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Enterprise SIP
I n t e g r a t i o n Te c h n i c a l N o t e
ii I n t e g r a t i o n Te c h n i c a l N o t e
Edition Note
This is the tenth edition of the Alcatel-Lucent OmniPCX Enterprise SIP Integration Technical
Note. Information in this document applies to CallXpress version 8.20.
P/N 1081-50553-00 Rev 10 October 2011
Produced by the AVST Technical Communications Department
Applied Voice & Speech Technologies, Inc., reserves the right to revise this guide without
notice.
Copyright Notice
2010 Applied Voice & Speech Technologies, Inc. (AVST). No part of this publication may
be reproduced, transmitted, transcribed, stored in a retrieval system, adapted, or translated
into any language in any form by any means without the written permission of AVST.
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protected by trademark and other laws of the United States, as well as international
conventions and the laws of other countries. Other such properties that are not owned by
AVST may not be used without the express permission from their owners.
Disclaimer
AVST exercises care in researching other manufacturers specifications but disclaims all
responsibility for their accuracy. All specifications shown here are subject to change without
notice.
Trademarks
CallXpress and AVST are a registered trademarks and Applied Voice & Speech
Technologies, and the AVST logo is a trademark of Applied Voice & Speech Technologies,
Inc.
Windows is a registered trademark of Microsoft Corporation
RADVISION is a registered trademark of RADVISION Inc.
All other products, brand names, or companies mentioned in this document may be
trademarks or registered trademarks of their respective companies.
Table of Contents
Overview ................................................................................................................................... 1
References ............................................................................................................................ 2
Features Supported by This Integration ................................................................................. 2
Overview
This Integration Technical Note (ITN) is written for dealers who are experienced with CallXpress
and are familiar with its procedures and terminology. This document also assumes that you are
familiar with the features and programming of the Alcatel-Lucent telephone system.
This document describes how to integrate CallXpress with an Alcatel-Lucent telephone system,
using the Session Initiation Protocol (SIP) integration. This integration operates exclusively over
a TCP/IP-based network; it uses no analog or digital voice telephony ports, but passes voice
communication and signaling information over the network.
The Alcatel-Lucent integration is a SIP trunk integration, CallXpress registers itself as a single
voice mail pilot number at the telephone systems SIP proxy. The proxy routes all incoming calls
for CallXpress to that pilot number, and CallXpress assigns them to lines as needed. CallXpress
also sends outbound calls and MWI messages to the pilot number so that the proxy can act on
them.
This ITN documents the procedure for setting up the integration. The process consists of
programming the Alcatel-Lucent OmniPCX Enterprise and configuring CallXpress. Critical
application considerations are also documented.
Use this document in conjunction with the Installing CallXpress and Administering CallXpress
guides and with the CallXpress online help system.
References
Information that is more current may be available from the following source:
World Wide WebAVST certified technicians can view or download documents and
program files from our web site: http://www.partnerxpress.com.
The following print documents are available for your reference:
Installing CallXpress
Administering CallXpress
Table 2: Alcatel-Lucent OmniPCX Enterprise SIP station set integration features supported
Feature Supported Notes
Automatic subscriber logon Yes
ANI/CLI Yes
Announce Busy greeting on forwarded calls Yes
Call screening Yes Note 1
Caller queuing Yes Note 2
DNIS Yes
End-to-end DTMF, attendant console Yes
End-to-end DTMF, proprietary telephones Yes
Fax Detection No
Internal calling party ID for reply Yes
Live record, integrated Yes
Live reply to sender Yes
Message notification callouts Yes
MWI, set/clear Yes
MWI, inband/outband Outband
Networking, analog Yes
Overflow from CallXpress to attendant Yes
Overflow to CallXpress from attendant Yes
PBX-provided disconnect signaling Yes
Revert to operator Yes
Transfers, blind Yes
Transfers, confirmed Yes
Transfers, fully supervised Yes
Transfers, monitored Yes
Trunk ID for call routing No
Notes
1. Only available when using supervised transfers
2. Caller Queuing is specific to each local Call Server. Call Servers within the system are
unaware of queued calls to the same subscriber on other Call Servers. For more
information, refer to the Critical Consideration Notes.
Installation Requirements
Review the following information before performing any of the procedures in this document. To
successfully install this integration, you must meet the installation requirements for both the
telephone system and CallXpress.
CallXpress Requirements
CallXpress version 8.20
AVST software key and feature file with the Alcatel OmniPCX Enterprise SIP integration
enabled and one RADVISION SIP and RTP license enabled for each port involved in
the integration
One or two network interface cards
* NOTE Alcatel-Lucent OmniPCX Enterprise release 9.0 and later requires a fully
qualified domain name (FQDN) as the Machine Name-Host.
IMPORTANT You must restart the Call Server Processing Unit to apply your changes on
PBX versions prior to 9.0.
Configuring CallXpress
Once the telephone system is programmed, you must configure CallXpress for the integration.
During setup, you need the following information, specific to this integration, to configure
CallXpress.
To configure CallXpress
1. If you are configuring the Call Server for the first time, enter the name of your site in the
Site Name box of the Installation Configuration dialog box, the mailbox length in digits in
the Mailbox Length box, and the first extension number for the first line in the First
Extension box.
2. Select Alcatel-Lucent as the manufacturer, OmniPCX Enterprise as the model, and
then SIP as the integration type.
3. Click OK. The Switch Options dialog box displays.
4. Make any changes to the default settings your site requires, and then click OK.
5. The Integration Options dialog box, Required Parameters View displays.
Field Value
available local TCP/IP addresses.
SIP Local Connection Port The port on which CallXpress listens for incoming
SIP messages. The default value is 5060.
SIP parser qualifier string The IP address of the Call Server platform. This
field allows the telephone system to locate the SIP
parser file on the platform.
Media packet size (milliseconds) CallXpress sends/receives packets containing the
number of milliseconds worth of audio data set
here. The default value is 20.
7. From the Local Integration Settings View, select Software DTMF Detection Settings.
The Software DTMF Settings view displays.
8. Confirm the DTMF Detection Type parameter is set to Hardware, the default value.
9. Click OK. The Switch Section Options dialog box displays.
10. In the Switch Section Options dialog box, select an Incoming Hunt Mode, and then type
the Hunt Group Access Code. In this example, 5000
NOTE This integration only supports terminal, circular, reverse terminal, and reverse
circular hunt modes.
This is a typical configuration when the servers are geographically separated and are located on
different subnets (also known as spatial redundancy). The DNS server on the IP network is
responsible to resolve the FQDN to an IP address. The DNS server is configured to delegate
the request to the unique OXE nodes for the FQDN associated with the OXE. The OXE node
whose current role is MAIN responds to the DNS queries and provides its IP address. If the
OXE role switches, the node responding to the DNS queries also switches. Therefore, the
switch failover is transparent to CallXpress as long as it is configured to work with FQDN
instead of an IP address. The failover server configuration should be configured to use the
FQDN.
CallXpress can support switch failover even if the OXE nodes are not configured to use a
FQDN. In this case, follow the steps outlined below to configure the IP addresses of the primary
and failover OXE nodes.
3. Click the Add Failover Server button. Two new rows are added to configure the
secondary SIP server.
4. In the Failover Server Settings view, enter Secondary SIP Server Address and the
Secondary SIP Server Port.
Table 4: Secondary SIP Server Address and the Secondary SIP Server Port example
Field Value
(1) Secondary SIP Server Enter the TCP/IP address or an FQDN of
Address the secondary node. For example, the IP
address 123.45.6.789 as displayed on the
Review/Modify SIP Gateway screen. (See
page 8 of this document.)
NOTE Alcatel-Lucent OmniPCX Enterprise
release 9.0 requires the Machine Name to
be a fully qualified domain name. Therefore,
when integrating with release 9.0 or later,
use the Machine name field as displayed on
the Review/Modify SIP Gateway screen.
The default value is blank.
IMPORTANT This value must match the
configuration on the Gateway of the
secondary node.
(1) Secondary SIP Server Port The default value is 5060
5. From the View drop-down box, select Integration Specific Parameters. The Integration
Options dialog box Integration Specific Parameters view displays.
The instructions in this section ensure that the binding order is correct when you set up the
integration. However, if you replace a NIC on the CallXpress server platform later, the platform's
operating system registers the new adapter at the bottom of its binding order. Restoring the
original binding order should correct any problems caused by the change.
IMPORTANT The following procedure shifts the binding order of the network interface cards. To
determine which NIC is associated with a specific network connection, right-click the connection
in the Network Connections window, and then select Properties.