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.

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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 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.

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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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The Remote Capabilities Configuration page is displayed. Select the Message waiting

interworking with DMS-100 (MWI) check box,2 and the Network name display method 2
(ND2) check box. At the bottom of the Remote Capabilities Configuration page, click Return

- Remote Capabilities (not shown), and the D-Channel Property Configuration page

reappears. Click on Submit (see lower left in Figure 5).

2

Note that although the Avaya B179 Conference Telephone does not support Message Waiting Indicator, this D

channel can also be used for other SIP Line IP telephones that do support it, so it is enabled here for that purpose.

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3.5. Configure Application Module Link (AML)

On the left column menu of the main Element Manager page, navigate to System  Interfaces
 Application Module Link, and click Add (not shown). The New Application Module Link

page is displayed. Enter the AML port number in the Port number text box. The SIP Line
Service can use ports 32 through 127. In the sample configuration, the SIP Line Service is
configured to use port 32. Enter an appropriate Description. Click Save to save the
configuration.

3.6. Configure Value Added Server (VAS)

On the left column menu of the main Element Manager page, navigate to System  Interfaces
 Value Added Server. Click Add and then click Ethernet LAN Link on the Add Value

Added Server page that is displayed next (not shown). On the Ethernet Link page that is
displayed next (not shown), enter a Value added server ID (64 in the sample configuration),
and select the AML number created in the previous section for Ethernet LAN Link. Ensure
that the Application Security check box is unchecked. Click Save (not shown). The screen
below shows the result of adding the value added server.

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3.7. Configure Zone for SIP Phones

On the left column menu of the main Element Manager page, navigate to System  IP Network
 Zones. On the Zones page, select Bandwidth Zones (not shown), and on the Zone Basic

Property and Bandwidth Management page, enter a Zone number (ZONE) and an
appropriate Description. Defaults can be used for the remaining fields. Click Save.

3.8. Configure SIP Line Route Data Block (RDB)

On the left column menu of the main Element Manager page, navigate to Routes and Trunks 
Routes and Trunks. Click Add route for the appropriate customer number.

The following screen shows the parameter settings after the route has been added. Set the
following parameters and leave default values for the remaining parameters. The Basic Route
Options, Network Options, General Options, and Advanced Configurations sections (not
shown) can be left at the defaults. Click Submit (not shown) to save the configuration changes.

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Route number (ROUT) Select a route number
Designator field for trunk (DES) Enter an appropriate name
Trunk type (TKTP) Select TIE trunk data block (TIE)
Incoming and outgoing trunk (ICOG) Select Incoming and Outgoing
(IAO)
Access code for the trunk route (ACOD) Enter the access code
The route is for a virtual trunk route (VTRK) Check the box
Zone for codec selection and bandwidth Enter a zone3
management (ZONE)
Node ID of signaling server of this route (NODE) Enter the node ID of the SIP Line

Gateway

Protocol ID for the route (PCID) Select SIP Line (SIPL)
Integrated services digital network option (ISDN) Check the box
Mode of operation (MODE) Select Route uses ISDN Signaling
Link (ISLD)
D channel number (DCH) Enter the D-channel number
Interface type for route (IFC) Select Meridian M1 (SL1)
Network calling name allowed (NCNA) Check the box
Network call redirection (NCRD) Check the box
Trunk route optimization (TRO) Check the box

3

Note that this must be a zone of type VTRK and must be different than the zone created for the SIP phones in

Section 3.7. In the sample configuration, the VTRK zone was 1.

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3.9. Configure SIP Line Virtual Trunk

When the Routes and Trunks screen is displayed after adding the route in Section 3.8, click
Add trunk corresponding to the newly added route to add new trunk members. The following

screen shows the parameter settings for one of the trunks after they have been added. Set the
following parameters and leave default values for the remaining parameters. Click Save to save
the configuration changes.

Multiple trunk input number Enter the number of trunks (only shown
when adding trunks)

Trunk data block Select IP Trunk (IPTI)
Terminal number An available terminal number.
Designator field for trunk A descriptive text.
Extended trunk Select Virtual trunk (VTRK)
Route number, Member number Current route number and starting member.

(only shown when adding trunks)

Card density Select Octal Density (8D)
Start arrangement Incoming Select Wink or Fast Flash (WNK)
Start arrangement Outgoing Select Wink or Fast Flash (WNK)
Trunk group access restriction Desired trunk group access restriction level.
Channel ID for this trunk An available starting channel ID.

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3.10. Configure Media Gateway Controller

This section describes configuration of the G.729 audio codec for the Media Gateway Controller
(MGC) to support calls between the B179 and non-IP telephones. On the left column menu of
the main Element Manager page, navigate to IP Network  Media Gateways. Click on the
IPMG that supports the digital and analog phones in the system.

On the IPMG Property Configuration screen, click Next (not Shown). Expand the VGW and
IP phone codec profile section. In that section, check the Select checkbox next to and expand
the Codec G729A section.

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If Annex B support is desired as in the sample configuration, check the 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.

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Figure 6 – MGC Codec Selection

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When the Media Gateway screen returns, select the radio button for the IPMG and click Reboot.

3.11. Configure SIP Line Telephone

This section describes the screens for configuring a SIP Line telephone to support the Avaya
B179 Conference Telephone. On the left column menu of the main Element Manager page,
navigate to Phones. On the Search For Phones page, click Add….

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On the New Phones page, select the Customer, select the Phone Type radio button, and then select
UEXT-SIPL – Universal Extension SIPL. Click Preview.

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The following screens show the parameter values after the phone has been added. In the
General Properties section, fill in the following fields, and leave the remaining fields at their
default values:

Customer Number Select the customer number
Terminal Number Enter a free TN number
Designation Enter a reference name
Zone Enter the zone from Section 3.7
SIP User Name The phone extension number used to

log in at the phone

Node Id The ID of this node
Optional Features: Max Client Count Select the check box
SIPN Set to 1
SIP3 Set to 0

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In the Features section, fill in the following fields, leaving the remaining fields at their defaults.
Note that only the first two feature settings are shown in the screen below; the scroll bar must be
used to display and set the remaining features, which are not shown here.

Call Party Name Display (CNDA) Allowed
Call Number Information (CNIA) Allowed
Restricted Conference or Transfer (FTTC)) Unrestricted Conf. or Transfer
Media Security Encryption (MSEC) Media Security Never (MSNV)
Station Control Password (SCPW) Enter password used to log in at the

phone

Trunk Group Access Restriction (TGAR) Set appropriately
Instrument Type (TYPE) UEXT
Universal Extension User (UTXY) SIPL

In the Keys section, fill in the following:

Key No. 0 SCR – Single Call Ringing
Directory Number Phone extension number
Multiple Appearance Redirection Prime (MARP) Select the Checkbox
First Name Enter a name
Last Name Enter a name
Key No. 1 HOT_U – Hotline(Universal)
UADN The phone extension prefixed by the

UADN Prefix

4

4

The UADN is used to make and receive calls between the SIP Line Gateway and the Universal Extensions.

However, this key is used only by the SIP Line Gateway (SLG) application. The UADN is not dialed by end users.
It is only used internally between the Call Server and the SIP Line Gateway application. See Section 3.2.

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Click Save (not shown) to save the configuration for this phone.

4. Configure Avaya B179 IP Conference Phone

This section describes how to access the B179 web interface and configure the phone to register
to Avaya Communication Server 1000. It assumes that the telephone has been administered an
IP address either through DCHP or static configuration. Additional configuration details are
provided in [3].

4.1. SIP Registration

In the web browser address field, enter the B179 IP address. The login page will appear as shown
below. Select Admin in the Profile dropdown list and enter the appropriate password.

Click Login, and the main configuration screen appears as shown below, where Status 
Network has been selected and shows the static network configuration that was configured on
the B179 in the sample configuration..

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Figure 7 – B179 Network Configuration Status

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4.2. Configure SIP Signalling Settings

To configure the SIP signalling settings, navigate to Settings  SIP, and fill in the following:

Under Account 1:

Enable account Select the Yes radio button
Account name Meaningful name for account status display on phone screen
User Extension (SIP User Name) of the SIP Line telephone configured
in Section 3.11
Realm Use the default of “*”
Authentication name Extension (SIP User Name) of the SIP Line telephone configured
in Section 3.11
Registrar and Proxy SIP domain configured in the CS 1000
Password The Station Control Password of the SIP Line telephone
configured in Section 3.11
Registration interval Enter a value (1800 was used in the sample configuration)

Under Advanced:

Enable Blind Transfer Select the No radio button5
Outbound proxy Enter the IP address of the CS 1000 Node (see Figure 2), port

5070, and lr (loose routing), as shown

Under Transport:

Protocol Select the TCP or UDP radio button (UDP shown)
Local UDP Port Enter 5060

Click Save.

5

This feature is not yet supported in this configuration

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Figure 8 – B179 SIP Configuration

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To configure the audio codec settings, navigate to Settings  Media, and select the priority for
codec selection. One combination shown below prefers the high fidelity G.722 codec if the other
party’s telephone can support it, with a fallback to G.711. Defaults can be used for the
remaining fields. Note that call transfer by CS 1000 telephones is not supported for G.722.

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Another combination that could be used in low bandwidth environments would be to give high
preference to the compressed G.729A codec, with fallback to G.711, as shown below. In the case
of G.729A, ensure that the VAD setting matches that configured in the CS 1000. In this case,
checking Enable VAD results in G.729AB. After choosing an appropriate codec preference list,
click Save. Note also that for this release of the Avaya B179, G.711 is required to be in the
codec list with G.729 in order for call hold by CS 1000 telephones to operate correctly.

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Figure 9 – B179 Codec Selection

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After the configuration has been saved, the B179 will register with the CS 1000, and a display
similar to those shown in Figures 10 and 11 below will appear on the telephone. The

Hostname (see Figure 7) is displayed at the center, and in the lower left corner is the Account
name (see Figure 8). To the left of the Account name is a square icon that indicates the SIP

registration status of the B179. If the square is filled in (Figure 10), the B179 has successfully
registered. If the square is not filled in (Figure 11), registration was unsuccessful.

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Figure 10 – Successful B179 Registration

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Figure 11 – Unsuccessful B179 Registration

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5. Verification Scenarios

Verification scenarios for the configuration described in these Application Notes included:

 Registration and recovery, including power cycling and network disruption.
 SIP signaling using UDP and TCP transport
 Basic calling among the B179 and the following CS 1000 supported telephones:

o 1120e UNIStim
o 1165e SIP
o 2007 UNIStim
o 2050PC UNIStim (soft phone)
o M3903 Digital

 RFC 2833 DTMF support
 G.711mu-law, G.722, G.729A, and G.729AB audio codec support.
 Hold, consultative hold.
 Manual conference by the B179.
 Unattended transfer.
 Placement of calls via the outbound call log.

The following restrictions to the above features apply:

 Attended call transfer of the B179 by CS 1000 supported telephones is supported, except for

M3900 series digital telephones and when G.722 codec is used.. Attended call transfer by
the B179 is not supported.

 Call hold by CS 1000 supported telephones is supported for G.729 only if G.711 is included

in the B179 codec list.

 Calls from the B179 via the inbound call log are not supported.
 Group conference by the B179 is not supported.

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6. Verification Steps

This section provides tests that can be performed to verify proper configuration of the CS 1000
and B179.

6.1. Verify Avaya Communication Server 1000

6.1.1. Verify D-Channel Status

Verify status of the SIP trunk and SIP Line D-Channels by navigating to System 
Maintenance, selecting Select by Overlay, LD 96 – D-Channel, and D-Channel Diagnostics.

The screen below shows the APPL_STATUS of the SIP trunk D-Channel as “OPER” and the
LINK_STATUS as “EST ACTV”. Note that for the SIP line D-Channel, the APPL_STATUS

is “DSBL” and the LINK_STATUS is “RST”. This is normal.

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6.1.2. Verify SIP Registration Status

In the Element Manager web interface, navigate to System  IP Network  Maintenance and
Reports on the left pane. Click GEN CMD.

The General Commands page is displayed. From the Group drop-down menu select SipLine,
from the Command drop-down menu select slgSetShowByUID, enter the B179 extension in
UserID, and click on RUN. The output shown indicates successful registration and displays
details of the registration parameters. Note that if the B179 has not registered, the error message
“Invalid userId 57010” will be returned instead of the detailed registration information.

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6.2. Verify Avaya B179 SIP Conference Phone

Successful registration of the phone can be verified by inspecting the status icon to the left of the
Account name, shown at the lower left of the telephone display. See Figures 10 and 11 for
examples of successful and unsuccessful registration. Registration and call tracing can be
performed on the B179 by navigating to Status  Log. Select SIP Trace on the left and click
Change. Ensure that the On radio button of the SIP logging field is selected. After attempting
registration, click Refresh to see the result. The log can be cleared at any time by clicking Clear
Log. The screen below shows the REGISTER message sent by the B179 for a successful
registration to the CS 1000.

7. Conclusion

As illustrated in these Application Notes, Avaya Communication Server 1000 and the Avaya
B179 SIP Conference Phone can be used together in an integrated solution supporting the
features described in Section 5.

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8. Additional References

Product documentation for Avaya products may be found at http://support.avaya.com.

[1] Communication Server 1000 - Element Manager System Reference –
Administration, Release: 7.5, Document Revision: 05.04, Document #NN43001-632.

[2] Communication Server 1000 SIP Line Fundamentals, Release 7.0, Document
#NN43001-508, 02.03, August 2010.

[3] Installation and Administration of B179, 110047-61-001, Rev 3d.

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©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 [e-mail]

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