Avaya B179 SIP Service Guide

Avaya Solution & Interoperability Test Lab
Application Notes for Avaya B179 SIP Conference Phone with Avaya Communication Server 1000 Release 7.5 – Issue
1.0
Abstract
These Application Notes describe a solution comprised of Avaya Communication Server 1000 Release 7.5 and the Avaya B179 SIP Conference Phone. The B179 is a SIP VoIP conference telephone that registers as a standard SIP Line client with Communication Server 1000. The solution supports calling among the B179 and other Communication Server 1000-supported non-SIP and SIP Line clients.
Testing was conducted by the Avaya Solution and Interoperability Test Lab at the request of Product Management.
FS; Reviewed: SPOC 06/8/2011
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1. Introduction
These Application Notes describe a solution comprised of Avaya Communication Server 1000 Release 7.5 and the Avaya B179 SIP Conference Phone. The B179 is a SIP VoIP conference telephone that registers as a standard SIP Line client with Communication Server 1000. The solution supports calling among the B179 and other Communication Server 1000-supported non­SIP and SIP Line clients. Testing was conducted by the Avaya Solution and Interoperability Test Lab at the request of Product Management.
As shown in Figure 1, all telephones, including the B179, are registered to Avaya Communication Server 1000, which is configured as a co-resident single server system. The telephones are configured in the 57xxx extension range.
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Figure 1: Network Configuration
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2. Equipment and Software Validated
Provider Hardware Component Software Version
Avaya Avaya Communication Server 1000E
Avaya Avaya 1120E IP Deskphone UNIStim: 0624C8A Avaya Avaya 1165e IP Deskphone SIP: 04.00.04.00
Avaya Avaya 1230 IP Deskphone SIP: 04.00.04.00 Avaya Avaya 2007 IP Deskphone UNIStim: 0621C8A Avaya Avaya IP Softphone 2050PC UNIStim: 4.01.041 Avaya Avaya M3903 Digital Phone N/A Avaya B179 SIP Conference Phone 2..2
Table 1: Hardware Components and Software Versions
Update
Type
Update Components
Patch None
7.50Q 7.50.17 (see Table 2 for applied updates)
Service
Pack
Deplist
Loadware
cs1000-baseWeb-7.50.17.01-1.i386.000 cs1000-dbcom-7.50.17-01.i386.000
cs1000-sps-7.50.17-01.i386.000 cs1000-linuxbase-7.50.17.04-00.i386.000 cs1000-Jboss-Quantum-7.50.17.01-1.i386.000 cs1000-bcc-7.50.17.03-00.i386.000 cs1000-dmWeb-7.50.17.04-00.i386.001 cs1000-shared-pbx-7.50.17-01.i386.000 cs1000-vtrk-7.50.17-11.i386.000 p30588_1, p30550_1, p30613_1, p30618_1, p30621_1, p30565_1, p30597_1,
p30595_1, p30591_1, p30560_1, p30594_1, p30619_1
IPMG TYPE CSP/SW MSP APP FPGA BOOT DBL1 DBL2
4 0 MGC BD01 AB01 BA07 AA18 BA07 DSP1AB03 N/A
Table 2: CS1000E Applied Updates
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3. Configure Avaya Communication Server 1000
This section describes the steps to configure the following, using CS 1000 Element Manager:
SIP Line serviceSIP Line D-ChannelApplication Module Link (AML)Value Added Server (VAS)Zone for SIP phonesSIP Line Route Data Block (RDB)SIP Line Virtual TrunkMedia Gateway ControllerSIP Line telephone corresponding to the B179 SIP Conference Phone
It is assumed that basic installation and configuration of the CS 1000 call server, signaling server, and node have been completed. Additional configuration details are provided in [1, 2].
3.1. Log in to Element Manager (EM)
Access the Unified Communications Management (UCM) web based interface by using the URL “http://<ip-address>” in an Internet browser window, where “<ip-address>” is the IP address of the call server. Note that the IP address for the Call Server may vary, and in this case “10.7.7.61” is used. Log in with the appropriate user ID and password.
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The following Unified Communications Management screen will be displayed. Click on the Element Name corresponding to the Element Type of “CS1000”.
The CS 1000 Element Manager page appears as shown below.
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3.2. Enable SIP Line Service
Select Customers in the left pane. The Customers screen is displayed. Click the link associated with the appropriate customer, in this case 00. The system can support more than one customer with different network settings and options. In the sample configuration, only one customer was configured on the system.
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The Customer Details screen is displayed next. Select SIP Line Service to edit its parameters.
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Check the SIP Line Service checkbox, enter an appropriate User Agent DN prefix1, and click Save.
3.3. Enable SIP Line Service on Telephony Node
On the Element Manager page, navigate to System IP Network Nodes: Servers, Media Cards. Note the IP address of the Node, as it will be used in configuring the B179 later. Select the Node ID on which SIP Line service is to be enabled.
Figure 2: CS 1000 Node Screen
1
The User Agent DN Prefix is used to form the User Agent DN. See Section 3.11.
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Scroll down the top section to display the Applications section on the right, and click on SIP
Line.
Figure 3: Node Details Screen
The SIP Line Configuration Details page is displayed. Check Enable gateway service on this node next to SIP Line Gateway Application:. Then click Save.
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Return to the Node Details Screen (Figure 3) and click on Voice Gateway (VGW) and Codecs. For G.722 and G.729 support, check Enabled next to Codec G.722: and Codec G.729:. If G.729 Annex B (silence suppression) is desired as in the sample configuration, check the Voice Activity Detection (VAD) checkbox. Note that the VAD setting should be consistent with the VAD setting in the B179 configuration (see Section 4.2 Figure 9). Click Save. Then click Save on the Node Details screen (Figure 3).
Figure 4 – Node Codec Selection
Select Transfer Now on the Node Saved page as shown below.
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Once the transfer completes, the Synchronize Configuration Files (Node ID <id>) page is displayed.
Check the appropriate Call Server and click Start Sync. The screen will automatically refresh until the synchronization is finished. The Synchronization Status field will update from Sync
required (as shown) to Synchronized (not shown). After synchronization completes, click Restart Applications to use the new SIP Gateway settings.
3.4. Configure SIP Line D-Channel
On the left column menu of the main Element Manager page, navigate to Routes and Trunks D-Channels. Under the Configuration section, select a D-Channel number from the Choose a D-Channel Number list (channel 3 in the sample configuration), and select DCH for the type.
Click to Add.
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The D-Channels Property Configuration screens below show the parameter values after configuring the D-channel. DCIP is selected for D channel Card Type, Meridian Meridian1 (SL1) is selected for Interface type for D-channel, and an appropriate Designator is entered. The remaining parameters have their default values.
Click the Basic options (BSCOPT) link to expand that section. Click Edit to configure Remote
Capabilities.
Figure 5 – D-Channel Basic Options
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