Configuring Alcatel OmniPCX Enterprise R9.1 with Avaya Communication Server 1000E R7.5 and ...

47 Pages·2011·2.36 MB·English

Avaya Solution & Interoperability Test Lab

Configuring Alcatel OmniPCX Enterprise R9.1 with
Avaya Communication Server 1000E R7.5 and Avaya
Aura® Session Manager R6.1 – Issue 1.0

Abstract

These Application Notes present a sample configuration for a network consisting of an
Avaya Communication Server 1000E and Alcatel OmniPCX Enterprise. These two
systems are connected via a common Avaya Aura® Session Manager.

Testing was conducted via the Internal Interoperability Program at the Avaya Solution
and Interoperability Test Lab.

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1. Introduction
The purpose of these interoperability Application Notes is to document Alcatel
OmniPCX Enterprise (OXE) with Avaya Communication Server 1000E (CS1000E)
which are both connected to an Avaya Aura® Session Manager via a separate SIP Trunk.
Voicemail integration between Alcatel OmniPCX Enterprise and Avaya Aura®
Messaging was not included in the scope of these Application Notes. The sample network
is shown in Figure 1, where the Alcatel OmniPCX Enterprise supports the Alcatel
ipTouch 4038 and 4068 IP Telephones. SIP trunks are used to connect Avaya
Communication Server 1000E and Alcatel OmniPCX Enterprise to Avaya Aura® Session
Manager. All inter-system calls are carried over these SIP trunks. Avaya Aura® Session
Manager can support flexible inter-system call routing based on dialed number, calling
number and system location, and can also provide protocol adaptation to allow for multi-
vendor systems to interoperate. Avaya Aura® Session Manager is managed by a separate
Avaya Aura® System Manager, which can manage multiple Avaya Aura® Session
Managers.

Figure 1: Connection of Alcatel OmniPCX Enterprise and Avaya Communication
Server 1000E via Avaya Aura® Session Manager using SIP Trunks

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Alcatel phones are registered to Alcatel OmniPCX Enterprise. Alcatel OmniPCX
Enterprise registered stations use extensions 36xxx. One SIP trunk is provisioned to the
Avaya Aura® Session Manager to manage calls to/from Alcatel OmniPCX Enterprise.
One SIP trunk is provisioned to the Avaya Aura® Session Manager to manage calls
to/from Avaya Communication Server 1000E.
1.1. Conclusion and Observations
As illustrated in these Application Notes, Alcatel OmniPCX Enterprise can interoperate
via Avaya Aura® Session Manager with Avaya Communication Server 1000E using SIP
trunks. The following interoperability issue was observed:
 For conference calls and attended/unattended transfers, phone displays were not
updated correctly for both username and number. This is an Alcatel OmniPCX
Enterprise issue as SIP 180 & 200 messages sent by Alcatel OmniPCX Enterprise
do not contain a user part in the contact header.
2. Equipment and Software Validated
The following equipment and software/firmware were used for the sample configuration:

Equipment Software/Firmware
Avaya Aura® Session Manager R6.1 SP3
Avaya S8510 Server
(Build 613006)
Avaya Aura® System Manager R6.1 SP3
Avaya S8800 Server
(Build 6.1.0.0.7345-6.1.5.112)
Avaya Communication Server 1000E R7.5,
Avaya Communication Server 1000E Version7.50.17
CP+PM Co-resident Linux Update 7.50_17 Sept13
Deplist: 7.50Q
Avaya 1140E Telephone UNIStim 0625C8A
Avaya 1120E Telephone UNIStim 0625C8A
Avaya 1120E Telephone SIP 4.01.15.00
Alcatel OmniPCX Enterprise Server 9.1 (I1.605-16-c)
Alcatel ipTouch NOE Telephone [phone]
Alcatel ipTouch NOE Telephone [phone]

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3. Configure Avaya Communication Server 1000E
This section describes the details for configuring Communication Server 1000E to route
calls to Session Manager over a SIP trunk. In the sample configuration, Communication
Server 1000E R7.5 was deployed as a co-resident system with the SIP Signaling Server
and Call Server applications all running on the same CP+PM server platform.

Session Manager R6.1 provides all the SIP Proxy Service (SPS) and Network Connect
Services (NCS) functions previously provided by the Network Routing Server (NRS)
application. As a result, the Network Routing Server application is no longer needed to
configure a SIP trunk between Communication Server 1000E R7.5 and Session Manager
R6.1.

These instructions assume System Manager has already been configured as the Primary
Security Server for the Avaya Unified Communications Management application and
Communication Server 1000E is registered as a member of the System Manager Security
framework. For more information on how to configure System Manager to integrate with
the Avaya Unified Communications Management application, see Reference [5] in
Section 7.

In addition, these instructions also assume the configuration of the Call Server and SIP
Signaling Server applications has been completed and Communication Server 1000E is
configured to support the 1140e SIP & UNIStim telephones. For information on how to
administer these functions of Communication Server 1000E. See References [6] through
[8] in Section 7.

Using the System Manager web interface, the following administration steps will be
described:
 Access Avaya Unified Communications Management (UCM) web interface from
System Manager
 Confirm Node and IP addresses
 Confirm Virtual D-Channel, Routes and Trunks
 Configure SIP Trunk to Avaya Aura® Session Manager
 Configure Route List Index and Distant Steering Code
 Save Configuration

Notes: Some administration screens have been abbreviated for clarity.

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3.1. Access Avaya Unified Communications Management
(UCM)Web Interface from System Manager
Point your browser to http://<ip-address>/SMGR, where <ip-address> is the IP
address of System Manager. Log in with the appropriate credentials. The System
Manager Home Page will be displayed. Under Services category on the right side of the
page, click on the UCM Services link.

The Avaya Unified Communications Management Elements page will open in a new
browser window. Click on the Element Name corresponding to CS1000 in the Element
Type column.

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3.2. Confirm Node and IP Addresses
Expand System IP Network on the left panel and select Nodes: Servers, Media Cards.
The IP Telephony Nodes page is displayed as shown below. Click Node id in the Node
ID column to view details of the node. In the sample configuration, 100 was used.

The Node Details screen is displayed with additional details as shown below. Make a
note of the Call server IP address and Signaling Server TLAN IPv4 address fields
highlighted below as these values are used to configure other sections.

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3.3. Confirm Virtual D-Channel, Routes and Trunks
Communication Server 1000E Call Server utilizes a virtual D-channel and associated
Route and Trunks to communicate with the Signaling Server. This section describes the
steps to verify that this administration has already been completed.
3.3.1. Confirm Virtual D-Channel Configuration
Expand Routes and Trunks on the left navigation panel and select D-Channels. The
screen below shows all the D-channels administered on the sample configuration.

In the sample configuration, there is a single D-channel assigned to Channel:15 with
Card Type: DCIP. Specifying DCIP as the type of channel indicates the D-channel is a
virtual D-channel.
3.3.2. Confirm Routes and Trunks Configuration
In addition to configuring a virtual D-channel, a Route and its associated Trunks need to
be administered.

Expand Routes and Trunks on the left navigation panel and select Routes and Trunks (not
shown) to verify that a route with enough trunks to handle the expected number of
simultaneous calls has been configured.

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As shown in the screen below, Route 15 has been configured with 30 trunks which
indicates the system can handle 30 simultaneous calls.

Select Edit to verify the configuration. The details of the virtual Route defined for
sample configuration is shown below. Verify SIP (SIP) has been selected for Protocol
ID for the route (PCID) field and the Node ID of signaling server of this route
(NODE) and D channel number (DCH) fields match the values identified in the
previous section.

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3.4. Configure SIP Trunk to Avaya Aura® Session Manager
Expand System  IP Network  Nodes: Servers, Media Cards. Click 100 in the Node
ID column (not shown) to edit configuration settings of node. Using the scroll bar on the
right side of the screen, navigate to the Applications section on the screen and select the
Gateway (SIPGw) link as highlighted below.

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On the Node ID: 100 - Virtual Trunk Gateway Configuration Details page, enter the
following values and use default values for remaining fields.
 SIP domain name Name of domain (In the sample configuration,
mmsil.local was used.)
 Local SIP port 5060
 Gateway endpoint name Descriptive name
 Application node ID Node ID (From Section 3.2 above)

The values defined for the sample configuration are shown below.

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