Page tree
Skip to end of metadata
Go to start of metadata


Table of Contents


Document Overview

This document provides a configuration guide for the Sonus Session Border Controller (SBC) 1000/2000 Series when connecting to Avaya 6.3.

This configuration guide supports features given in Avaya configuration guide.


The interoperability compliance testing focuses on verifying inbound and outbound calls flows between Sonus SBC 1000/2000 and Avaya 6.3.


This is a technical document intended for telecommunications engineers with the purpose of configuring both the Sonus SBC and the third-party product. There will be steps that require navigating third-party as well as the Sonus SBC Command Line Interface (CLI). Understanding the basic concepts of TCP/UDP, IP/Routing, and SIP/RTP are also necessary to complete the configuration and for troubleshooting, if necessary.

This configuration guide is offered as a convenience to Sonus customers. The specifications and information regarding the product in this guide are subject to change without notice. All statements, information, and recommendations in this guide are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided “AS IS”. Users must take full responsibility for the application of the specifications and information in this guide.


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

Table : Requirements



Software Version

Sonus Networks

SBC 2000


Tenor AFM200P108-09-26

Third-party Equipment


Avaya CM & SM 6.3
Avaya 9608 Phone


VentaFax I

Reference Configuration

The following reference configuration shows connectivity between third-party and Sonus SBC 1000/2000.

Figure : Connectivity Between Third-Party and Sonus SBC 1000


For any questions regarding this document or the content herein, please contact your maintenance and support provider.


Third-Party Product Features

The testing was executed with the British Telecom test plan. The following features were tested:

  • Basic originated and terminated calls 
  • Basic inbound/outbound call 
  • Basic inbound/outbound call with privacy 
  • Hold and resume* 
  • Call Transfer (Blind transfer)
  • Call Transfer (Consult transfer)
  • Call Forwarding Unconditional 
  • Call Pickup 
  • Hunt group
  • Conference call 
  • Long calls
  • FAX 
  • DTMF

*British Telecom SBC will handle the call hold response gracefully (expects the call to be on hold) and passes it on to the next hop, however we cannot guarantee that all destination networks/devices will handle the response the same way.

Verify License

SIP Calls 



Avaya 6.3 Configuration 

The following steps provide an example of how to configure Avaya 6.3:

1. Trunk Group

Run add trunk-group next to add new trunk.

Figure : Trunk Group

2. DS1

Run list configuration trunks to list all available boards.

Figure : List Configuration Trunks

Run change ds1 001V2 to change the settings accordingly.

Figure : DS1


3. Signaling Group

Run add signaling-group next to add new signaling group.

Figure : Signaling Group


4. Route Pattern

Run change route-pattern (number) to add/change the route pattern. 

Figure : Route Pattern


5. ARS Analysis Entry

Run change ars analysis (dialed number) to add/change the called number handling.

Figure : ARS Analysis Entry


6. Station

Run add station next to add new station. 

Figure : Station


Sonus SBC 1000/2000 Configuration

The following steps provide an example of how to configure Sonus SBC 1000/2000:

1. SIP Profile

Select Settings > SIP > SIP Profiles

SIP Profiles control how the Sonus SBC 1000/2000 communicates with SIP devices. These control important characteristics such as session timers, SIP Header customization, SIP timers, MIME payloads, and option tags. The following figure shows the default SIP Profile used for the SBC 1000/2000 for this testing effort:

Figure : SIP Profiles


2. SIP Server

Select Settings > SIP > SIP Server Tables

SIP Server Tables contain information about the SIP devices connected to the Sonus SBC 1000/2000. The entries in the tables provide information about the IP addresses, ports, and protocols used to communicate with each SIP server. The entries also contain links to counters that are useful for troubleshooting.

Figure : BT1


Figure : BT2


Figure : Fax


3. Media Profile

Select Settings > Media > Media Profiles

Media Profiles specify the individual voice and fax compression codecs and their associated settings for inclusion into a Media List. Different codecs provide varying levels of compression, allowing the reduction of bandwidth requirements at the expense of voice quality. The following figures are the media profiles of the voice codecs used for the SBC 1000/2000 in this testing effort and are shown for reference only:

Figure : Voice Codec G711 A-Law


Figure : Voice Codec G711 U-Law


Figure : Voice Codec G729


Figure : Fax Codec T.38


4. Media List

Select Settings > Media > Media List

The Media List shows the selected voice and fax compression codecs and their associated settings.

Figure : Media Lists

5. Transformation Table

Select Settings > Transformation

Transformation Tables facilitate the conversion of names, numbers and other fields when routing a call. For example, Transformation Tables can convert a public PSTN number into a private extension number, or into a SIP address (URI). Every entry in a Call Routing Table requires a Transformation Table, and are sequentially selected from the Transformation Table. In addition, Transformation Tables are configurable as a reusable pool that Action Sets can reference.

Figure : From Avaya 6 ISDN 1


Figure : From Avaya 6 ISDN 2


Figure : From BT 1

Figure : From BT 2

Figure : From BT to Fax

6. Cause Code Reroutes

Select Settings > Telephony Mapping Tables > Cause Code Reroutes

Terminating ISDN calls return a Q.850 Cause Code when they end. These codes can determine whether or not to reroute the call to another signalling group. A Cause Code Reroutes table contains one or more Q.850 Cause Codes that, when matched, trigger a reroute. To use a Cause Code Reroutes table, go to Call Routing Table, select create or modify, and then from a drop down menu, select the appropriate Cause Code Reroutes table.

Figure : Cause Code Reroutes


7. Call Routing Table

Select Settings > Call Routing Table

Call Routing allows calls to be carried between signaling groups, thus allowing calls to be carried between ports, and between protocols (like ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible configuration of what calls will be carried, and how the calls are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message translations, Cause Code Reroutes, Media Lists and the three types of Signaling Groups (ISDN, SIP and CAS).

Figure : From Avaya 6 ISDN

Figure : From BT 1


Figure : From BT 2

8. Tone Table

Select Settings > Tone Tables

Tone tables allow the Sonus SBC 1000/2000 Administrator to customize the tones a user hears when placing a call. You can modify the tone to match your local PSTN or PBX. The default tone table is configured for the values used in the United States for the following categories: Ringback, Dial, Busy, Congestion, Call Waiting, Disconnect, and Confirmation.

Figure : Tone Table

9. Node Interfaces

Select Settings > Node Interfaces > Ports

This section describes how to configure the DS1 port types (T1/E1) on the Sonus SBC 1000/2000 system. Using this feature sets the DS1 ports on the SBC 1000/2000 to either T1 or E1. In most geographical areas only T1 or E1 is available. However, in some countries both are available and the SBC may be required to support both simultaneously. 

Figure : Ports


10. Signaling Groups

Select Settings > Signaling Groups

Signaling Groups allow telephony channels to be grouped together for the purposes of routing and shared configuration. These groups are the entity to which calls are routed, as well as the location from which Call Routes are selected. These are also the location from which Tone Tables and Action Sets are selected. In the case of SIP, this will specify protocol settings and link to server, media, and mapping tables.

Figure : Internal Side

Figure : External Side




Test Results

Table : Test Results

1Incoming call Pass 
2Outgoing call Pass 
3Call forward Pass 
4Call forward to PSTN Pass 
5Ad-hoc conferencing Pass 
6Blind transfer Pass 
7Consultative transfer Pass 
8Call hold Pass 
9Call pickup group Pass 
10Hunt group Pass 
11RFC 3261 compliance Pass 
12IP transport protocols Pass 
13User-agent header Pass 
14SIP proxy failover Pass 
15SIP 183 - Early Media Pass 
16Cease INVITE retransmission Pass 
17141 & 1470 Service Prefixes Pass 
18"Withheld" incoming number Pass 
19SIP URI Pass 
20Media negotiation rejection Pass 
21RFC 3550 compliance Pass 
22Default packetisation times Pass 
23RTP clock-source accuracy Pass 
24Packet fragmentation Pass 
25Jitter estimator Pass 
26RFC 2833 support Pass 
27RFC 2833 support from compliant devices Pass 
28RFC 4733 recommendation Pass 
29Platform originated call call-party formatting Pass 
30CPE originated call call-party formatting Pass 
31CPE originated calls to service numbers Pass 
32T.38 fax relay faxing Pass 
33G.711 fax pass-through faxing Pass 
34G.711 fax up-speed faxing Pass 
35Keep alive behaviour Pass 
36Message re-transmission behaviour Pass 
37Keep-alive timers Pass 
38Alerting and supervisory tones Pass 
39Busy endpoint behaviour Pass 
40Trunk oversubscribed response Pass 
41Call-hold reject Pass 
42RFC 3264 compliance Pass 
43Call-hold black-holing Pass 
44G.729a Support Pass 
45G.711 Packet Loss Concealment Pass 



These Application Notes describe the configuration steps required for Sonus SBC 1000/2000 to successfully interoperate with Avaya 6.3. All feature and serviceability test cases were completed and passed with the exceptions/observations noted in Test Results.