between analog and digital signals in PCM (Pulse
Code Modulation) systems. A-law is used primarily in European telephone networks and contrasts with the
North American mu ()-law standard. See also mu ()-law.
ANI
In CAS signalling, the sending of the calling numbers is known as Automatic Number Identification.
AOC
In ISDN signalling, an Advice Of Charge (AOC-D) message is sent to advise of the current charge (D)uring a
call or an AOC-E message is sent to advise of the total charge at the (E)nd of a call.
Area Code
The preliminary digits that a user must dial to be connected to a particular outgoing trunk group or line. In North
America, an area code has three digits and is used with a NXX (office code) number. For instance, in the North
American telephone number 561-955-1212, the numbers are defined as follows:
Table 546: North American Numbering Plan
No.
Description
561
Area Code, corresponding to a geographical zone in a non-LNP (Local Number Portability)

network.
955
NXX (office code), which corresponds to a specific area such as a city region.
1212
Unique number to reach a specific destination.
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Appendix E - Glossary
Outside North America, the area code may have any number of digits, depending on the national
telecommunication regulation of the country. In France, for instance, the numbering terminology is xZABPQ
12 34, where:
Table 547: France Numbering Plan
No.
Description
Operator forwarding the call. This prefix can be made of 4 digits.
Geographical (regional) zone of the number (in France, there are five zones). It has two digits.
ABPQ
First four digits corresponding to a local zone defined by central offices.
12 34
Unique number to reach a specific destination.
In this context, the area code corresponds to the Z portion of the numbering plan. Because virtually every
country has a different dialing plan nomenclature, it is recommended to identify the equivalent of an area code
for the location of your communication unit.
Authentication
Authentication provides a way of identifying a user, typically by having the user enter a valid user name and
valid password before access is granted. The process of authentication is based on each user having a unique
set of criteria for gaining access. The AAA server compares a user's authentication credentials with other user
credentials stored in a database. If the credentials match, the user is granted access to the network. If the
credentials are at variance, authentication fails and network access is denied.
Basic Rate Interface (BRI)
An Integrated Services Digital Network configuration defined in the physical layer standard I.430 produced by
the ITU. This configuration consists of two 64 kbit/s bearer channels (B channels) and one 16 kbit/s data
channel (D channel). The B channels are used for voice or user data, and the D channel is used for any
combination of: data, control/signalling and X.25 packet networking. The two B channels can be bonded
together giving a total data rate of 128 kbit/s. BRI is the kind of ISDN interface most likely to be found in
residential service.
Call Routing
Calls through the Mediatrix can be routed based on a set of routing criteria.
Channel Associated Signalling (CAS)
With this method of signalling, each traffic channel has a dedicated signalling channel. In other words the
signalling for a particular traffic circuit is permanently associated with that circuit. Channel-associated callcontrol is still widely used today mostly in South America, Africa, Australia and in Europe.
Country Code (CC)
In international direct telephone dialing, a code that consists of 1-, 2-, or 3-digit numbers in which the first digit
designates the region and succeeding digits, if any, designate the country.
Dialed Number Identification Service (DNIS)
DNIS is a telephone service that identifies for the receiver of a call the number that the caller dialed. It's a
common feature of 800 and 900 lines. If you have multiple 800 or 900 numbers to the same destination, DNIS
tells which number was called. DNIS works by passing the touch tone digits (dual tone multi frequency or MF
digits) to the destination where a special facility can read and display them or make them available for call
center programming.
Digital Subscriber Lines (DSL)
A technology for bringing high-bandwidth information to homes and small businesses over ordinary copper
telephone lines. xDSL refers to different variations of DSL, such as ADSL, HDSL, and RADSL.
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Distinguished Encoding Rules (DER)

DER for ASN.1, as defined in ITU-T Recommendation X.509, is a more restrictive encoding standard than the
alternative BER (Basic Encoding Rules) for ASN.1, as defined in ITU-T Recommendation X.209, upon which
DER is based. Both BER and DER provide a platform-independent method of encoding objects such as
certificates and messages for transmission between devices and applications.
Domain Name Server (DNS)
Internet service that translates domain names into IP addresses. For instance, the domain name
www.example.com might translate to 198.105.232.4.
Dual-Tone Multi-Frequency (DTMF)
In telephone systems, multi-frequency signalling in which a standard set combinations of two specific voice
band frequencies, one from a group of four low frequencies and the other from a group of four higher
frequencies, are used. Although some military telephones have 16 keys, telephones using DTMF usually have
12 keys. Each key corresponds to a different pair of frequencies. Each pair of frequencies corresponds to one
of the ten decimal digits, or to the symbol # or *, the * being reserved for special purposes.
Dynamic Host Configuration Protocol (DHCP)
TCP/IP protocol that enables PCs and workstations to get temporary or permanent IP addresses (out of a pool)
from centrally-administered servers.
Echo Cancellation
Technique that allows for the isolation and filtering of unwanted signals caused by echoes from the main
transmitted signal.
Far End Disconnect
Refers to methods for detecting that a remote party has hung up. This is also known as Hangup Supervision.
There are several methods that may be used by a PBX/ACD/CO to signal that the remote party has hung up,
including cleardown tone, or a wink.
Federal Communications Commission (FCC)
U.S. government regulatory body for radio, television, interstate telecommunications services, and
international services originating in the United States.
Firewall
A firewall in a networked environment prevents some communications forbidden by the security policy. It has
the basic task of controlling traffic between different zones of trust. Typical zones of trust include the Internet
(a zone with no trust) and an internal network (a zone with high trust).
Foreign Exchange Office (FXO)
A network-provided service in which a telephone in a given local exchange area is connected, via a private
line, to a central office in another, i.e., foreign, exchange, rather than the local exchange areas central office.
This is the office end of an FX circuit (frequently a PBX).
Foreign Exchange Service/Station (FXS)
A network-provided service in which a telephone in a given local exchange area is connected, via a private
line, to a central office in another, i.e., foreign, exchange, rather than the local exchange areas central office.
This is the station (telephone) end of an FX circuit. An FXS port will provide dial tone and ring voltage.
Full-Duplex Connection
Refers to a transmission using two separate channels for transmission and reception and that can transmit in
both ways at the same time. See also Half-Duplex Connection.
735
G.703
ITU-T recommendation for the physical and electrical characteristics of hierarchical digital interfaces at rates
up to 140Mbit/s.
G.704
ITU-T recommendation for synchronous frame structures on G.703 interfaces up to 45Mbit/s. The
conventional use of G.704 on a 2Mbit/s primary rate circuit provides 30 discrete 64kbit/s channels, with a
further 64kbit/s channel available for common channel signalling.
G.711
Algorithm designed to transmit and receive A-law PCM (Pulse Code Modulation) voice at digital bit rates of 48
kbps, 56 kbps, and 64 kbps. It is used for digital telephone sets on digital PBX and ISDN channels.
G.723.1
A codec that provides the greatest compression, 5.3 kbps or 6.3 kbps; typically specified for multimedia
applications such as H.323 videoconferencing.
G.726
An implementation of ITU-T G.726 standard for conversion linear or A-law or -law PCM to and from a 40, 32,
24 or 16 kbit/s channel.
G.729
A codec that provides near toll quality at a low delay which uses compression to 8 kbps (8:1 compression rate).
Gateway
A device linking two different types of networks that use different protocols (for example, between the packet
network and the Public Switched Telephone Network).
Half-Duplex Connection
Refers to a transmission using the same channel for both transmission and reception therefore it can't transmit
and receive at the same time. See also Full-Duplex Connection.
Hunt Group
The hunt group hunts an incoming call to multiple interfaces. It accepts a call routed to it by a routing table or
directly from an interface and creates another call that is offered to one of the configured destination interfaces.
If this destination cannot be reached, the hunt group tries another destination until one of the configured
destinations accepts the call. When an interface accepts a call, the interface hunting is complete and the hunt
group service merges the original call with the new call to the interface that accepted the call.
Impedance
Impedance is the apparent resistance, in an electric circuit, to the flow of an alternating current, analogous to
the actual electrical resistance to a direct current, being the ratio of electromotive force to the current.
Information Transfer Capability (ITC)
A request to the network exchange equipment to ask if a particular type of encoding is allowed. It is also
called ISDN bearer capability or ISDN service.
Integrated Services Digital Network (ISDN)
A set of digital transmission protocols defined by the international standards body for telecommunications, the
ITU-T (formerly called the CCITT). These protocols are accepted as standards by virtually every
telecommunications carrier all over the world.
ISDN complements the traditional telephone system so that a single pair of telephone wires is capable of
carrying voice and data simultaneously. It is a fully digital network where all devices and applications present
themselves in a digital form.
736
International Telecommunication Union (ITU)

Organization based in Geneva, Switzerland, that is the most important telecom standards-setting body in the
world.
Internet-Drafts
Internet-Drafts are working documents of the IETF, its areas, and its working groups. Note that other groups
may also distribute working documents as Internet-Drafts.
Internet Protocol (IP)
A standard describing software that keeps track of the Internets addresses for different nodes, routes outgoing
messages, and recognizes incoming messages.
IP Forwarding
Allows the packet to be forwarded to a specific network based on the packets criteria (source IP address and
source Ethernet link).
Jitter
A distortion caused by the variation of a signal from its references which can cause data transmission errors,
particularly at high speeds.
Light Emitting Diode (LED)
A semiconductor diode that emits light when a current is passed through it.
Local Area Network (LAN)
Data-only communications network confined to a limited geographic area, with moderate to high data rates.
See also WAN.
Local Firewall
Allows you to dynamically create and configure rules to filter incoming packets with the unit as destination. The
traffic is analyzed and filtered by all the rules configured.
Management Information Base (MIB)
Specifications containing definitions of management information so that networked systems can be remotely
monitored, configured and controlled.
Media Access Control (MAC) Address
A layer 2 address, 6 bytes long, associated with a particular network device; used to identify devices in a
network; also called hardware or physical address.
Mu ()-Law
The PCM (Pulse Code Modulation) voice coding and companding standard used in Japan and North America.
See also A-Law.
Music on Hold (MoH)
Refers to the practice of playing pre-recorded music to fill the silence that would be heard by telephone callers
who have been placed on hold. It is especially common in situations involving customer service.
Network
A group of computers, terminals, and other devices and the hardware and software that enable them to
exchange data and share resources over short or long distances. A network can consist of any combination
of local area networks (LAN) or wide area networks (WAN).
737
Network Address Translation (NAT)

NAT, also known as network masquerading or IP masquerading, rewrites the source and/or destination
addresses/ports of IP packets as they pass through a router or firewall. It is most commonly used to connect
multiple computers to the Internet (or any other IP network) by using one IP address. This allows home users
and small businesses to cheaply and efficiently connect their network to the Internet. The basic purpose of
NAT is to multiplex traffic from the internal network and present it to the Internet as if it was coming from a
single computer having only one IP address.
There are two types of NAT rules:
Source rules: They are applied on the source address of outgoing packets.
Destination rules: They are applied on the destination address of incoming packets.
Network Firewall
Allows dynamically creating and configuring rules to filter packets forwarded by the unit. Since this is a network
firewall, rules only apply to packets forwarded by the unit. The traffic is analyzed and filtered by all the rules
configured.
Off-hook
A line condition caused when a telephone handset is removed from its cradle.
On-hook
A line condition caused when a telephone handset is resting in its cradle.
Packet
Includes three principal elements: control information (such as destination, origin, length of packet), data to be
transmitted, and error detection. The structure of a packet depends on the protocol.
Plain Old Telephone System (POTS)
Standard telephone service used by most residential locations; basic service supplying standard single line
telephones, telephone lines, and access to the public switched network.
Point to Point Protocol over Ethernet (PPPoE)
A proposal specifying how a host personal computer interacts with a broadband modem (i.e., DSL, cable,
wireless, etc.) to access the growing number of Highspeed data networks. Relying on two widely accepted
standards, Ethernet and the point-to-point protocol (PPP), the PPPoE implementation requires virtually no
more knowledge on the part of the end user other than that required for standard Dialup Internet access. In
addition, PPPoE requires no major changes in the operational model for Internet Service Providers (ISPs) and
carriers. The base protocol is defined in RFC 2516.
Port
Network access point, the identifier used to distinguish among multiple simultaneous connections to a host.
Portable Operating System Interface (POSIX)
POSIX is a set of standard operating system interfaces based on the UNIX operating system. The need for
standardization arose because enterprises using computers wanted to be able to develop programs that could
be moved among different manufacturer's computer systems without having to be recoded.
Primary Rate Interface (PRI)
A telecommunications standard for carrying multiple DS0 voice and data transmissions between two physical
locations. All data and voice channels are (ISDN) and operate at 64 kbit/s.
North America and Japan use a T1 of 23 B channels and one D channel which corresponds to a T1 line.
Europe, Australia and most of the rest of the world use the slightly higher capacity E1, which is composed of
31 B channels and one D channel.
Fewer active B channels (also called user channels) can be used for a fractional T1. More channels can be
used with more T1's, or with a fractional or full T3 or E3.
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Presentation Indicator (PI)

An information element (IE) field that determines whether a callers CLI can be displayed on a Caller ID
device or otherwise presented to the called party.
Private Branch Exchange (PBX)
A small to medium sized telephone system and switch that provides communications between onsite
telephones and exterior communications networks.
Protocol
A formal set of rules developed by international standards bodies, LAN equipment vendors, or groups
governing the format, control, and timing of network communications. A set of conventions dealing with
transmissions between two systems. Typically defines how to implement a group of services in one or two
layers of the OSI reference model. Protocols can describe low-level details of machine-to-machine interfaces
or high-level exchanges between allocation programs.
Proxy Server
An intermediary program that acts as both a server and a client for the purpose of making requests on behalf
of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other
servers. A proxy interprets, and, if necessary, rewrites a request message before forwarding it.
Public Switched Telephone Network (PSTN)
The local telephone company network that carries voice data over analog telephone lines.
QSIG
QSIG is an ISDN based signaling protocol for signaling between private branch exchanges (PBXs) in a Private
Integrated Services Network (PISN). It makes use of the connection-level Q.931 protocol and the applicationlevel ROSE protocol. ISDN "proper" functions as the physical link layer.
Quality of Service (QoS)
Measure of the telephone service quality provided to a subscriber. This could be, for example, the longest time
someone should wait after picking up the handset before they receive dial tone (three seconds in most U.S.
states).
Real Time Control Protocol (RTCP)
RTCP is the control protocol designed to work in conjunction with RTP. It is standardized in RFC 1889 and
1890. In an RTP session, participants periodically send RTCP packets to convey feedback on quality of data
delivery and information of membership.
Realtime Transport Protocol (RTP)
An IETF standard for streaming realtime multimedia over IP in packets. Supports transport of real-time data
like interactive voice and video over packet switched networks.
Registrar Server
A server that accepts REGISTER requests. A registrar is typically co-located with a proxy or redirect server
and MAY offer location services.
Request for Comment (RFC)
A formal document from the IIETF that is the result of committee drafting and subsequent review by interested
parties. Some RFCs are informational in nature. Of those that are intended to become Internet standards, the
final version of the RFC becomes the standard and no further comments or changes are permitted. Change
can occur, however, through subsequent RFCs that supersede or elaborate on all or parts of previous RFCs.
Screening Indicator (SI)
A service provided by ISDN that can be used to test the trustworthiness of the calling partys number. This
signalling-related information element is found in octet 3a of the ISDN SETUP message.
739
Server
A computer or device on a network that works in conjunction with a client to perform some operation.
Session Description Protocol (SDP)
Describes multimedia sessions for the purpose of session announcement, session invitation and other forms
of multimedia session initiation. SDP communicates the existence of a session and conveys sufficient
information to enable participation in the session. SDP is described in RFC 2327.
Session Initiation Protocol (SIP)
A protocol for transporting call setup, routing, authentication, and other feature messages to endpoints within
the IP domain, whether those messages originate from outside the IP cloud over SCN resources or within the
cloud.
A standard of network management that uses a common software agent to manage local and wide area
network equipment from different vendors; part of the Transmission Control Protocol / Internet Protocol (TCP/
IP) suite and defined in RFC 1157.
Simple Network Time Protocol (SNTP)
SNTP, which is an adaptation of the Network Time Protocol (NTP), is widely used to synchronize computer
clocks in the global Internet. It provides comprehensive mechanisms to access national time and frequency
dissemination services, organize the time-synchronization subnet and adjust the local clock in each
participating subnet peer. In most places of the Internet of today, NTP provides accuracies of 1-50 ms,
depending on the characteristics of the synchronization source and network paths.
Subnet
An efficient means of splitting packets into two fields to separate packets for local destinations from packets
for remote destinations in TCP/IP networks.
Switched Circuit Network (SCN)
A communication network, such as the public switched telephone network (PSTN), in which any user may be
connected to any other user through the use of message, circuit, or packet switching and control devices.
T.38
An ITU-T Recommendation for Real-time fax over IP. T.38 addresses IP fax transmissions for IP-enabled fax
devices and fax gateways, defining the translation of T.30 fax signals and Internet Fax Protocols (IFP) packets.
Telephony
The science of translating sound into electrical signals, transmitting them, and then converting them back into
sound.
Transmission Control Protocol/Internet Protocol (TCP/IP)
The basic communication language or protocol of the Internet. It can also be used as a communications
protocol in a private network (either an intranet or an extranet).
Trivial File Transfer Protocol (TFTP)
A simplified version of FTP that transfers files but does not provide password protection, directory capability,
or allow transmission of multiple files with one command.
User Datagram Protocol (UDP)
An efficient but unreliable, connectionless protocol that is layered over IP, as is TCP. Application programs are
needed to supplement the protocol to provide error processing and retransmission of data. UDP is an OSI
layer 4 protocol.
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Virtual LAN (VLAN)

A network of computers that behave as if they are connected to the same wire even though they may actually
be physically located on different segments of a LAN. One of the biggest advantages of VLANs is that when
a computer is physically moved to another location, it can stay on the same VLAN without any hardware
reconfiguration.
Virtual Private Network (VPN)
A private communications network usually used within a company, or by several different companies or
organizations, to communicate over a public network. VPN message traffic is carried on public networking
infrastructure (e.g. the Internet) using standard (often insecure) protocols, or over a service provider's network
providing VPN service guarded by well defined Service Level Agreement (SLA) between the VPN customer
and the VPN service provider.
Voice Over IP (VoIP)
The technology used to transmit voice conversations over a data network using the Internet Protocol. Such
data network may be the Internet or a corporate Intranet.
Wide Area Network (WAN)
A large (geographically dispersed) network, usually constructed with serial lines, that covers a large
geographic area. A WAN connects LANs using transmission lines provided by a common carrier.
741
742
P P E N D I X
List of Acronyms
A
AC
ACS
AES
ANI
AOC-E
Access Concentrator
Auto Configuration Server
Advanced Encryption Standard
Automatic Number Identification
Advice of Charge End-of-Call
B
BRI
Basic Rate Interface
C
CA
CAS
CCBS
CCNR
CDR
CHAP
CLI
CLIP
CLIR
CNG
CNIP
COLP
COLR
CS-ACELP
CWMP
Certification Authority
Channel Associated Signalling
Completion of Calls to Busy Subscriber
Completion of Calls on No Reply
Call Detail Record
Challenge Handshake Authentication Protocol
Command Line Interface
Calling Line Information Presentation
Calling Line Information Restriction
Comfort Noise Generator
Calling Name Identity Presentation
Connected Line Identification Presentation
Connected Line Identification Restriction
Conjugate Structure-Algebraic Code Excited Linear Prediction
CPE WAN Management Protocol
D
DER
DNIS
DNS
DSCP
DSS1
DST
DTMF
Distinguished Encoding Rules

Dialed Number Identification Service
Domain Name Server
Differentiated Services Code Point
Daylight Saving Time
Dual Tone Multi-Frequency
F
FQDN
FSK
Fully Qualified Domain Name

Frequency Shift Keying
G
GMT
Greenwich Mean Time
H
HTML
HTTP
HTTPS
Hz
Hyper Text Markup Language

HyperText Transfer Protocol
HTTP over the Transport Layer Security
Hertz
I
ICMP
IEEE
IETF
ISDN
ITC
ITU
Internet Control Message Protocol

Institute of Electrical & Electronics Engineers
Internet Engineering Task Force
Integrated Services Digital Network
Information Transfer Capability
International Telecommunication Union
743
Appendix F - List of Acronyms
K
kbps
KiloBits Per Second
L
LAN
LED
LLDP-MED
LLPD
Local Area Network

Light Emitting Diode
Link Layer Discovery Protocol-Media Endpoint Discovery
Link Layer Discovery Protocol
M
MAC
MFC
MIB
MIKEY
MSN
MTU
MWI
Media Access Control

Multi-Frequency Code
Management Information Base
Multimedia Internet KEYing
Multiple Subscriber Number
Maximum Transmission Unit
N
NAPTR
NAT
NBNS
NPI
NT
NTP
Naming Authority Pointer request

Network Address Translation
NetBIOS Name Server
Numbering Plan Indicator
Network Termination
Network Time Protocol
O
OCSP
Online Certificate Status Protocol
P
PAP
PBX
PEM
PI
PPP
PPPoE
PRACk
PRI
PSTN
Password Authentication Protocol.

Private Branch eXchange
Privacy Enhanced Mail
Presentation Indicator
Point-to-Point Protocol
Point-to-Point Protocol over Ethernet
Provisional Response Acknowledgement
Primary Rate Interface
Public Switched Telephone Network
Q
QoS
Quality of Service
R
RADIUS
RFC
RTCP
Remote Authentication Dial-In User Service

Request For Comment
Real Time Control Protocol
S
SCN
SDES
SDP
SHA
SI
SIP
SNTP
SRTCP
SRTP
SSH
SSL
STD
STP
Switched Circuit Network

Secure Description
Session Description Protocol
Secure Hash Algorithm
Screening Indicator
Session Initiation Protocol
Simple Network Time Protocol
Secure Real-Time Transport Control Protocol
Secure Real-Time Transport Protocol
Secure Socket Shell
Secure Sockets Layer
Standard Saving Time
Spanning Tree Protocol
T
TBRL
TCP
744
Terminal Balance Return Loss

Transmission Control Protocol
TCP/IP
TE
TEI
TFTP
TLS
TON
Transmission Control Protocol/Internet Protocol

Terminal Equipment
Terminal Endpoint Identifier
Transport Layer Security
Type of Number
U
UDP
UMN
UNI
URI
UTC
User Datagram Protocol

Unit Manager Network
User-Network Interface
Uniform Resource Identifier
Universal Time Coordinated
V
VAD
VLAN
Voice Activity Detector

Virtual Local Area Network
W
WAN
WINS
Wide Area Network

Windows Internet Name Service
745
Appendix F - List of Acronyms
746
P P E N D I X
List of Standards Supported

B
bootp-dhcp-option-88 ................................................................................................................................................... 91, 92
D
draft draft-poetzl-bliss-call-completion-00 ......................................................................................................................... 414
draft-choudhuri-sip-info-digit-00 ................................................................................................................................ 354, 381
draft-ietf-http-authentication-03 ........................................................................................................................ 519, 535, 543
draft-ietf-sipping-realtimefax-00 ........................................................................................................................................ 309
E
ETS 300 207 - Call Diversion and Call Rerouting ............................................................................................................ 209
ETSI 300 659-1 January 2001 (Annex B) ......................................................................................................................... 160
ETSI EN 300659-3 ........................................................................................................................................................... 164
I
IEEE 802.1Q -Virtual Bridged Local Area Networks ........................................................................................................... 98
IEEE 802.1X - Port Based Network Access Control ......................................................................................................... 106
IEEE Std 1003.1-2001 - IEEE Standard for Information Technology---Portable Operating System Interface (POSIX) 457
ITU-T Recommendation E.164 - The international public telecommunication numbering plan ........................................ 455
ITU-T Recommendation F.69 - List of Telex Destination Codes ...................................................................................... 455
ITU-T Recommendation G.703 - Physical/Electrical Characteristics of Hierarchical Digital Interfaces ............ 182, 193, 219
ITU-T Recommendation G.704 - Synchronous Frame Structures Used at Primary and Secondary Hierarchy Levels ... 182,
193, ............................................................................................................................................................. 219
ITU-T Recommendation G.711 ........................................................................................................................................ 349
ITU-T Recommendation G.723.1 ..................................................................................................................................... 350
ITU-T Recommendation I.430 - Basic user-network interface - Layer 1 specification ...................................................... 193
ITU-T Recommendation I.431 - Primary rate user-network interface - Layer 1 specification ........................................... 182
ITU-T Recommendation Q.24 - Multifrequency push-button signal reception .................................................................. 354
ITU-T Recommendation Q.310-Q.332 - Specifications of Signalling System R1 ............................................................. 251
ITU-T Recommendation Q.421 - Digital line signalling code ............................................................................................ 219
ITU-T Recommendation Q.441 - Signalling code ............................................................................................................. 219
ITU-T Recommendation Q.921 - ISDN user-network interface - Data link layer specification ................................. 182, 193
ITU-T Recommendation Q.931 - ISDN user-network interface layer 3 specification for basic call control ....... 182, 193, 458
ITU-T Recommendation T.38 ................................................................................................................................... 309, 353
ITU-T Recommendation X.121 - International numbering plan for public data networks ................................................. 455
R
RFC 1034 - Domain Names - Concepts and Facilities ....................................................................................................... 90
RFC 1035 - Domain Names - Implementation and Specification ....................................................................................... 90
RFC 1157 - Simple Network Management Protocol (SNMP) ........................................................................................... 559
RFC 1332 - The PPP Internet Protocol Control Protocol (IPCP) ...................................................................................... 103
RFC 1334 - PPP Authentication Protocols ....................................................................................................................... 103
RFC 1350 - The TFTP Protocol (Revision 2) ........................................................................................... 441, 519, 535, 543
RFC 1661 - The Point-to-Point Protocol (PPP) ................................................................................................................ 103
RFC 1877 - PPP Internet Protocol Control Protocol Extensions for Name Server Addresses ......................................... 103
RFC 1886 - DNS Extensions to support IP version 6 ......................................................................................................... 90
RFC 1890 - RTP Profile for Audio and Video Conferences with Minimal Control ............................................................ 354
RFC 1910 - User-based Security Model for SNMPv2 ...................................................................................................... 559
RFC 1945 - Hypertext Transfer Protocol - HTTP/1.0 ......................................................................................................... 41
RFC 1994 - Challenge Handshake Authentication Protocol (CHAP) ............................................................................... 103
RFC 2030 - Simple Network Time Protocol (SNTP) Version 4 for IPv4, IPv6 and OSI ...................................................... 91
RFC 2104 - HMAC Keyed-Hashing for Message Authentication ..................................................................................... 559
747
Appendix G - List of Standards Supported
RFC 2131 - Dynamic Host Configuration Protocol ................................................................................................... 107, 147

RFC 2132 - DHCP Options and BOOTP Vendor Extensions ................................................................................... 107, 147
RFC 2181 - Clarifications to the DNS Specification ........................................................................................................... 90
RFC 2246 - The TLS Protocol Version 1.0 ............................................................................................... 303, 520, 536, 545
RFC 2459 - X.509 Digital Certificates ............................................................................................................... 520, 536, 545
RFC 2460 - IPv6 ................................................................................................................................................................. 98
RFC 2464 - Transmission of IPv6 Packets over Ethernet Networks .................................................................................. 98
RFC 2475 - An Architecture for Differentiated Services ................................................................................................... 113
RFC 2516 - A Method for Transmitting PPP Over Ethernet (PPPoE) .............................................................................. 103
RFC 2543 - SIP, Session Initiation Protocol ............................................................................................. 279, 285, 287, 319
RFC 2576 - Coexistence between Version 1, Version 2, and Version 3 of the Internet-standard Network Management
Framework .................................................................................................................................................. 559
RFC 2616 - Hypertext Transfer Protocol - HTTP/1.1 ......................................................................... 41, 441, 519, 535, 543
RFC 2617 - HTTP Authentication - Basic and Digest Access Authentication .......................................... 519, 535, 543, 583
RFC 2705 - Media Gateway Control Protocol (MGCP) Version 1.0 ................................................................................. 395
RFC 2741 - Agent Extensibility (AgentX) Protocol Version 1 ........................................................................................... 559
RFC 2818 - HTTP Over TLS ............................................................................................................................ 520, 536, 545
RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals ................................................... 354
RFC 2865 - Remote Authentication Dial In User Service (RADIUS) ................................................................................ 583
RFC 2866 - RADIUS Accounting ...................................................................................................................................... 583
RFC 2976 - The SIP INFO Method .................................................................................................................................. 411
RFC 3164 - The BSD Syslog Protocol ............................................................................................................................... 71
RFC 3261 - SIP, Session Initiation Protocol ..................................................................... 279, 285, 287, 299, 303, 310, 318
RFC 3262 - Reliability of Provisional Responses in the Session Initiation Protocol (SIP) ................................................ 340
RFC 3263 - Session Initiation Protocol (SIP) - Locating SIP Servers .............................................................................. 324
RFC 3264 - An Offer/Answer Model with the Session Description Protocol (SDP) .................. 309, 312, 318, 319, 321, 322
RFC 3265 - Session Initiation Protocol (SIP)-Specific Event Notification ......................................................................... 432
RFC 3268 - Advanced Encryption Standard (AES) Ciphersuites for Transport Layer Security (TLS) ............. 520, 536, 545
RFC 3280 - Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile ..... 520, 536,
545, ............................................................................................................................................................. 553
RFC 3311 - The Session Initiation Protocol (SIP) UPDATE Method ................................................................................ 340
RFC 3315 - DHCPv6 ........................................................................................................................................................ 107
RFC 3323 - A Privacy Mechanism for the Session Initiation Protocol (SIP) ..................................................................... 488
RFC 3325 - Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks .. 488
RFC 3398 - Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping 332
RFC 3411 - An Architecture for Describing Simple Network Management Protocol (SNMP) Management Frameworks 559
RFC 3412 - Message Processing and Dispatching for the Simple Network Management Protocol (SNMP) .................. 559
RFC 3413 - Simple Network Management Protocol (SNMP) Applications ...................................................................... 559
RFC 3414 - User-based Security Model (USM) for version 3 of the Simple Network Management Protocol (SNMPv3) 559
RFC 3415 - View-based Access Control Model (VACM) for the Simple Network Management Protocol (SNMP) .......... 559
RFC 3416 - Version 2 of the Protocol Operations for the Simple Network Management Protocol (SNMP) ..................... 559
RFC 3417 - Transport Mappings for the Simple Network Management Protocol (SNMP) ............................................... 559
RFC 3442 - The Classless Static Route Option for Dynamic Host Configuration Protocol (DHCP) version 4 ................. 129
RFC 3515 - The Session Initiation Protocol (SIP) Refer Method ..................................................................................... 309
RFC 3550 - RTP - A Transport Protocol for Real-Time Applications ............................................................................... 349
RFC 3551 - RTP Profile for Audio and Video Conferences with Minimal Control ............................................................ 349
RFC 3617 - Uniform Resource Identifier (URI) Scheme for the Trivial File Transfer Protocol ......................... 519, 535, 543
RFC 3711 - The Secure Real-time Transport Protocol (SRTP) ....................................................................................... 373
RFC 3826 - The Advanced Encryption Standard (AES) Cipher Algorithm in the SNMP User-based Security Model ..... 559
RFC 3830 - MIKEY - Multimedia Internet KEYing ............................................................................................................ 373
RFC 3842 - The Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)
..................................................................................................................................................................... 432
RFC 3863 - Presence Information Data Format (PIDF) ................................................................................................... 287
RFC 3903 - Session Initiation Protocol (SIP) Extension for Event State Publication ............................................... 279, 287
RFC 4040 - RTP Payload Format for a 64 kbit/s Transparent Call .......................................................................... 352, 353
RFC 4193 - Unique_local_address .................................................................................................................................... 98
RFC 4235 - An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP) .................................... 414
RFC 4251 - The Secure Shell (SSH) Protocol Architecture ............................................................................................... 22
RFC 4291 - IP Version 6 Addressing Architecture ............................................................................................................. 98
RFC 4443 - ICMPv6 ........................................................................................................................................................... 98
RFC 4567 - Key Management Extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol
(RTSP) ........................................................................................................................................................ 373
RFC 4568 - SDES - Security Descriptions for Media Streams ......................................................................................... 373
RFC 4579 - Session Initiation Protocol (SIP) - Call Control - Conferencing for User Agents ........................................... 425
RFC 4861 - IPv6_neighbor_discovery ................................................................................................................................ 98
RFC 4862 - IPv6_stateless_autoconf ................................................................................................................................. 98
748
RFC 5806 - Diversion Indication in SIP ............................................................................................................................ 462

RFC 854 - Telnet Protocol Specification ............................................................................................................................ 21
RFC 959 - File Transfer Protocol ...................................................................................................................... 519, 535, 543
T
TR-069 - CPE WAN Management Protocol v1.1 (Issue 1 Amendment 2) ....................................................................... 565
TR-098 - Internet Gateway Device Data Model for TR-069 (Issue 1 Amendment 2) ....................................................... 565
TR-104 - Provisioning Parameters for VoIP CPE ............................................................................................................. 565
TR-106 - Data Model Template for TR-069-Enabled Devices .......................................................................................... 565
TR-111 - Applying TR-069 to Remote Management of Home Networking Devices ......................................................... 565
B
bootp-dhcp-option-8895, 96
D
draft draft-poetzl-bliss-call-completion-00424
draft-choudhuri-sip-info-digit-00362, 391
draft-ietf-http-authentication-03531, 549, 559
draft-ietf-sipping-realtimefax-00315
E
ETS 300 207 - Call Diversion and Call Rerouting213
ETSI 300 659-1 January 2001 (Annex B)164
ETSI EN 300659-3168
I
IEEE 802.1Q -Virtual Bridged Local Area Networks102
IEEE 802.1X - Port Based Network Access Control110
IEEE Std 1003.1-2001 - IEEE Standard for Information Technology---Portable Operating
System Interface (POSIX)469
ITU-T Recommendation E.164 - The international public telecommunication numbering
plan467
ITU-T Recommendation F.69 - List of Telex Destination Codes467
ITU-T Recommendation G.703 - Physical/Electrical Characteristics of Hierarchical Digital Interfaces186, 197, 223
ITU-T Recommendation G.704 - Synchronous Frame Structures Used at Primary and Secondary Hierarchy Levels186, 197, 223
ITU-T Recommendation G.723.1358
ITU-T Recommendation I.430 - Basic user-network interface - Layer 1 specification197
ITU-T Recommendation I.431 - Primary rate user-network interface - Layer 1 specification186
ITU-T Recommendation Q.24 - Multifrequency push-button signal reception362
ITU-T Recommendation Q.310-Q.332 - Specifications of Signalling System R1255
ITU-T Recommendation Q.421 - Digital line signalling code223
ITU-T Recommendation Q.441 - Signalling code223
ITU-T Recommendation Q.921 - ISDN user-network interface - Data link layer specificaDgw v2.0 Application
749
tion186, 197
ITU-T Recommendation Q.931 - ISDN user-network interface layer 3 specification for basic call control186, 197, 470
ITU-T Recommendation T.38315, 361
ITU-T Recommendation X.121 - International numbering plan for public data networks467
R
RFC 1034 - Domain Names - Concepts and Facilities94
RFC 1035 - Domain Names - Implementation and Specification94
RFC 1157 - Simple Network Management Protocol (SNMP)575
RFC 1332 - The PPP Internet Protocol Control Protocol (IPCP)107
RFC 1334 - PPP Authentication Protocols107
RFC 1350 - The TFTP Protocol (Revision 2)451, 531, 549, 559
RFC 1661 - The Point-to-Point Protocol (PPP)107
RFC 1877 - PPP Internet Protocol Control Protocol Extensions for Name Server Addresses
107
RFC 1886 - DNS Extensions to support IP version 694

RFC 1890 - RTP Profile for Audio and Video Conferences with Minimal Control362
RFC 1910 - User-based Security Model for SNMPv2575
RFC 1945 - Hypertext Transfer Protocol - HTTP/1.041
RFC 1994 - Challenge Handshake Authentication Protocol (CHAP)107
RFC 2030 - Simple Network Time Protocol (SNTP) Version 4 for IPv4, IPv6 and OSI95
RFC 2104 - HMAC Keyed-Hashing for Message Authentication575
RFC 2131 - Dynamic Host Configuration Protocol112, 151
RFC 2132 - DHCP Options and BOOTP Vendor Extensions112, 151
RFC 2181 - Clarifications to the DNS Specification94
RFC 2246 - The TLS Protocol Version 1.0307, 532, 550, 560
RFC 2459 - X.509 Digital Certificates532, 550, 560
RFC 2460 - IPv6102
RFC 2464 - Transmission of IPv6 Packets over Ethernet Networks102
RFC 2475 - An Architecture for Differentiated Services117
RFC 2516 - A Method for Transmitting PPP Over Ethernet (PPPoE)107
RFC 2543 - SIP, Session Initiation Protocol283, 289, 291, 325
RFC 2576 - Coexistence between Version 1, Version 2, and Version 3 of the Internet-standard Network Management Framework575
RFC 2616 - Hypertext Transfer Protocol - HTTP/1.141, 451, 531, 549, 559
RFC 2617 - HTTP Authentication - Basic and Digest Access Authentication531, 549, 559, 607
RFC 2705 - Media Gateway Control Protocol (MGCP) Version 1.0405
RFC 2741 - Agent Extensibility (AgentX) Protocol Version 1575
RFC 2818 - HTTP Over TLS532, 550, 560
RFC 2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals362
RFC 2865 - Remote Authentication Dial In User Service (RADIUS)607
RFC 2866 - RADIUS Accounting607
RFC 2976 - The SIP INFO Method421
RFC 3164 - The BSD Syslog Protocol73
RFC 3261 - SIP, Session Initiation Protocol283, 289, 291, 303, 307, 316, 324
RFC 3262 - Reliability of Provisional Responses in the Session Initiation Protocol (SIP)346
750
RFC 3263 - Session Initiation Protocol (SIP) - Locating SIP Servers330

RFC 3264 - An Offer/Answer Model with the Session Description Protocol (SDP)315, 318,
325, 327, 328
RFC 3265 - Session Initiation Protocol (SIP)-Specific Event Notification441

RFC 3268 - Advanced Encryption Standard (AES) Ciphersuites for Transport Layer Security (TLS)532, 550, 560
RFC 3280 - Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile532, 550, 560, 569
RFC 3311 - The Session Initiation Protocol (SIP) UPDATE Method346
RFC 3315 - DHCPv6112
RFC 3323 - A Privacy Mechanism for the Session Initiation Protocol (SIP)501
RFC 3325 - Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity
within Trusted Networks501
RFC 3398 - Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping339
RFC 3411 - An Architecture for Describing Simple Network Management Protocol (SNMP) Management Frameworks575
RFC 3412 - Message Processing and Dispatching for the Simple Network Management
Protocol (SNMP)575
RFC 3413 - Simple Network Management Protocol (SNMP) Applications575
RFC 3414 - User-based Security Model (USM) for version 3 of the Simple Network Management Protocol (SNMPv3)575
RFC 3415 - View-based Access Control Model (VACM) for the Simple Network Management Protocol (SNMP)575
RFC 3416 - Version 2 of the Protocol Operations for the Simple Network Management
Protocol (SNMP)575
RFC 3417 - Transport Mappings for the Simple Network Management Protocol (SNMP)
575
RFC 3442 - The Classless Static Route Option for Dynamic Host Configuration Protocol
(DHCP) version 4133
RFC 3515 - The Session Initiation Protocol (SIP) Refer Method315
RFC 3550 - RTP - A Transport Protocol for Real-Time Applications357
RFC 3551 - RTP Profile for Audio and Video Conferences with Minimal Control357
RFC 3617 - Uniform Resource Identifier (URI) Scheme for the Trivial File Transfer Protocol531, 549, 559
RFC 3711 - The Secure Real-time Transport Protocol (SRTP)381
RFC 3826 - The Advanced Encryption Standard (AES) Cipher Algorithm in the SNMP
User-based Security Model575
RFC 3830 - MIKEY - Multimedia Internet KEYing381
RFC 3842 - The Message Summary and Message Waiting Indication Event Package for
the Session Initiation Protocol (SIP)441
RFC 3863 - Presence Information Data Format (PIDF)291
RFC 3903 - Session Initiation Protocol (SIP) Extension for Event State Publication283, 291
RFC 4040 - RTP Payload Format for a 64 kbit/s Transparent Call360, 361
RFC 4193 - Unique_local_address102
RFC 4235 - An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol
(SIP)424
751
RFC 4251 - The Secure Shell (SSH) Protocol Architecture22

RFC 4291 - IP Version 6 Addressing Architecture102
RFC 4443 - ICMPv6102
RFC 4567 - Key Management Extensions for Session Description Protocol (SDP) and Real
Time Streaming Protocol (RTSP)381
RFC 4568 - SDES - Security Descriptions for Media Streams381
RFC 4579 - Session Initiation Protocol (SIP) - Call Control - Conferencing for User Agents
434
RFC 4861 - IPv6_neighbor_discovery102

RFC 4862 - IPv6_stateless_autoconf102
RFC 5806 - Diversion Indication in SIP474
RFC 854 - Telnet Protocol Specification21
RFC 959 - File Transfer Protocol531, 549, 559
T
TR-069 - CPE WAN Management Protocol v1.1 (Issue 1 Amendment 2)581
TR-098 - Internet Gateway Device Data Model for TR-069 (Issue 1 Amendment 2)581
TR-104 - Provisioning Parameters for VoIP CPE581
TR-106 - Data Model Template for TR-069-Enabled Devices581
TR-111 - Applying TR-069 to Remote Management of Home Networking Devices581
752
Index
Numerics
10 BaseT
defined 733
100 BaseT
defined 733
802.1q, in local QoS 119
A
AAA, service
user access 17, 18
access control
RADIUS 608
accounting type 609
authentication type 608
servers configuration 609
type 608
user
access rights 607
add 607
delete 607
modify 607
access rights 607
description 611
RADIUS servers 610
access, CLI keyword 27
accounting type, in RADIUS 609
ACK branch matching, in SIP 332
acronyms 743
activate licence 617
A-Law 357
defined 733
alias, CLI keyword 28
answering on caller ID, on FXO port 172
AOC-E support, ISDN interface 216, 217
audience, intended xxv
authentication information 303
creating 304
deleting 306
editing 304
moving 306
authentication type, in RADIUS 608
auto cancel timeout, ISDN interface 212
auto-configuration
E&M interfaces 257
ISDN interfaces 182
R2 interfaces 225
automatic
call 423
configuration update
number of retries 547
on restart 544
auto-routing
auto-routable parameter 525
enabling 523
introduction 523
SIP gateway 525
B
base ports 401
bearer capabilities mapping, vs. codec 363
preferred codec choice 321
BRI interface configuration

bypass connection 207
calling name max length 202
channel allocation strategy 201
clock mode 199
clock reference 61
connection type 199
copy config to all interfaces 198
encoding scheme, fallback 200
encoding scheme, preferred 200
endpoint type 198
exclusive B-channel selection 202
inband DTMF dialing 201
inband tone generation 201
maximum active calls 201
network location 200
overlap dialing 202
restart on startup 202
select interface 198
sending complete information 202
signal information element 201
signalling protocol 200
bridging, hardware configuration 65
bypass
activation 169
activation DTMF map 170
deactivation timeout 170
Bypass connection 6
bypass connection, in BRI 207
C
call
automatic 423
completion
CCBS 424
CCBS, using 427
CCNR 424
CCNR, using 427
emergency
enabling 411
emergency override 443, 445
forward
on busy 413, 415
on no answer 416, 417
unconditional 418, 419
hold 439
direction attributes 325
putting on hold 431, 440
second 440, 441
transfer
attended 428, 429
blind 428, 429
waiting 430
disabling 431
enabling 431
IMS-3GPP Communication Waiting 431
using 431
Call Detail Record 461
call failure timeout, on FXO port 174
call properties translation, in call routing
built from 509
creating 508
753
Index
defined 508
deleting 510
editing 508
modified property 509
moving 510
name 508
call rejection (drop) causes 514
call router
call properties translation
built from 509
creating 508
defined 508
deleting 510
editing 508
modified property 509
moving 510
name 508
hairpinning 521
hunts
creating 511
defined 511
deleting 518
destinations 512
editing 511
moving 518
selection algorithm 512
timeout 512
introduction 467
limitations 468
mappings
defined 491
mappings, expression
accessing 493
criteria 494
deleting 500
moving 500
sub mappings 499
transformation 497
mappings, type
accessing 491
input call property 492
moving 500
transformation 492
regular expression 469
routes
call property to compare 487
creating 486
defined 485
deleting 491
destination, of call 490
editing 486
expression to compare 488
mapping, apply 489
moving 490
signalling, of call 489
source to compare 487
routing type
Called Bearer Channel 474
Called E164 471
Called Host 472
Called Name 472
Called NPI 471
Called Phone Context 474
Called SIP username 474
Called TON 471
754
Called URI 472

Calling Bearer Channel 474
Calling E164 471
Calling Host 472
Calling ITC 473
Calling Name 472
Calling NPI 471
Calling Phone Context 474
Calling PI 472
Calling SI 472
Calling SIP username 474
Calling TON 471
Calling Uri 472
Date/Time 473
Last Diverting E.164 475
Last Diverting Number Presentation 476
Last Diverting Party Number Type 475
Last Diverting Private Type of Number 475
Last Diverting Public Type of Number 475
Last Diverting Reason 474
Original Diverting E.164 475
Original Diverting Number Presentation 476
Original Diverting Party Number Type 475
Original Diverting Private Type of Number 475
Original Diverting Public Type of Number 475
Original Diverting Reason 474
special tags 476
signalling properties
180 with SDP 502
183 without SDP 503
call properties translation 504
creating 501
defined 501
deleting 505
destination host 502
early connect 502
early disconnect 502
editing 501
moving 505
name 502
privacy level 503
SIP headers translation 504
SIP headers translation
built from 506
creating 505
defined 505
deleting 507
editing 505
modified header 506
moving 507
name 506
SIP redirects
creating 519
defined 519
deleting 520
editing 519
moving 520
status information 484
call waiting
private number for SIP INFO 324
Called Bearer Channel, routing type 474
Called E164, routing type 471
Called Host, routing type 472
Called Name, routing type 472
Called NPI, routing type 471
Called Phone Context, routing type 474
Called SIP username, routing type 474

Called TON, routing type 471
Called URI, routing type 472
caller ID
customization, in POTS 162
DTMF signalling 164
FSK generation 164
generation 164
Calling Bearer Chanel, routing type 474
Calling E164, routing type 471
Calling Host, routing type 472
Calling ITC, routing type 473
calling name max length, ISDN interface
BRI configuration 202
PRI configuration 192
Calling Name, routing type 472
Calling NPI, routing type 471
Calling Phone Context, routing type 474
Calling PI, routing type 472
Calling SI, routing type 472
Calling SIP username, routing type 474
Calling TON, routing type 471
Calling Uri, routing type 472
CAS. see R2 configuration
CCBS
setting up 424
using 427
CCNR
setting up 424
using 427
cd, CLI keyword 29
CDR (Call Detail Record) 461
CED tone detection 400
behaviour 400
certificate
validation, in TLS 322
certificates
deleting 571
transferring 573
upload through web browser 572
validation, in firmware pack update 566
channel allocation strategy
E&M interface, configuration 260
ISDN interface
R2 interface, configuration 228
channel range
ISDN interface
Clear Channel
defined 360
enabling
for data transmission 372
for voice transmission 372
packetization time 372
payload type 372
priority
for data codecs 372
for voice codecs 372
clear channel fax
call flow 375
limitation 376
Clear Mode
defined 360
enabling
payload type 371
priority
for data codecs 371
CLI
accessing via Console port 20
accessing via SSH 22
accessing via Telnet 21
command completion 24
contexts 23
current unit status 25
defined 19
help 25
history 24
keywords
access 27
alias 28
cd 29
columnars 29
commands 29
defval 30
dump 30
get 31
help 32
indexes 33
keys 33
logs off 34
logs on 34
ls 34
name 35
objects 35
pcapture 36
ping 36, 37
scalars 37
set 37
show 38
sysinfo 39
tables 39
type 39
unalias 28
macros 24
service restart 24
syslog messages 25
welcome message 25
CLIP, ISDN interface 214
CLIR, ISDN interface 214
override 215
clock mode
ISDN interface
clock reference
clock reference, in ISDN
CNG tone detection 400, 459
codec
bearer capabilities mapping 363
preferred codec choice 321
755
Index
Clear Channel
defined 360
enabling
payload type 372
priority
for data codecs 372
Clear Mode
defined 360
enabling
payload type 371
priority
for data codecs 371
G.711
defined 357
enabling
priority
G.723
bit rate 366
defined 358
enabling for voice transmission 366
priority for voice codecs 366
G.726
defined 358
enabling
payload type 369
priority
G.729
defined 359
voice activity detection 370
T.38
defined 361
enabling 377
number of redundancy packets 377
priority 377
X-CCD Clear Channel
defined 361
enabling
payload type 374
priority
for data codecs 374
756
COLP, ISDN interface 215

COLR, ISDN interface 215
override 216
columnars, CLI keyword 29
commands, CLI keyword 29
conference call
limitation 433
configuration
executing script
from a PC 543
from server 539
exporting to script 536
to a PC 537
to a URL 536
configuration backup image
backing up 549
decryption 549
location path 549
privacy key 549
remote server
HTTP server, configuring 550
HTTPS server, configuring 550
SNTP server, configuring 549
TFTP server, configuring 549
restoring 549
server, IP address 549
transfer protocols 549
configuration script
executing configuration
from a PC 543
from server 539
exporting configuration 536
to a PC 537
to a URL 536
configuration script download
automatic update
number of retries 547
on restart 544
decryption 539
DHCP auto-provisioning
DHCP
auto-provisioning (scripts) 546
downloading 544
location path 539
privacy key 539
remote server
script example 548
server, IP address 539
connection type, ISDN interface
Console port
accessing CLI via 20
contact domain, in SIP 293
Content-type, unsupported 335
Country 168
country selection, R2 interface 229
country-specific parameters
CNG tone detection 459
country 455
custom tone configuration 447
dialing settings 457
list of countries 621

user gain
input 456
output 456
custom tones, configuring 447
CWMP
ACS parameters 584
Allow Unauthenticated UDP Connection Requests 594
general configuration 583
parameter type validation 595, 596
periodic inform parameters 586
profile creation 589
TR-069 configuration 588
TR-111 590
D
Date/Time, routing type 473
decryption
configuration backup image 549
configuration script download 539
default gateway
client, setting 93
server, in embedded DHCP server 154
default router. see default gateway
defval, CLI keyword 30
delayed hot line 437
detection custom repetition, on FXO port 176
DHCP connection type 102
DHCP server
configuring 112
embedded
default gateway 154
defined 151
DNS servers 155
domain name server 154
lease time 153
leases 157
NBNS servers 157
NTP servers 156
subnets, managing 153
DHCPv4 classless static route option 133
diagnostic traces, enabling 73
PCM capture 76
dial tone detection mode, on FXO port 171
dial tone detection timeout, on FXO port 172
dialing
settings
delay between MFR1s 458
DTMF duration value 458
inter-digit dial delay 458
Differentiated Services (DS) Field, in local QoS 117
direct IP address
call 438
direction attribute "sendonly" 321, 327
direction attributes, in SIP 325
disabling endpoints 69
distinctive ring 623
DNS
client 94
supported queries 351
DNS SRV
record lock, in SIP 330
document
conventions xxix
intended audience xxv
objectives xxiv
structure xxv
domain name
download, firmware. see firmware pack update
DTMF
base ports 401
defined 735
detection 396
duration value 458
machine detection
CED tone detection, behaviour 400
V.21 modulation detection 400
out-of-band 395
signalling, caller ID 164
transport type
over the SIP protocol 396
DTMF map
allowed 409
defined
# and * characters 407
combining two expressions 406
special characters 406
timer 407
using 406
validating 407
definition 405
general parameters 408
refused 411
dump, CLI keyword 30
E
E&M
partial reset 258
E&M configuration
auto-configuration 257
channel range 260
clock mode 259
digit timers 267
encoding scheme 261
introduction 255
IP synchronization 272
enabling 272
master units 272
resetting synchronization data 273
slave units 273
line coding 260
line framing 260
link timers 269
preset 257
PRI statistics 274
selecting signalling protocol 256
signalling 262
B bit setting 263
C bit setting 263
D bit setting 263
override default settings 263
757
Index
timers 265
tones 270
emergency call
enabling 411
enabling endpoints 69
encoding scheme
encoding scheme, ISDN interface
fallback
preferred
encoding scheme, preferred
endpoint
administration 70
locking/unlocking 69
unit shutting down behsviour 71
unregistered, behaviour when 298
endpoint services
automatic call 423
call completion
CCBS 424
CCNR 424
call forward
on busy 413
on no answer 416
unconditional 418
call transfer
attended transfer 428
blind transfer 428
call waiting 430
conference call 433
limitation 433
direct IP address call 438
emergency call override 443, 445
hold 439
hook-flash processing 422
second call 440
endpoint type, ISDN interface
error causes mapping, setting 339
cause to SIP 342
SIP to cause 340
Ethernet connection
setting speed of 113
Ethernet, link configuration 110
event handling, in SIP 351
events
monitoring parameters
monitoring parameters, in events 81
notification
notification events 79
rule
deleting 81
exclusive B-channel selection, ISDN interface
758
F
facility ISDN services
enabling 213
factory reset
reverting to 16
see also partial reset
fax
call waiting tone, disabling 431
calling tone detection, enabling 459
clear channel
call flow 375
limitation 376
T.38
defined 361
enabling 377
priority 377
user gain vs communication quality 456
file manager 613
firewall, local
default policy 123
disabling 127
partial reset 123
rule, creating 124
rule, deleting 127
rule, editing 124
rule, moving 126
firewall, network
default policy 137
disabling 141
rule, creating 138
rule, deleting 141
rule, editing 138
rule, moving 141
firmware pack update
certificate validation 566
downgrade 568
firmware pack, defined 559
launching 567
remote server
rollback 564
Spanning Tree Protocol 568
update files path 565
update files server
IP address 565
port number 565
zip file, extracting 567
flash hook detection range 168
force end of call on call failure, on FXO port 173
forced end of call on silence detection mode, on FXO port 174
forced end of call on tone detection mode, on FXO port 174
Foreign Exchange Office (FXO)
defined 735
see also endpoint
Foreign Exchange Service/Station (FXS)
defined 735
see also endpoint
forked provisional reponses behaviour, in SIP 347
FSK generation, caller ID 164

FXO configuration
answering on caller ID 172
call failure timeout 174
detection custom repetition 176
dial tone detection mode 171
dial tone detection timeout 172
force end of call on call failure 173
forced end of call on silence detection mode 174
forced end of call on tone detection mode 174
link state verification 173
link state verification timeout 173
not allowed behavior 173
pre dial delay 171
silence detection timeout 174
tone detection custom cadence 175
tone detection custom frequency 175
wait before answering delay 172
wait for callee to answer 172
FXS
auto cancel timeout 165
bypass
activation 169
country override
flash hook detection range 168
loop current 168
disconnect delay 165
inband ringback 166
line supervision mode 165
override country customization 167
power drop on disconnect duration 166
service activation 166
shutdown behavior 166
G
G.711
defined 357
enabling
priority
G.723
bit rate 366
defined 358
G.726
defined 358
enabling
payload type 369
priority
G.729
defined 359

voice activity detection 370
gateways, multiple SIP
adding 279
keep alive destinations 289
messaging servers 286
proxy servers 287
registrar servers 285
removing 279
get, CLI keyword 31
group port management 69
H
hairpinning 521
hardware configuration
BRI interface
clock reference 61
power feeding over ISDN 64
bridging 65
PRI interface
clock reference 63
line type 63, 224, 256
signaling 63, 224, 256
header, SIP user agent
sending 301
help, CLI keyword 32
hold, putting a call on 431, 439, 440
home domain proxy override 349
hook-flash processing 422
HTTP
remote server, configuring 451, 532, 550, 560
HTTP User-Agent Header
format 43
HTTPS
remote server, configuring 532, 550, 560
hunts, in call routing
creating 511
defined 511
deleting 518
destinations 512
editing 511
moving 518
selection algorithm 512
timeout 512
I
IEEE 802.1q, in local QoS 119
IMS-3GPP Communication Waiting 431
inband DTMF dialing, ISDN interface
inband tone generation, ISDN interface
indexes, CLI keyword 33
InformationFollowing operation, ISDN interface
intended audience xxv
inter-digit dial delay 458
IP address
configuration backup server 549
759
Index
configuration script download server 539

firmware pack update files server 565
LAN, setting for 102
messaging server host 285
reserving in network server 101
SIP outbound proxy 285
SIP proxy 285
SIP registrar 285
syslog daemon 73
unit, setting for 102
vocal identification of 163
IP address call 438
IP routing
DHCPv4 classless static route option 133
IPv4 forwarding 129
rule, creating 130
rule, deleting 131
rule, editing 130
rule, moving 131
static IPv4 routes 132
status information 131, 132
IP synchronization, in E&M 272
enabling 272
master units 272
slave units 273
IP synchronization, in ISDN 218
enabling 219
master units 219
slave units 220
IP synchronization, in R2 248
enabling 248
master units 248
slave units 249
IPv4
forwarding 129
IPv4, static routes 132
ISDN
partial reset 183, 613
ISDN configuration
auto calcel timeout 212
BRI interface
bypass connection 207
clock mode 199
connection type 199
copy config to all interfaces 198
endpoint type 198
overlap dialing 202
interop parameters
760
copy config to all interfaces 208, 212, 213

maximum facility waiting delay 209
Progress Indicator in Alerting 209
Progress Indicator in Call Proceeding 208
Progress Indicator in Connect 209
Progress Indicator in Progress 209
Progress Indicator in Setup 208
Progress Indicator in Setup ACK 208
introduction 179
enabling 219
master units 219
slave units 220
Layer 1 Timer 3 212
preset 183
PRI interface
channel range 190
clock mode 188
copy config to all interfaces 187, 526
endpoint type 187
InformationFollowing operation 196
line coding 189
line framing 189
line type 182
link establishment 192, 202
overlap dialing 192
statistics 183
services
AOC-E support 216, 217
CLIP 214
CLIR 214
CLIR override 215
COLP 215
COLR 215
COLR override 216
facility services, enabling 213
Maintenance Service Call Termination 216
MSN 218
supplementary services, enabling 213
J
jitter buffer protection 391
K
keep alive, in SIP 288
keys, CLI keyword 33
L
Lan1, network interface 102
Last Diverting E.164, routing type 475
Last Diverting Number Presentation, routing type 476
Last Diverting Party Number Type, routing type 475
Last Diverting Private Type of Number, routing type 475
Last Diverting Public Type of Number, routing type 475
Last Diverting Reason, routing type 474
Layer 1 Timer 3, ISDN interface 212
lease times in embedded DHCP server 153
leases in embedded DHCP server 157
LEDs behaviour
in firmware pack update mode 568
licence, activate 617
limitations, DSP 6
line coding
ISDN interface
line framing
ISDN interface
line signalling protocol
line type
PRI configuration 63, 182, 224, 256
link establishment, ISDN interface
PRI configuration 192, 202
link state verification timeout, on FXO port 173
link state verification, on FXO port 173
LLDP configuration, network interface 109
local ring behaviour, in SIP 328
locking endpoints 69
logs off, CLI keyword 34
logs on, CLI keyword 34
loop current 168
ls, CLI keyword 34
M
MAC address 101
defined 737
vocal identification of 163
machine detection
CED tone detection, behaviour 400
Maintenance Service Call Termination, ISDN interface 216
management interface
partial reset 617
selecting 617
mappings, in call routing
defined 491
expression
accessing 493
criteria 494
deleting 500
moving 500
sub mappings 499
transformation 497
type
accessing 491
input call property 492
moving 500
transformation 492
Max-Forwards header, in SIP 324
maximum active calls
ISDN interface
maximum facility waiting delay, ISDN interface 209
message waiting indicator 441
visual type 443
messaging subscription, in SIP 353
MFR1
dialling delay between 458
MSN, ISDN interface 218
MTU
configuring 110
MTU, requirements 661
Mu ()-Law 357
defined 737
music on hold
configuring 451
remote server
N
name, CLI keyword 35
NAT
disabling 150
in configuration backup image 554
in configuration script download 544
partial reset 143
NAT, destination
rule, creating 147
rule, deleting 150
rule, editing 147
rule, moving 150
NAT, source
rule, creating 143
rule, deleting 150
rule, editing 143
rule, moving 150
NBNS server, in embedded DHCP server 157
network interfaces
connection type 102
defining 102
DHCP client identifier presentation
DHCP
client identifier presentation 108
Ethernet link configuration 110
Lan1 102
LLDP configuration 109
partial reset 101
PPP configuration 107
PPP negotiation 108
priority 106
Rescue 102
Uplink 102
network location, ISDN interface
761
Index
network parameters
automatic configuration interface 91
default gateway client 93
DNS client 94
host 91
SNTP client 95
syslocation, updating 100
sysname, updating 100
time zone, defining custom 96
network traffic control
in QoS 120
not allowed behavior, on FXO port 173
NTP server, in embedded DHCP server 156
O
objectives of document xxiv
objects, CLI keyword 35
Offer/Answer model 316, 319, 323
Original Diverting E.164, routing type 475
Original Diverting Number Presentation, routing type 476
Original Diverting Party Number Type, routing type 475
Original Diverting Private Type of Number, routing type 475
Original Diverting Public Type of Number, routing type 475
Original Diverting Reason, routing type 474
outbound proxy server, configuring 284
out-of-band DTMF 395
overlap dialing, ISDN interface
P
packetization time
Clear Channel 372
Clear Mode 371
G.711 365
G.723 367
G.726 369
G.729 370
X-CCD Clear Channel 374
disabling 15
E&M preset 258
ISDN preset 183, 613
local firewall service 123
management interface 617
NAT service 143
network interfaces 101
R2 preset 226
RADIUS authentication 609
SNMP 575, 580
user access 18
user password 608
see also factory reset
password
modify 613, 617
payload type
Clear Channel 372
Clear Mode 371
G.726 369
using the one found in answer 398
X-CCD Clear Channel 374
pcapture, CLI keyword 36
PCM capture 76
penalty box, in SIP 337
762
ping, CLI keyword 36, 37

port number
configuration backup server 549
configuration script download server 539
firmware pack update files server 565
messaging serverr host 285
SIP outbound proxy 285
SIP proxy 285
SIP registrar 285
POTS
auto cancal timeout 165
caller ID customization 162
FXO configuration
answering on caller ID 172
call failure timeout 174
detection custom repetition 176
dial tone detection mode 171
dial tone detection timeout 172
force end of call on call failure 173
forced end of call on silence detection mode 174
forced end of call on tone detection mode 174
link state verification 173
link state verification timeout 173
not allowed behavior 173
pre dial delay 171
silence detection timeout 174
tone detection custom cadence 175
tone detection custom frequency 175
wait before answering delay 172
wait foe callee to answer 172
FXS bypass
activation 169
FXS Country Override Flash Hook Detection Range 168
FXS Country Override Loop Current 168
FXS Disconnect Delay 165
FXS Inband Ringback 166
FXS Line Supervision Mode 165
FXS Override Country Customization 167
FXS power drop on disconnect duration 166
FXS service activation 166
FXS shutdown behavior 166
power feeding over ISDN 64
PPP configuration, network interface 107
PPP negotiation 108
PPPoE connection type 102
PRACK, support 346
pre dial delay, on FXO port 171
preset
E&M interfaces 257
ISDN interfaces 183
R2 interfaces 225
PRI interface configuration
channel range 190
clock mode 188
clock reference 63
copy config to all interfaces 187, 526
endpoint type 187

InformationFollowing operation 196
line coding 189
line framing 189
line type 63, 182, 224, 256
link establishment 192, 202
overlap dialing 192
signaling 63, 224, 256
priority, of network interfaces 106
privacy key
privacy level of calls 503
Progress Indicator in Alerting, ISDN interface 209
Progress Indicator in Call Proceeding, ISDN interface 208
Progress Indicator in Connect, ISDN interface 209
Progress Indicator in Progress, ISDN interface 209
Progress Indicator in Setup ACK, ISDN interface 208
Progress Indicator in Setup, ISDN interface 208
proxy server
SNMP, configuring via
override 330
proxy server, configuring 284
Q
QoS
defined 739
network traffic control 120
VLAN 119
QoS, local
802.1q 119
Differentiated Services (DS) Field 117
QSIG
signalling protocol 189, 200
R
R2
partial reset 226
R2 configuration
channel range 228
clock mode 227
country selection 229
encoding scheme 229
introduction 223
enabling 248
master units 248
slave units 249
line coding 228
line framing 228
line signalling protocol 229
preset 225
PRI statistics 250

R2 digit timers 238
R2 link timers 239
R2 signaling 230
C bit setting 231
D bit setting 231
override default country settings 231
R2 timers 234
R2 tones backward groups 244
R2 tones forward groups 243, 271
register signalling protocol 229
selecting R2 signaling protocol 224
RADIUS
accounting type 609
authentication type 608
servers access rights 610
servers configuration 609
Reason header 345
Referred-by header 345
register home domain override, in SIP 330
register signalling protocol
R2 interface 229
registrar server, configuring 284
registration
delay value, in SIP 301
expiration value, in SIP 297
expiration, in SIP 294, 297
refresh, in SIP 294, 296
regular expression, defined 469
related documentation xxv
Require header, ignoring, in SIP 332
Rescue, network interface 102
reset
factory reset procedure 16
partial reset procedure 15
disabling 15
resolve route header, in SIP 331
restart
unit 50, 55
restart on startup, ISDN interface
ring
distinctive 623
ring management 67
routes, in call routing
call property to compare 487
creating 486
defined 485
deleting 491
destination, of call 490
editing 486
expression to compare 488
mapping, apply 489
moving 490
signalling, of call 489
source to compare 487
RTCP 117
RTP
enforcing symmetric 383
RTP statistics
collection period 387
end-of-connection notification 387
end-of-period notification 388
statistics displayed 385
RTP, early
763
Index
listening to 331
rule, in events
deleting 81
S
scalars, CLI keyword 37
SDP
direction attribute level 328
SDP detect peer direction attribute, in SIP 326
SDP on hold connection address, in SIP 327
second call, service 441
secure communication
introduction 51
secure media 381
secure signalling 309
sending complete information, ISDN interface
services, ISDN
AOC-E support 216, 217
CLIP 214
CLIR 214
override 215
COLP 215
COLR 215
override 216
enabling 213
Maintenance Service Call Termination 216
MSN 218
services, managing 55
session ID/number in Origin field, in SIP 329
session refresh 347
session timers 347
set, CLI keyword 37
show, CLI keyword 38
signal information element, ISDN interface
signaling
PRI configuration 63, 224, 256
signalling properties, in call routing
180 with SDP 502
183 without SDP 503
call properties translation 504
creating 501
defined 501
deleting 505
destination host 502
early connect 502
early disconnect 502
editing 501
moving 505
name 502
privacy level 503
SIP headers translation 504
signalling protocol
ISDN interface
silence detection timeout, on FXO port 174
SIP headers translation, in call routing
built from 506
creating 505
764
defined 505
deleting 507
editing 505
modified header 506
moving 507
name 506
SIP INFO
call waiting private number for 324
SIP INFO without content answer 334
SIP redirects, in call routing
creating 519
defined 519
deleting 520
editing 519
moving 520
SIP, setting
ACK branch matching 332
allow audio and image negotiation 320
allow less media in response 320
allow media reactivation in answer 320
authentication information
creating 304
deleting 306
editing 304
moving 306
call waiting private number for SIP INFO 324
calling party name of the caller ID 168
certificate trust level for TLS connections 308
default registration value in registrations 294
default username value 316
direction attribute "sendonly" 321, 327
direction attributes present 325
DNS SRV record lock 330
early RTP, listening to 331
error causes mapping 339
cause to SIP 342
SIP to cause 340
escape pound (#) in SIP URI Username 323
event handling 351
expiration value in registration 297
force DNS NAPTR in TLS 312
forked provisional reponses behaviour 347
gateways
adding 279
auto-routing, setting for 525
keep alive destinations 289
messaging servers 286
proxy servers 287
registrar servers 285
removing 279
home domain proxy override 349
Ignore OPTONS on no usable endpoints 317
ignore plus (+) character in username 323
ignore Require header 332
keep alive 288
local ring behaviour 328
map plus to TON International 323
Max-Forwards header 324
messaging subscription 353
Offer/Answer model 316, 319, 323
on hold SDP connection address 327
OPTIONS Method Support 316
outbound proxy server 284
loose router status 289
payload type in answer, using 398
penalty box 337

persistent TLS connection port 308
PRACK 346
proxy server 284
quantity of RTP packets 399
Reason header 345
Referred-by header 345
register home domain override 330
registrar server 284
registration configuration 293
registration contact matching 318
registration delay value 301
registration expiration 294, 297
registration refresh 294, 296
reject code for unsupported SDP offer 333, 335
resolve route header 331
SDP detect peer direction attribute 326
SDP direction attribute level 328
secure header 316
session ID/number in Origin field 329
session refresh 347
session timers 347
SIP INFO without content answer 334
supported DNS queries 351
symmetric RTP, enforcing 383
TCP/TLS connect timeout 309, 350
TLS certificate validation 322
TLS client authentication 311
TLS persistent connections status 284
transmission timeout 318
transport type 307
TCP 307
UDP 307
UDP source port behaviour 310
unit registration 293
unsupported Content-Type 335
UPDATE 346
user agents
authentication information 303
contact domain 293
friendly user name 291
header, enabling to send 301
main user name 291
User-Agent Header
format 334
voiceband data mode, call in 394
SipEp, service
endpoint behaviour when unregistered 298
registration parameters 296
user behaviour when unregistered 299
SNMP
command
syntax, normal 652
syntax, row 653
SNMPv3 privacy 575
statistics 580
traps, sending 575
SNMP, service
sending traps 575
statistics 580
SNTP
client 95
remote server, configuring 532, 549, 560
time zone, defining custom 95
Spanning Tree Protocol 568
IPv6 stateless autoconfiguration 105

special tags, in routing types values 476
special vocal features
IP address 163
MAC address 163
SSH
standards supported xxix
Static connection type 102
statistics
SNMP 580
subnets
managing in embedded DHCP server 153
supplementary ISDN services
enabling 213
sysinfo, CLI keyword 39
syslocation, updating 100
syslog daemon
configuring the application 77
diagnostic traces 73
PCM capture 76
IP address 73
messages level 73
view messages in the CLI 25
sysname, updating 100
T
T.38
defined 361
enabling 377
priority 377
tables, CLI keyword 39
TCP
connect timeout 309, 350
transport type 307
Telnet
TFTP
remote server, configuring 451, 531, 549, 560
time zone, configuring 95, 96
TLS
certificate trust levels for connections in 308
certificate validation 322
client authentication 311
connect timeout 309, 350
force DNS NAPTR in 312
persistent connection port 308
persistent connections status 284
tone detection custom cadence, on FXO port 175
tone detection custom frequency, on FXO port 175
transfer protocols
firmware pack update 565
transferring a call
attended transfer 429
blind 429
transmission timeout, setting 318
traps, where to send in SNMP 575
type, CLI keyword 39
765
Index
U
U 110
UDP
source port behaviour 310
transport type 307
unalias, CLI keyword 28
unit
locking/unlocking endpoints 69
restarting 50, 55
shutting down behaviour 71
unlocking endpoints 69
UPDATE, support 346
Uplink, network interface 102
user
add 607
delete 607
modify 607
unregistered, behaviour when 299
user access, setting 17, 18
partial reset 18
secure password policies 17
user gain
input 456
output 456
User-Agent Header
HTTP format 43
SIP format 334
using this manual xxix
using 44
Web, service
TCP port 41
X
X-CCD Clear Channel
defined 361
enabling
payload type 374
priority
for data codecs 374
V
VLAN
creating 115
in QoS 119
vocal features, special
IP address 163
MAC address 163
vocal unit information 163
voice activity detection
G.729 370
voiceband data mode
starting a call in 394
W
wait before answering delay, on FXO port 172
wait for calle to answer, on FXO port 172
web interface
automatic call 423
call completion
CCBS 424
CCNR 424
call forward
on busy 413
on no answer 416
unconditional 418
call hold 439
call transfer
attended 428
blind 428
call waiting 430
conference call 433
introduction 41
second call 440
766

MTX DGW v2-0 SCG

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Discover the Power of 5

Software Configuration Guide

Dgw v2.0 Application

October 08, 2014

Software Configuration Guide

Dgw v2.0 Application Software Configuration Guide

Software Configuration Guide

Key Features ................................................................................................................................................5

Where to Go From Here .............................................................................................................................18

Dgw v2.0 Application

Accessing the CLI.......................................................................................................................................20

Working in the CLI ......................................................................................................................................23

List of Commands / Keywords ....................................................................................................................27

Using the Web Interface .............................................................................................................................44

Dgw v2.0 Application

Software Configuration Guide

Submitting Changes ...................................................................................................................................50

Dgw v2.0 Application

Additional Parameters ................................................................................................................................98

Rescue Interface Configuration ................................................................................................................106

LLDP Configuration ..................................................................................................................................109

Dgw v2.0 Application

Software Configuration Guide

DHCP Server Configuration .....................................................................................................................112

Ethernet Connection Speed......................................................................................................................113

Disabling the Local Firewall ......................................................................................................................127

Network Configuration Examples .............................................................................................................134

Moving a Network Firewall Rule .............................................................................................................................. 141

Disabling the Network Firewall .................................................................................................................141

Creating/Editing a Source NAT Rule ........................................................................................................143

DHCP Basic Configuration .......................................................................................................................153

General POTS Configuration....................................................................................................................162

FXS Configuration ....................................................................................................................................165

FXS Bypass ..............................................................................................................................................169

Dgw v2.0 Application

Software Configuration Guide

FXO Line Verification .............................................................................................................................................. 173

Setting PRI Hardware Parameters ...........................................................................................................181

PRI ISDN Statistics...................................................................................................................................183

BRI Configuration .....................................................................................................................................197

Interop Parameters Configuration.............................................................................................................208

ISDN Timers Configuration .......................................................................................................................212

Selecting the R2 Signaling Protocol .........................................................................................................224

R2 Channel Associated Signaling ............................................................................................................226

R2 Timers Variants ...................................................................................................................................234

Dgw v2.0 Application

R2 Timers Variants.................................................................................................................................................. 235

R2 Digit Timers Variants...........................................................................................................................238

R2 Link Timers Variants ...........................................................................................................................239

R2 Tones Variants ....................................................................................................................................241

IP Sync Configuration ...............................................................................................................................248

PRI R2 CAS Statistics ..............................................................................................................................250

E&M CAS Parameters

Selecting the E&M Signalling Protocol .....................................................................................................256

E&M Channel Associated Signaling .........................................................................................................258

E&M Timers Variants................................................................................................................................265

E&M Digit Timers Variants........................................................................................................................267

E&M Link Timers Variants ........................................................................................................................269

E&M Tones Variants.................................................................................................................................270

IP Sync Configuration ...............................................................................................................................272

PRI E&M Statistics....................................................................................................................................274

Dgw v2.0 Application

Software Configuration Guide