Avaya GXV3240, GXV3275 Application Notes

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These Application Notes describe the configuration steps required to integrate Grandstream GXV3240 and GXV3275 Multimedia IP Phones for Android™ with Avaya Aura® Session Manager and Avaya Aura® Communication Manager.

Readers should pay attention to Section 2, in particular the scope of testing as outlined in Section 2.1 as well as the observations noted in Section 2.2, to ensure that their own use cases are adequately covered by this scope and results.

Information in these Application Notes has been obtained through DevConnect compliance testing and additional technical discussions. Testing was conducted via the DevConnect Program at the Avaya Solution and Interoperability Test Lab.

Avaya Solution & Interoperability Test Lab

Application Notes for Configuring Grandstream GXV3240 and GXV3275 Multimedia IP Phones for Android™ with Avaya Aura® Session Manager and Avaya Aura® Communication Manager - Issue 1.0

Abstract

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1. Introduction

2. General Test Approach and Test Results

The interoperability compliance testing included both feature and serviceability testing.

The featured testing focused on verifying the ability of the Grandstream Multimedia IP Phones to register as SIP endpoints with Avaya Aura® Session Manager, establish voice and point-to-point video calls, and exercise various telephony features (e.g. hold/resume, transfer, conference, etc.).

The serviceability testing focused on verifying the ability of the Grandstream Multimedia IP Phones to handle various outages such as network disconnects and server/phone reboots.

DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The jointly-defined test plan focuses on exercising APIs and/or standards-based interfaces pertinent to the interoperability of the tested products and their functionalities. DevConnect Compliance Testing is not intended to substitute full product performance or feature testing performed by DevConnect members, nor is it to be construed as an endorsement by Avaya of the suitability or completeness of a DevConnect member’s solution.

2.1. Interoperability Compliance Testing

Interoperability compliance testing covered the following features and functionality:

 Successful registration of the Grandstream Multimedia IP Phones with Session Manager.  Voice calls between Grandstream Multimedia IP Phones, Avaya 6200 Series analog

telephone, Avaya 2400 Series digital telephone, Avaya 1600 Series IP Deskphones, Avaya 9600 Series IP Deskphones (96x1 models, both H.323 and SIP), Avaya E129 SIP Deskphones, Avaya E169 Media Station, Avaya one-X® Communicator (both H.323 and SIP), and Avaya Communicator (for Windows, Android, and iPad).

 Point-to-Point video calls between the Grandstream Multimedia IP Phones, Avaya one-

X® Communicator (SIP and H.323), and Avaya Communicator (for Windows and iPad).

 G.711MU, G.711.A, G.729A, G.722-64k, and G.726A.32K codec support.  Guest login/logout  Caller ID and display updates.  Direct IP-IP Media (i.e. media shuffling).  Proper recognition of DTMF tones by navigating voicemail menus.  Proper operation of voicemail with Message Waiting Indication.  Telephony features including: Mute, Hold/Resume, Transfer, Conference, Music on

Hold, Call Coverage Paths, Call Forwarding (Unconditional, Busy, and No Answer), Call Park/Answer Back, Call Pickup, and Automatic Redial.

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The serviceability testing focused on verifying the ability of the Grandstream phones to handle various outages such as network disconnects and server/phone reboots.

The wireless functionality of the Grandstream phones was not tested.

2.2. Test Results

The Grandstream Multimedia IP Phones successfully passed compliance testing with the following observations:

 G.726-32K: G.726-32K codec negotiation failed during compliance testing; however,

Grandstream has delivered a fix for this issue in their system firmware version 1.0.3.25.

 Conference URI / display updates: When the conference is established, the endpoints

involved in the conference receive a SIP UPDATE message with a Contact header containing the “isfocus” feature tag. This feature tag indicates that the URI in the Contact header field is a conference URI. Avaya phones update their display to show the conference URI display information (e.g. Conference 2) which indicates the number of other parties in the conference. The Grandstream phones did not update their display to show the conference URI display information; however, Grandstream has delivered a fix for this issue in their system firmware version 1.0.3.25.

 Long Hold Recall: The Grandstream phones do not audibly or visually alert the user of

a held call when the Avaya Aura® Communication Manager long hold recall timer expires.

 481 Call Leg/Transaction Does Not Exist: Multiple “481 Call Leg/Transaction Does

Not Exist” SIP messages are generated for transfer/conference scenarios. This is

essentially a race condition. For example, after the REFER for a transfer is sent, both parties send a BYE for the call leg going away. When each party receives the BYE, it responds with a 481 Call Leg/Transaction Does Not Exist (since each party has already sent its own BYE for that call leg).

2.3. Support

Grandstream can be reached using the information provided on the following web sites:

 General Contact Information: http://www.grandstream.com/company/contact-us  Support Requests: http://esupport.grandstream.com/support/customerportal/login.php

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3. Reference Configuration

Figure 1 illustrates the test configuration used to verify Grandstream Multimedia IP Phones integration with Avaya Aura® Session Manager and Avaya Aura® Communication Manager. The configuration consists of an Avaya Aura® Communication Manager Server with an Avaya G450 Media Gateway providing connectivity to the PSTN via an ISDN-PRI trunk, Avaya Aura® System Manager, and Avaya Aura® Session Manager. Avaya Aura® Messaging was used as the voicemail system. The Grandstream endpoints registered with Session Manager as SIP Users.

Figure 1: Grandstream Multimedia IP Phones

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4. Equipment and Software Validated

Equipment/Software

Release/Version

Avaya Aura® Communication Manager with an Avaya G450 Media Gateway

R6.3 Update 22098 Avaya Aura® System Manager

R6.3.11

Avaya Aura® Session Manager

R6.3.11

Avaya Aura® Messaging

R6.3

Avaya 6200 Series Analog Phones

Avaya 2400 Series Digital Telephones

Avaya 1600 Series IP Deskphones

1.3.6

Avaya E129 SIP Deskphones

1.0

Avaya E169 Media Station

1.1

Avaya 9600 Series IP Deskphone (96x1 – SIP and H.323)

6.4

Avaya one-X® Communicator (SIP and H.323)

6.2.5

Avaya Communicator (for Windows, Android, and iPad)

2.1 (Windows and Android)

2.0.2 (iPad)

Grandstream GXV3240 and GXV3275 Multimedia IP Phones

Hardware Revision: V1.7A (GXV3240) V1.4B (GXV3275) System Version: 1.0.3.15 Android Version: 4.2.2

The following equipment and software were used for the sample configuration provided:

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5. Configure Avaya Aura® Communication Manager

These Application Notes assume that basic Communication Manager administration has already been performed, including the configuration needed to establish a SIP trunk to Session Manager. Consult Reference [1] for further details if necessary.

All configuration in this section is administered using the System Access Terminal (SAT).

5.1. Configure Video Parameters

Enable IP Video on the signaling group for the SIP trunk between Communication Manager and Session Manager. Enter the “change signaling-group” command. Set IP Video to y. By default, Direct IP-IP Audio Connections is enabled for audio media shuffling.

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Set the multimedia parameters for the ip-codec-set used by the endpoints. Enter the “change ip- codec-set” command. Set Allow Direct-IP Multimedia to y. Set the two call rate parameters to desired values (2048 was used during compliance testing).

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5.2. Verify Station Configuration

It is assumed all Avaya endpoints have already been configured. The stations used for the Grandstream endpoints do not need to be configured via the SAT. Rather, stations will be automatically created in Communication Manager when the SIP Users are added in Section 6.1. After completing the steps in Section 6.1 to create a SIP User, enter the “display station” command within the SAT to verify that station was successfully added to Communication Manager.

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6. Configure Avaya Aura® Session Manager

Configuration of Avaya Aura® Session Manager is performed via Avaya Aura® System Manager. These Application Notes assume that basic System Manager and Session Manager administration has already been performed, including the configuration needed to establish a SIP trunk to Communication Manager. Consult Reference [2] for further details if necessary.

Access the System Manager Administration web interface by entering https://<ip-

address>/SMGR as the URL in a web browser, where <ip-address> is the IP address of System

Manager. Log in using appropriate credentials.

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Once logged in, the following screen is displayed.

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6.1. Configure Users

This document assumes all Users for Avaya SIP endpoints have already been provisioned in System Manager.

Add a User for each Grandstream phone. Navigate to Home  Users  User Management  Manage Users. Click the New button.

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On the Identity tab, enter descriptive values for Last Name and First Name. Enter the Login Name (e.g. user@sipdomain; this document assumes the SIP domain was previously configured. Consult Reference [2] for further details if necessary).

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Click the Communication Profile tab. Enter the Communication Profile Password, and then enter it again in Confirm Password.

Click the New button under Communication Address. Select Avaya SIP for Type. Enter the Full Qualified Address for the user and then click the Add button.

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Check the box next to Session Manager Profile. For Primary Session Manager, select the Session Manager SIP Entity from the drop-down menu.

Under Application Sequences, select the sequence (e.g. cm10217) used to route calls to Communication Manager from the drop-down menu for the Origination Sequence and Termination Sequence.

Under Call Routing Settings, select the Home Location (e.g. Lab) from the drop-down menu.

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Check the box next to CM Endpoint Profile. Select the Communication Manager SIP Entity from the System drop-down menu. Select Endpoint for Profile Type. Enter the Communication Manager station Extension (e.g. 53108). From the Template drop-down menu, select 9641SIP_DEFAULT_CM_6_3. Enter the desired value for the station Security Code. For Voice Mail Number, enter the Avaya Aura® Messaging pilot number. Click the Endpoint Editor button next to Extension field.

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Under the General Options tab, set Coverage Path 1 to the coverage path used for Avaya Aura® Messaging.

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Click the Feature Options tab. Check the box for IP Video and then click the Done button to return to the user Communication Profile tab as shown on the next page. Click the Commit button.

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7. Configure Grandstream GXV3240 and GXV3275 Multimedia IP Phones for Android™

It is assumed that the basic configuration for the Grandstream phones has already been implemented and that the phones are ready for the integration with Avaya Aura® Session Manager. The sub-sections below provide only the steps required to configure the Grandstream to interoperate with Avaya.

Note: Only the configuration steps required to integrate the Grandstream GXV3240 phone model with Avaya are shown in the sub-sections below; however the same steps are required and applicable to the Grandstream GXV3275 phone model.

7.1. Web Interface

Access the phone web interface by opening a web browser and entering the following URL:

http://ip-address, where ip-address is the IP address of the phone. Log in using appropriate

credentials.

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After making and saving any configuration changes throughout the remainder of Section 7, an Apply button will appear at the top of the web interface. Click the Apply button (not shown) to immediately apply the changes.

7.2. Configure General Settings

Click the Account tab and select an Account sub-tab to configure (e.g. Account 1). Click General Settings on the left. Configure the following parameters:

 Account Active: check the Yes box.  Account Name: Enter descriptive text.  SIP Server: Enter the Session Manager signaling IP address.  SIP User ID: Enter the SIP User created in Section 6.1.  SIP Authentication ID: Enter the SIP User created in Section 6.1.  SIP Authentication Password: Enter the password for the User created in Section 6.1.  Voice Mail UserID: Enter the Avaya Aura® Messaging pilot number. Configuration of

Avaya Aura® Messaging is outside the scope of this document. The mailboxes for the Grandstream phones are configured the same and Avaya phones).

 Name: Enter descriptive text (optional)

Use the default values for the remaining fields. Click the Save button at the bottom of the screen.

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7.3. Configure SIP Settings

Click the SIP Settings tab on the left. Check the Yes box for SUBSCRIBE for MWI. Use the default values for the remaining fields. Click the Save button at the bottom of the screen.

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7.4. Configure Code Settings

Click the Codec Settings tab on the left. The settings on the screen below show that values that were used during compliance testing; however, Preferred Vocoder settings were modified occasionally during the testing to test various codecs. After making any changes, click the Save button at the bottom of the screen.

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7.5. Configure Packetization Mode

By default, the Grandstream phones support packetization mode set to 1. At the time of testing, the Avaya video endpoints did not support packetization mode set to 1. Therefore, the packetization mode value on the Grandstream phones was changed to 0.

To change the packetization mode value on the Grandstream phone, open a secure shell session to the phone’s IP address. Log in using appropriate credentials. The commands required to change the value (and output) are shown below.

GXV3240 > config

CONFIG > get 957

957=1

CONFIG > set 957 0

Set 957=0

CONFIG> commit

nvram commited

CONFIG > exit

GXV3240 > reboot

You have chosen to reboot.

Do you want to continue (y/N)? y

8. Verification Steps

This section includes steps that can be followed to verify the configuration.

8.1. Verify Point to Point Audio and Video Calls

Place point to point audio and video calls between the Grandstream phones and Avaya endpoints. Verify 2-way audio as well as 2-way video for the video calls.

9. Conclusion

These Application Notes describe the procedures required for the Grandstream Multimedia IP Phones to interoperate with Avaya Aura® Session Manager and Avaya Aura® Communication Manager to support the reference configuration shown in Figure 1. Refer to Section 2.2 for testing results and any observations noted during testing.

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10. Additional References

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

[1] Administering Avaya Aura® Communication Manager, Release 6.3, Issue 10, June 2014. [2] Administering Avaya Aura® Session Manager, Release 6.3, Issue 7, September 2014. [3] Administering Avaya Aura® Messaging, Release 6.3.2, Issue 1, December 2014.

Product documentation for Grandstream products may be found at:

http://www.grandstream.com.

[4] GXV3240 IP Multimedia Phone for Android™ Administration Guide [5] GXV3275 IP Multimedia Phone for Android™ Administration Guide

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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 DevConnect Program at [email protected].