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Avaya Solution & Interoperability Test Lab

Application Notes for Avaya Aura® Communication

Manager 6.0.1, Avaya Aura® Session Manager 6.1, and

Avaya Aura® Session Border Controller 6.0 with Frontier

Communications SIP Trunking – Issue 1.1

Abstract

These Application Notes describe the steps to configure Session Initiation Protocol (SIP)

Trunking between Frontier Communications SIP Trunking and an Avaya SIP-enabled

enterprise solution. The Avaya solution consists of Avaya Aura® Session Border Controller,

Avaya Aura® Session Manager, Avaya Aura® Communication Manager Evolution Server,

and various Avaya endpoints. This documented solution does not extend to configurations

without the Avaya Aura® Session Border Controller or Avaya Aura® Session Manager.

Frontier Communications is a member of the Avaya DevConnect Service Provider program.

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.

DDT; Reviewed:

SPOC 2/7/2012

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Table of Contents

Introduction ............................................................................................................................. 4

General Test Approach and Test Results ................................................................................ 4

2.1. Interoperability Compliance Testing ................................................................................ 4

2.2. Test Results ...................................................................................................................... 5

2.3. Support ............................................................................................................................. 6

3. Reference Configuration ......................................................................................................... 6

4. Equipment and Software Validated ........................................................................................ 8

5. Configure Avaya Aura® Communication Manager ............................................................... 9

5.1. Licensing and Capacity .................................................................................................... 9

5.2. System Features.............................................................................................................. 10

5.3. IP Node Names............................................................................................................... 11

5.4. Codecs ............................................................................................................................ 11

5.5. IP Network Region ......................................................................................................... 12

5.6. Signaling Group ............................................................................................................. 13

5.7. Trunk Group ................................................................................................................... 15

5.8. Inbound Routing............................................................................................................. 17

5.9. Calling Party Information............................................................................................... 18

5.10.

Outbound Routing ...................................................................................................... 19

5.11.

Saving Communication Manager Configuration Changes ......................................... 22

6. Configure Avaya Aura® Session Manager .......................................................................... 23

6.1. System Manager Login and Navigation ......................................................................... 24

6.2. Specify SIP Domain ....................................................................................................... 25

6.3. Add Location .................................................................................................................. 26

6.4. Add SIP Entities ............................................................................................................. 28

6.5. Add Entity Links ............................................................................................................ 32

6.6. Add Routing Policies ..................................................................................................... 33

6.7. Add Dial Patterns ........................................................................................................... 34

6.8. Verify Avaya Aura® Session Manager Instance ........................................................... 37

7. Configure Avaya Aura ® Session Border Controller ........................................................... 39

7.1. Installation Wizard ......................................................................................................... 39

7.1.1. Network Settings ..................................................................................................... 40

7.1.2. Logins ..................................................................................................................... 41

7.1.3. VPN Access ............................................................................................................ 42

7.1.4. SBC ......................................................................................................................... 43

7.1.5. Confirm Installation ................................................................................................ 45

7.2. Post Installation Configuration....................................................................................... 46

7.2.1. Options Frequency .................................................................................................. 47

7.2.2. Blocked Headers ..................................................................................................... 49

7.2.3. Diversion Header .................................................................................................... 51

7.2.4. P-Asserted Identity URI .......................................................................................... 56

7.2.5. From URI ................................................................................................................ 63

DDT; Reviewed:

SPOC 2/7/2012

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7.2.6. Save the Configuration ........................................................................................... 65

8. Configure Frontier Communications SIP Trunking ............................................................. 65

9. Verification /Troubleshooting Steps ..................................................................................... 65

9.1. Verification..................................................................................................................... 65

9.2. Troubleshooting ............................................................................................................. 67

10. Conclusion ......................................................................................................................... 68

11. References .......................................................................................................................... 68

12. Appendix A: Avaya Aura® SBC Configuration File ........................................................ 69

DDT; Reviewed:

SPOC 2/7/2012

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1. Introduction

These Application Notes describe the steps to configure Session Initiation Protocol (SIP)

Trunking between Frontier Communications SIP Trunking and an Avaya SIP-enabled enterprise

solution. The Avaya solution consists of Avaya Aura® Session Border Controller, Avaya

Aura® Session Manager and Avaya Aura® Communication Manager Evolution Server along

with various Avaya endpoints.

Customers using this Avaya SIP-enabled enterprise solution with Frontier Communications SIP

Trunking are able to place and receive PSTN calls via a broadband WAN connection and the SIP

protocol. This converged network solution is an alternative to traditional PSTN trunks such as

ISDN-PRI.

2. General Test Approach and Test Results

The general test approach was to configure a simulated enterprise site using Communication

Manager, Session Manager and the Session Border Controller to connect to the public Internet

using a broadband connection. The enterprise site was configured to connect to Frontier

Communications SIP Trunking. This configuration (shown in

Figure 1)

was used to exercise the

features and functionality listed in

Section 2.1.

Frontier Communications SIP Trunking passed compliance testing.

2.1. Interoperability Compliance Testing

To verify SIP Trunking interoperability, the following features and functionality were covered

during the interoperability compliance test:



Incoming PSTN calls to various Avaya phone types. Phone types included H.323, SIP,

digital, and analog telephones at the enterprise. All inbound PSTN calls were routed to

the enterprise across the SIP trunk from the service provider.

Outgoing PSTN calls from various Avaya phone types. Phone types included H.323, SIP,

digital, and analog telephones at the enterprise. All outbound PSTN calls were routed

from the enterprise across the SIP trunk to the service provider.

Inbound and outbound PSTN calls to/from Avaya one-X Communicator (soft client).

Avaya one-X Communicator supports two modes (Road Warrior and Telecommuter).

Each supported mode was tested. Avaya one-X Communicator also supports two Voice

Over IP (VoIP) protocols: H.323 and SIP. Each supported protocol was tested.

Various call types including: local, long distance, international, outbound toll-free,

operator assisted calls and local directory assistance (411).

Codecs G.729A, G.711MU and G.711A.

G.711MU Fax.

DTMF transmission using RFC 2833.

Caller ID presentation and Caller ID restriction.

Voicemail navigation for inbound and outbound calls.

User features such as hold and resume, transfer, and conference.

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DDT; Reviewed:

SPOC 2/7/2012



Off-net call forwarding and mobility (extension to cellular).

Items not supported or not tested included the following:



Inbound toll-free international and emergency calls (911) are supported but were not

tested as part of the compliance test.



Operator services (0) and operator assisted calls (0 + 10 digits) were not supported in the

Frontier Communications test environment.



T.38 Fax is not supported.



Network Call Redirection using the SIP REFER method or a 302 response.

2.2. Test Results

Interoperability testing of Frontier Communications SIP Trunking was completed with

successful results for all test cases with the exception of the observations/limitations described

below.



Calling Party Number (PSTN transfers):

The calling party number displayed on the

PSTN phone is not updated to reflect the true connected party on calls that are

transferred to the PSTN. After the call transfer is complete, the calling party number

displays the number of the transferring party and not the actual connected party. The

PSTN phone display is ultimately controlled by the PSTN provider, thus this behavior is

not necessarily indicative of a limitation of the combined Avaya/Frontier

Communications SIP Trunk solution. It is listed here simply as an observation.

No Support for G.729B:

Frontier Communications SIP Trunk does not support G.729B

codec.

SIP Network REFER to an off-net extension is not supported:

When a

Communication Manager vector is programmed to redirect an inbound call to a PSTN

number after the call is answered by an announcement, Frontier Communications SIP

Trunk will send a“403 Forbidden” to the REFER SIP message and the call will drop.

Frontier Communications SIP Trunk does not support REFER messages.

SIP 302 Redirect Method is not supported:

When a Communication Manager vector is

programmed to redirect an inbound call to a PSTN number before answering the call in

the vector with Network Call Redirection activated, Frontier Communications SIP Trunk

will send an ACK to the “302 Moved Temporarily” SIP message from the enterprise but

will not redirect the call to the new party in the Contact header of the 302 message. The

inbound call initiator hears a busy signal in this failure scenario.



DDT; Reviewed:

SPOC 2/7/2012

Solution & Interoperability Test Lab Application Notes

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