Avaya B179 SIP User manual

JPA; Reviewed:
SPOC 07/08/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
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B179SIPCMSM
Avaya Solution & Interoperability Test Lab
Configuring the Avaya B179 SIP Conference Phone with
Avaya Aura® Communication Manager and Avaya Aura®
Session Manager – Issue 1.0
Abstract
These Application Notes describe the steps to configure the Avaya B179 SIP Conference
Phone to work with Avaya Aura® Communication Manager and Avaya Aura® Session
Manager.

JPA; Reviewed:
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Solution & Interoperability Test Lab Application Notes
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1. Introduction
These Application Notes describe a basic configuration of the Avaya B179 SIP Conference
Phone to work with Avaya Aura® Communication Manager and Avaya Aura® Session
Manager.
Figure 1 shows the configuration used for testing.
Figure 1: Network Diagram

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2. Equipment and Software Validated
The following equipment and software were used for the configuration provided:
Hardware/Software Component Software Version
Communication Manager (S8300D Server
with G430 Media Gateway) R016x.00.1.510.1
Session Manager (S8800 Server) 6.1.0.0.610023
System Manager (S8800 Server) 6.1.0 (6.1.0.4.5072-6.1.4.62)
9640G IP Telephone (SIP) 2.6.4 (SIP96xx_2_6_4_0.bin)
9641G IP Telephone (SIP) 6.0 (96x1_SALBR6_0_V4r52.tar)
9608 IP Telephone (H.323) 6.008
(S9608_11_HALBR6_0_0_8S_V4r52.tar)
6408+ Digital Telephone N/A
Desktop Video Device 1.0.0 (1_0_0_012849)
one-X® Communicator (SIP) 6.0.1.16-SP1-25226
one-X® Communicator (H.323) 6.0.1.16-SP1-25226
300 IP SIP Telephone 2.2 & 2.2.1
3. Configuration of the Avaya B179 SIP Conference
Telephone
As shown in Figure 2 below, use the buttons on the B179 SIP Conference Phone to config the IP
address settings. The B179 can be set with a static IP address or use DHCP. In this case a static
IP Address was entered, and VLAN 7 was used.
On the Phone select MENU then use the ▲and▼keys to get to the SETTINGS menu, then Click
OK (OK is acts as the select key). In the SETTINGS menu (not shown) select ADVANCED, and
enter the required password. Navigate to the NETWORK menu, and then select IP. Set the IP
Address to either DHCP or STATIC. A dark box next to an entry indicates it is selected. If static,
click OK and step through the prompts to enter the IP Address, Mask, and Gateway for the
endpoint, clicking OK after each entry. At the end the NETWORK menu re-appears. If a VLAN
is to be used, select ETHERNET, followed by VLAN. Select ENABLE VLAN, enter the VLAN
number to be used, followed by OK. When done continue to press C until you reach the main
menu, which will cause the phone to reboot. At this point the remaining configuration can be
done via the web interface.

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Figure 2: Menu Navigation Buttons
Using an HTTP browser, enter the IP address of the B179 SIP Conference Phone in the browser
URL field. The Login screen will appear as shown in Figure 3. Select Admin for the login and
enter the appropriate password, then click Login.
Figure 3: Login Screen

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Once logged in, the initial display screen appears as shown in Figure 4.
Figure 4: Initial Screen
Select Settings and then Network. Figure 5 shows the configuration used in this test. The
network section shows the IP Address information previously entered. The Quality of Service
section shows the VLAN assigned. In this section the user may also assign QoS tagging if the
network the phone is connected to supports QoS.
Figure 5: Network Settings Screen

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Select the Media tab. Figure 6 shows the configuration used in this test. The Codec section
selects the priority for each codec supported. If a codec is not to be used the priority should be
set to 0. If the G.729 codec is to be used, VAD should be Yes. For full feature functionality, it’s
required that the G.711 codec be included, even if it’s the lowest priority on the list.
Figure 6: Media Settings Screen

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Select the SIP tab. Figure 7 shows the configuration used in this test.
Account 1:
-Enable account: Yes
-Account Name: Use a meaningful name.
-User: Enter the extension assigned in Session Manager, in this case 36052.
-Realm: * (the default)
-Authentication Name and Password: Enter the extension and password assigned in
Session Manager.
-Registrar and Proxy: Enter the IP Address for Session Manager, in this case
10.1.2.210.
-Registration interval: Use the default 300
Tranport:
-Protocol: TCP
-Local Port: 5060
Figure 7: SIP Settings Screen
The rest of the configuration was left at default values. When complete, click Save. This will
cause the phone to reboot.

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4. Configuration of the Avaya Aura® Session Manager
This section shows the Session Manager screens used to configure an endpoint with extension
36052, for the B179 SIP Conference Phone. For additional information on configuring Session
Manager, see reference [3].
Login to System Manager and select User Management Manage Users. A list of users will
appear. Click New to create a new endpoint. The Identity and Comunication Profile tabs need to
be configured for a new user.
Figure 8 shows the Identity configuration used. Enter a meaningful Name for the user, as well
as a meaningful Localized and Endpoint Display Name. For Login Name enter the extension
and domain that the endpoint will use to register to Session Manager. Enter a password for the
user. Note that this password is not the password the endpoint will use to register to Session
Manager.
Figure 8: User Profile Identity
Figure 9 shows the Communication Profile configuration used.
Communication Profile:
-Communication Profile Password: Enter the password the endpoint will use to
register to Session Manager
-Confirm Password: Re-enter the same password.

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Communication Address:
-Click New to create a new entry.
-Type: Avaya SIP
-Fully Qualified Address: Enter appropriate Handle (extension) and select
appropriate Domain.
-Click Add to add the new entry.
Session Manager Profile:
-Primary Session Manager: Select appropriate Session Manager
-Origination and Termination Application Sequence: Select appropriate sequence
-Home Location: Select appropriate location.
Endpoint Profile:
-System: Select appropriate Communication Manager System
-Profile Type: Endpoint
-Extension: Enter extension for this endpoint, 36052
-Template: Select the appropriate template. DEFAULT_9630SIP_CM_6_0 was used.
-Port: IP (this will change to a virtual port when form is submitted)
Click Commit (not shown) when done.
Figure 9: User Communication Profile

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5. Configuration of the Avaya Aura® Communication
Manager
Basic configuration of Communication manager is outside the scope of this document. A station
to support the B179 SIP Conference Phone is created in Communication Manager automatically
when using System Manager to add a User in Session Manager, as described in Section 4.
If the native conference feature of the B179 SIP Conference phone is to be used, it’s necessary
that the station in Communication Manager have at least 4 call appearances, as each call initiated
by the B179 SIP Conference Phone will use a call apperance. This was manually done via the
SAT interface in Communication Manager, using the “change station 36052” command, as
shown in Figure 10. For additional information on configuring Configuration Manager, see
references [4] and [5]. System Manager can also be used to add/change an endpoint’s button
assignments.
Figure 10: Change Station
6. Observations
At this time, for full feature functionality, it is recommended that multiple codecs be configured
using the priority value to determine the preference. At a minimum one of the G.711 codecs must
be selected. This may not be required in future releases.
UDP cannot be used as the SIP signaling transport protocol if VLAN tagging is used.
For full functionality with the conference feature, native to the B179 SIP Conference Phone, the
station should be administered in Communication Manager with at least 4 call appearances.

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7. Verification Steps
Look at the screen of the B179 SIP Conference Phone. A dark square next to the extension
indicates the endpoint has succesfully registered. A clear/empty square indicates the endpoint is
not registered.
Figure 11: Registered Endpoint
If the endpoint is not registered, verify that the configuration was entered correctly. When the
endpoint registers verify it is working properly by placing calls to and from the enpoint.
8. Conclusion
These Application Notes described the configuration for the Avaya B179 SIP Conference Phone
to connect to Session Manager and Communication Manager. The Avaya B179 SIP Conference
Phone sucessfully registed and was able to place and receive calls. Some observations were
noted in section 6.
9. Additional References
The Avaya product documentation is available at http://support.avaya.com.
[1] Avaya B179 SIP Conference Phone User Guide
[2] Avaya B179 SIP Conference Phone Installation and Administration Guide
[3] Administering Avaya Aura® Session Manager; Issue 1, 03-603324, Release 6.1, Nov 2010
[4] Administering Avaya Aura® Communication Manager; Issue 6.0, 03-300509, Release 6.0,
June 2010
[5] Avaya Aura® Communication Manager Screen Reference; Issue 2.0, 03-602878, Release
6.0, June 2010

JPA; Reviewed:
SPOC 07/08/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
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B179SIPCMSM
©2011 Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks
are the property of their respective owners. The information provided in these Application
Notes is subject to change without notice. The configurations, technical data, and
recommendations provided in these Application Notes are believed to be accurate and
dependable, but are presented without express or implied warranty. Users are responsible for
their application of any products specified in these Application Notes.
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya Solution &
Interoperability Test Lab at [email protected]