Eacom Evs3080 Ip PBX Training

EACOM Business VoIP Solution Training
04/2009
Agenda
MARKET OVERVIEW
SOLUTION FAMILY
FUNCTION IN DETAILS
Confidential and Proprietary, Eacome Inc. 2008 2

Business Customers Concerns
Growth Cost Function

Future growth Ever-increasing Effectively communicate
requires integration of with customers and
communication cost partners, internal
voice video and web
voice,
management operation

EVS3080/3080 Target Market
• Chained Business • Telephone
• Outsourcing Marketing
• Transportation • Customer Support
• Hotline
• Command and
Dispatching Center
Business Functionality
Model Requirement
Customization Size
Demand Consideration
• Clinic • Small and Middle

• Hospital sized Business

Main Factors Affects VoIP Deployment
Service Quality
Reliabilities – VoIP has yet to prove as reliable as plain phone system
Complexities – traditional IP PBX has to integrate different components from different vendors, e.g. media
gateway, SIP server, voice mail, conference bridge, IP phone, conference phone,
Asterisk like PC based system is prone to various of mis-configurations and system failures
Voice Quality is Challenging
LAN QoS – virus, wiring quality, room quality
WAN QoS – broadband bandwidth, packet loss, jitter buffer,
Gateway Voice Quality–
Quality reliability,
reliability voice quality
IP Phone – compatibility issues, interoperability issues, low cost usually means low quality.
Cost Advantage
CapEx is significantly higher than analog PBX — basic configuration 4
4-5
5 times more than analog IP PBX
with similar functions
Higher OpEx – requires VoIP hardware, software, network expert
Higher
g integration
g cost – installation,, setup
p and technical support
pp are expensive.
p
Limited Functionalities
Open source system has poor performance, proprietary system has poor interoperability
Application development is complicated and expensive for proprietary systems

EVS3080 Technical Features
Single device integrated with multiple functions – secure and reliable
Integrated with data router, media gateway, SIP servers, application server and media server
Purposely designed embedded platform with dedicated DSP processor
Low power design, easily accommodate to the different environment
Highly secure design immunized to virus or software failures
Functioning even without broadband connection
Extensive PBX Feature Support
Basic calling features, operator, auto operator, customizable
greetings and ring back tone
Flexible inbound and outbound call routing scheme, number based, policy based
Customized Feature for Enterprise
p
Flexible trunk options – analog trunking or VoIP trunking
Powerful calling features – call forward, queue, IVR, unified messaging, conferencing, SIP peering
User friendly
y configuration
g – built-in web server GUI for p
provisioning
g and configuration
g
Economics
Capital Expenditure – price close to small PBX, with support of IP phone, IAD or software phone
Operating Expenditure – simple installation & operations, user friendly, cost savings

EACOM Business Communication Solution
Company
LAN
Branch
Office VoIP
Service
Provider

EACOM IP PBX Model
EACOM’s IP PBX’s model name begins with EVS – Eacom VoIP Server
Following EVS symbol are four digits feature code representing the configuration and
Following EVS symbol are four digits feature code representing the configuration and
function difference, which is shown as following

EACOM Business VoIP Solution Chart
EVS30xx IP PBX and
Collaboration Server
VIP
Enterprise
Enterprise EIP
EIP
VoIP Economic
Speaker VoIP
Phone Phone
VoiceCrystal IP
Professional VoIP
Conference Console
Conference Console

Current IP PBX Model List
Model Optional FXS
FXO Ports
FXO Ports Storage Media
Storage Media
Name Ports
EVS3020 2 2 I t
Internal FLASH
l FLASH
EVS3080 4 2 Internal FLASH
EVS3041 4 2 External USB Drive
EVS3080 8 2 Internal FLASH

EACOM VoIP Phone Model
EACOM’s Business VoIP Desktop phone is categorized in 3 families, the Economic,
Enterprise and Professional Conference Console
Economic – full functional VoIP Desktop Phone w/ PoE, LCD and feature keys
Enterprise – Enterprise grade Full Duplex VoIP Desktop Speaker Phone w/ high high
definition voice CODEC,, PoE,, contemporary
p y LCD display
p y and feature key
y
Professional – Professional grade professional VoIP conference console with Crystal
clear sound processing technology and high definition voice CODEC
Model Name
Model Name Product Description
Product Description
Economic
EIP308, EIP310 VoIP Business Desktop Phone
Enterprise
VIP3xx* VoIP Fullduplex Speaker Phone
VIP5xx** VoIP Fullduplex Speakerphone w/ HD voice
Professional
VoiceCrystal IP 2000* Professional VoIP Conference Console
VoiceCrystal IP 4000*** Professional VoIP Conference Console w/ HD Voice
* Available 1Q 2009
** Available 2Q 2009
*** Available 2nd Half 2009
Typical
yp PBX Network Setup
p Scenario – DMZ Mode
EVS3080
I t
Internet
t
Router forward all request

initiated by Internet user to
Linksys the Router’s ppublic IP to
DMZ
W54GT PBX
Ethernet WAN EVS3080B

Switch

Setup PBX in Router
Router’s
s DMZ
Purpose
PBX can expose to the Internet so it is easy for other phones or PBX to find it
PBX can forward media between the PBX/Phones behind the NAT/Firewall
PBX only sees only voice traffic and other traffic still go through router, so PBX can
perform better
Guidelines
Go register a dynamic domain name (DDNS) if router uses dynamic IP address, e.g.
d d
dyndns.org, www.3322.org
3322
Setup router’s DDNS according to specific router instructions
Put PBX in the router’s LAN and assign a static IP address to PBX
Enables router’s DMZ, pointing to PBX’s IP address
Enables router’s port range forwarding to reserve SIP and RTP ports that PBC will
use
Setup DMZ support in PBX, tell PBX to use router’s public IP in SIP signaling
Try to register or access PBX web from other broadband link

Setup DDNS in Router
Choose the DDNS service

provider name, DynDns is
preferred
f d
Fill in the domain name,

user name and password
you got when you
registered the domain with
DDNS service provider

Setup DMZ in Router
Assign a static IP for the PBX

and enable router’s DMZ to point
to PBX IP

Assign Static IP to WAN in PBX
Please use different subnet for LAN an

WAN

Enable DMZ support in PBX
User router’s static IP address

User router’s DDNS domain
external IP address
name for external IP address
Specifies WAN’s subnet

information

User Setup in DMZ mode
In DMZ mode, for the extension that registers to

the PBX from Internet
Internet, please disable “Can
Can Re-
invite” option in the user property, otherwise
one way voice can happen

SIP Server and NAT/Firewall Traversal
Field‐proven SIP Servers
SIP Registration
SIP Proxy
SIP B2B UA EVS3080 forward
voice media flow in
NAT/Firewall Traversal between
Built-in Session Border Controller (RTP Relay)

SIP regular refreshing registration
Media flow relay
Voice media
flow is blocked
by firewall

EVS3080 Deployment Scenario 1 – Trust Host Based Trunking
1. PBX trust each other and does not require authentication – no
need to register to other PBXs
2. Each PBX has its network ID – something like area code in PSTN
3. Each PBX defines prefix dialing to interconnect with other PBX
4.
4 Each
E h PBX hash itsit own connection
ti tot PSTN,
PSTN flexing
fl i routing
ti rules
l
can be set to land the PSTN calls according to best rate among
the PBXs
LAN
EVS3080A PSTN
81xxx
Internet
EVS3080 EVS3080
EVS3080
84xxx 85xxx
83xxx
LAN LAN
LAN

EVS3080 Deployment Scenario 2 – Authentication Based Trunking
1. A and B are independent of each other and have their own
PSTN trunking interface
2. A and B has unified extension plan, all the extensions can be
accessed from A and B
3. A and B has different local PSTN access number but unified
auto attendant
LAN
PSTN
EVS3080A
1. EVS3080A and EVS3080B has independent extension group, A

group is 3xxx
3xxx, B group is 2xxx
2. EVS3080A log in EVS3080B with 2xxx account, EVS3080B EVS3080B
register to EVS3080A with 3xxx account
3. Adding VOIP outgoing dialing rule, in EVS3080A add outgoing LAN
SIP account and dialing 2xxx to reach B, same setup in
EVS3080B
4. EVS3080A and EVS3080B dial out to PSTN with prefix +9xxx
5. EVS3080A can dial out to EVS3080B’s PSTN with prefix
+39xxx, EVS3080B dial out to EVS3080A’s PSTN with prefix
29xxx
+29xxx

EVS3080 Deployment Scenario 2 – Centralized Management
PSTN
LAN A
EVS3080A
EVS3080B
1. All the SIP phones in group B are registered with LAN B

EVS3080A EVS3080B only
EVS3080A, l provides
id llocall PSTN access
and trunking interface
2. No special dialing rules are needed for the calling
extensions
3. EVS3080A registers with EVS3080B to provide access to
EVS3080B’s PSTN trunking interface
4. Prefix based dialing for PSTN dialing out, +9xxx for
EVS3080A local access, +8xxx for EVS3080B local
access

EVS3080 Built-in Web Server Provisioning
g Interface
ONLY Firefox web browser

is fully supported now

How to Change Configuration of PBX
When any y changes

g made, youy
need to save the change and
apply the changes so it can be
effective immediately

User Management
Select one or more

users to configure Click EDIT to change
configurations or DELETE
to delete users

User Configurations
Each configuration item

has online tip notification
to assist user

Bulk Add Users
Specify the starting extension

and number of continuous
extensions to be added

User Management
Open Account
Open batch accounts – setup numbers range
range, static password or random
password
Single Account– static password or random password.
Delete Accounts
Delete batch accounts by selecting multiple users
D l t single
Delete i l user
User management
Each user can setup dedicated voice mail and email account
User defined call forward, find-me, follow-me services
Web GUI based users management, user online status monitoring

User Follow me Configuration
Configure follow me features on

per extension basis. Can ring
number one by one or
simultaneously

PBX Feature
Call Forward
Individual setup of the call forward on conditions
Call transfer
A calls B,, B push
p ‘TRANSFER’ button,, dial C’s number and push
p ‘SEND’ or
‘#’, A and C can talk
Call Park
A calls B, B transfer the call to 700 and hang up. B goes to another location
and dial the 700 to retrieve the call
Call Pickup
Call Pickup
A calls B, B and C are in the same user group, C can get the call by dial “*8”
Call Hold
A calls B, B push ‘HOLD’ button and dial C’s number, B can switch between
the call to B and C by push ‘HOLD’
A can push ‘CONFERENCE’
CONFERENCE to start conference call

Configure the Call Park Features

Outbound Call Control
Multiple Outbound Trunking Interface Options
Integrated PSTN analog FXO media gateway, configurable 2/4/8 port FXO port
Integrated modular FXS media gateway for connecting with analog phone or fax machine
FXO fail-over port for direct connect to the PSTN network when the power is lost
Integrated SIP Peering for multiple EVS3080 deployment
Integrated unlimited VoIP trunking interface seamlessly connected with multiple VoIP service
provider, with call in/call out.
In the distributed deployment model, the outbound call can be routed to the most economic
t ki interface
trunking i t f to
t enter
t into
i t PSTN network,
t k saving
i llong di
distance
t calling
lli costt
VoIP Trunking
Can setup unlimited SIP trunk to the VoIP service providers
providers.
Dial Plan
Prefix based number routing
g
Dialing out number policies
User group based calling rights control

SIP Trunk Setup
Authentication Based
Provide the VoIP Service Provider provided user name
name, password and SIP
server address for authenticating
The SIP server of the authenticated account will automatically become the
tr sted host
trusted

Setup SIP Trunk
Click New SIP trunk to add

outgoing sip trunk

Setup SIP Trunk Setup
Service providers’s domain

name or IP Address, default
port is 5060 or use :xxxx to
specify
PBX’ B
PBX B’s
s SIP registration
name and password
SIP trunk
t k options
ti

SIP Trunk Status
SIP trunk registration status

Outbound Call Control – Define FXO trunk
Give trunk a name and select the

FXO ports should be included in the
trunk
Define the trunk features,

features most importantly
you should decide how FXO decides the
hang up condition, use polarity reversal or
busy tone detection. If busy detection is
enabled, the busy tone frequency and
cadence
d has
h to t be
b sett to
t the
th country
t
standard where the EVS3080 is deployed.

Outbound Call Control – Define Outgoing
g g Calling
g Rule
For each SIP or FXO trunk, you

need to specify a calling rule which
defines what calling prefix should
be used to access the trunk
You can also specify a fall over trunk

when the trunk is unavailable due to
busy or other conditions, the call can
be made through fall over trunk

Outbound Call Control – Define Dialplan
A dialplan is the set of calling rules

and special features that user can use.
Each user can have its own dialplan
setup.

Inbound Call Control
You can also define callee or caller

number based matching
Define how call should be directed,

can be call queue, conference,
voicemail or other special routine

Inbound Call Control
Multiple Inbound Trunking Interface Options
Integrated PSTN analog FXO gateway, configurable 2/4/8 FXO ports
Integrated unlimited VoIP trunking interface seamlessly connected with multiple VoIP
service provider, with call in/call out.
Integrated SIP peering capability enable distributed EVS3080 deployment
Enable enterprise with multiple local numbers to more conveniently connected to
customers
Powerful Number Based Call Routing
Powerful Number Based Call Routing
Redirect to auto attendant
Redirect to extension number
Redirect to voice mail
Redirect to call queue
Redirect to audio conference room
Redirect to voice mail recording

Calling Queue
Queue– Automatic Call Distribution
Customizable queue number
P i it b d
Priority based operator routing, Ring Together or Roundrobin
t ti Ri T th R d bi
Configurable Voice Greetings with prompt files upgrade capability
Can have customized IVR for each queue
Can have customized IVR for each queue

Calling Queue Setup
Define the extension of the call

queue, ring strategy,
background music

Switchboard Setup
Use switch board options to define

voice menu, ring group, upload voice
prompt files, define user directory
and time based routing rules

Switchboard Setup
Auto attendant
Upload voice greetings or record live through phone
Can specify the time triggered auto attendant
Can be a number or voice prompts or customizable IVR
Operators
Can specify multiple operator numbers
Ring together or sequentially ring within operator groups
Support blind or non-blind call transfer
If the transfer failed,
failed the call can be directed to operator again
Calling Status Monitoring and Control
Can install web based calling
g status monitoring
g and control application
pp

GUI Based IVR Menu

Music-on-hold
Music on hold Setup
Use tftp or http to upload music files

to the PBX

Other Functions
Configurable Music‐on‐hold
Allows selection among
g multiple
p music files uploaded
p from PC
Audio convert tools for converting mp3 files to music RAW files
Unified Messaging
Dedicated voice mail for each individual user
USB flash based disk or hard disk based massive storage
Support message waiting indication message
Automatically forward voice mail to user specified email account
IP Broadcasting
IP Broadcasting
Dial 1800, all the users in group 0 will hear the broadcast message

Other Functions
Configurable Music‐on‐hold
Allows selection among
g multiple
p music files uploaded
p from PC
Audio convert tools for converting mp3 files to music RAW files
Unified Messaging
Dedicated voice mail for each individual user
SD card or hard disk based massive storage
Support message waiting indication message
Automatically forward voice mail to user specified email account

Audio Conferencing Setup

Audio Conference Features
Two built in conference room with 10 participants each
Customizable access number, VoIP or PSTN
Customizable access number VoIP or PSTN
5 Ways to dial into conference room
Call from PSTN or VoIP directly to the conference room
User dial the conference room extension
Transferred from AA or operator
Dial from other PBX
Customizable IVR
Conference room PIN management
Conference moderator has special PIN
Normal participants have their own PIN
Advanced Conference Control
Real time conference status monitoring
Participants control from web or DTMF, invite participants, convert to the moderator
Conference Recording

FXO Hardware Parameter Setup
FXO port information, you can also

change the country settings to match
your local network

FXO Hangup Condition Deteciton Setup
FXO related information can be setup

in trunk menu.
FXO supportt busy

b detection
d t ti or
polarity reversal to determine the
hang up condition

Update PBX Firmware
Put image file in tftp or http

server directory, specify the
sever name or URL, then
upload. After uploading,
reboot the PBX

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EACOM Business VoIP Solution Training

Confidential and Proprietary, Eacome Inc. 2008 2

Growth Cost Function

Confidential and Proprietary, Eacome Inc. 2008 4

• Clinic • Small and Middle

Confidential and Proprietary, Eacome Inc. 2008 5

Confidential and Proprietary, Eacome Inc. 2008 6

Confidential and Proprietary, Eacome Inc. 2008 7

Confidential and Proprietary, Eacome Inc. 2008 8

Confidential and Proprietary, Eacome Inc. 2008 10

Confidential and Proprietary, Eacome Inc. 2008 11

Confidential and Proprietary, Eacome Inc. 2008 12

Router forward all request

Ethernet WAN EVS3080B

Confidential and Proprietary, Eacome Inc. 2008 15

Confidential and Proprietary, Eacome Inc. 2008 16

Choose the DDNS service

Fill in the domain name,

Confidential and Proprietary, Eacome Inc. 2008 17

Assign a static IP for the PBX

Confidential and Proprietary, Eacome Inc. 2008 18

Please use different subnet for LAN an

Confidential and Proprietary, Eacome Inc. 2008 19

User router’s static IP address

Specifies WAN’s subnet

Confidential and Proprietary, Eacome Inc. 2008 20

In DMZ mode, for the extension that registers to

Confidential and Proprietary, Eacome Inc. 2008 21

Built-in Session Border Controller (RTP Relay)

Confidential and Proprietary, Eacome Inc. 2008 22

Confidential and Proprietary, Eacome Inc. 2008 23

1. EVS3080A and EVS3080B has independent extension group, A

Confidential and Proprietary, Eacome Inc. 2008 24

1. All the SIP phones in group B are registered with LAN B

Confidential and Proprietary, Eacome Inc. 2008 25

ONLY Firefox web browser

Confidential and Proprietary, Eacome Inc. 2008 26

When any y changes

Confidential and Proprietary, Eacome Inc. 2008 27

Select one or more

Confidential and Proprietary, Eacome Inc. 2008 28

Each configuration item

Confidential and Proprietary, Eacome Inc. 2008 29

Specify the starting extension

Confidential and Proprietary, Eacome Inc. 2008 30

Confidential and Proprietary, Eacome Inc. 2008 31

Configure follow me features on

Confidential and Proprietary, Eacome Inc. 2008 32

Confidential and Proprietary, Eacome Inc. 2008 33

Confidential and Proprietary, Eacome Inc. 2008 34

Confidential and Proprietary, Eacome Inc. 2008 35

Confidential and Proprietary, Eacome Inc. 2008 36

Click New SIP trunk to add

Confidential and Proprietary, Eacome Inc. 2008 37

Service providers’s domain

Confidential and Proprietary, Eacome Inc. 2008 38

SIP trunk registration status

Confidential and Proprietary, Eacome Inc. 2008 39

Give trunk a name and select the

Define the trunk features,

Confidential and Proprietary, Eacome Inc. 2008 40

For each SIP or FXO trunk, you

You can also specify a fall over trunk

Built-in Session Border Controller (RTP Relay)