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
Solution & Interoperability Test Lab Application Notes
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.
FS; Reviewed:
SPOC 06/8/2011
Figure 1: Network Configuration
Solution & Interoperability Test Lab Application Notes
This section describes the steps to configure the following, using CS 1000 Element Manager:
SIP Line service
SIP Line D-Channel
Application Module Link (AML)
Value Added Server (VAS)
Zone for SIP phones
SIP Line Route Data Block (RDB)
SIP Line Virtual Trunk
Media Gateway Controller
SIP 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.
FS; Reviewed:
SPOC 06/8/2011
Solution & Interoperability Test Lab Application Notes
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.
FS; Reviewed:
SPOC 06/8/2011
Solution & Interoperability Test Lab Application Notes
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.
FS; Reviewed:
SPOC 06/8/2011
Solution & Interoperability Test Lab Application Notes
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.
FS; Reviewed:
SPOC 06/8/2011
Solution & Interoperability Test Lab Application Notes
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 4.2Figure 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.
FS; Reviewed:
SPOC 06/8/2011
Solution & Interoperability Test Lab Application Notes
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.
FS; Reviewed:
SPOC 06/8/2011
Solution & Interoperability Test Lab Application Notes
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
FS; Reviewed:
SPOC 06/8/2011
Solution & Interoperability Test Lab Application Notes