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Table of Contents

 

Document Overview

This document provides a configuration guide for the Ribbon SBC Edge Series (Session Border Controller) when connecting to the Avaya CM.

This configuration guide supports features given in the Avaya Communication Manager 7.1 configuration guide.

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound calls flows between the Ribbon SBC Edge and the Avaya CM.

Audience

This is a technical document intended for telecommunications engineers with the purpose of configuring both the Ribbon SBC and the third-party product. Some steps will require navigating the third-party product  as well as the Ribbon SBC. 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 Ribbon 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.

Requirements

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

Requirements

 

Equipment

Software Version

Ribbon Networks

Ribbon SBC Edge 2000
Build Number

V8.0.0
502

Third-party Equipment

Avaya Communication Manager

Avaya System Manager

CM 7.1.2.0.0.532.24184

7.1

OS

 

 

Other software

 

 



Reference Configuration

The following reference configuration shows connectivity between the third-party product and the Ribbon SBC Edge.

Reference Configuration

Support

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

 Technical support for Ribbon SBC Edge series is available by phone or logging a trouble ticket.

 

Third-Party Product Features

Third-Party Product Features

 

  • Features
    • Initial Calls To/From External Phones
    • Incomplete Call Attempts
    • Voicemail and DTMF Tone Support
    • PSTN Numbering Plans
    • Private and Unknown Calls
    • Call Hold & Call Park
    • Call Forwarding & Call Transfer
    • Long Duration Calls

 

Prerequisites

Fax & Voicemail systems are necessary to execute some of the tests. 

Having knowledge of Regular Expressions is useful to configure the transformation tables. Consider reviewing this topic before setting them. See Appendix A for further information regarding Regular Expressions.

 

Verify License

The DS1 Ports feature is needed to work with PRI trunks.

 

1. ISDN-PRI Configuration

a. Determine the physical location for the PRI

Use the hardware configuration report to determine the Board Number and Board Type that will be used to set the DS1 circuit.

Boards

b. Create a DS1

Create the DS1 circuit using the location from the previous step. Select the correct option for the Signaling Mode, Line Coding and Framing Mode

DS1

c. Create a Signaling Group

Specify the Primary D-Channel, the Group Type and the Group Number.

Signaling Group

 

d. Create a Trunk Group

Provide the Group Name, Group Number, Direction, Group type and TAC number.

Trunk Group

 

2. Public Unknown Numbering Configuration

Add the numbers in the public-unknown-numbering table. These entries are used to specify desired digits for name and number display on display-equipped stations in an ISDN network.

Unknown Numbering

3. Dial plan Analysis Configuration

The Dial Plan Analysis Table is the system’s guide to translating the digits dialed by users. It determines the beginning digits and total length for each type of call that Communication Manager needs to interpret.

Configure the Dialplan Analysis table according to your needs.

Dialplan Analysis

4. Feature Access Code Configuration

Feature access codes (FACs) activate or cancel system features when dialed.

Configure the Feature access codes according to your needs.

Feature Access Code

5. ARS analysis

These entries define the parameters that the Automatic Route Selection (ARS) table uses to route a call.

Configure the ars analysis according to your needs.

ARS

6. Route Pattern Configuration

Each route pattern contains a list of trunk groups that the Communication Manager uses to route a call.

Configure the route-pattern according to your needs.

Route Pattern

7. Special Applications

SA8052 - ISDN Redirecting Number.

This feature creates a redirecting number IE, or original called number IE, in the SETUP message for redirecting a call offnet over an ISDN trunk to either a Bellcore or Nortel switch. A redirection in this case is either a call forward or call coverage. This feature is needed to notify these other switches that the call is a redirection and not a direct termination. This is especially useful if the voice mail system is attached to one of these off net switches in order to receive the correct greeting.

ISDN Redirecting Number.


SA8983 - Replace CPN when calls forwarded off-net.

This Special Application SA8983 modifies the calling party number for outgoing ISDN trunk calls that are the result of either call forwarding to an external number or the EC500 feature. With this feature active, the calling party number for these calls will be that of the call forwarding or EC500 station rather than that of the call originator. This feature works for both station-to-station and for incoming trunk calls to the call forwarding or EC500 station. All types of call forwarding – unconditional, busy/don’t answer, and enhanced call forwarding – are supported.

Replace CPN when calls forwarded off-net.


SA8146 - Display Update for Redirected Calls.

This feature enables an update of the originator’s display for calls that are redirected. The originator’s display will be updated to contain the principal station’s name, the redirection reason, and the name and extension of the “redirected to” station for intraswitch redirections.

For interswitch redirections, connected via ISDN-PRI, the originator’s display will be updated with the name and number of the “redirected to” station and the principal station’s name if the principal is on the same switch.

Display Update for Redirected Calls.

Ribbon SBC Edge Configuration

The following new configurations are included in this section:

  1. Port configuration
    1. T1 Port
    2. Ethernet Port
  2. Media List
  3. SIP Profile
  4. Sip Server Table
  5. Signaling Groups
    1. ISDN Signaling Group
    2. SIP Signaling Group
  6. Call Routing
    1. Transformation Table
    2. Call Routing Table
  7. Remote Authorization Tables
  8. Message Manipulation
  9. Modify the Signaling Groups

1. Port configuration

a. T1 Port

The Telephony Port settings allow you to configure the physical properties of the telephony trunks (T1/E1) and analog ports (FXS). Digital trunk type selection (T1 or E1, ISDN or CAS) and analog line profile selection are done here.

Go to Node Interfaces -> Ports and select the (T1) Port you want to configure.

Configure the Data Layer section and click Apply.

T1 Port

b. Ethernet Port

Go to Node Interfaces -> Ports and select the Port Ethernet you want to configure.

Configure the Ethernet Port according to your topology and  click Apply.

Ethernet Port

 

Go to Node Interfaces -> Logical Interfaces. Select the interface you want to configure.

Configure the Interface according to your topology and  click Apply.

Logical Interfaces

 

2. Media List

Media Profiles allow you to specify the individual voice and fax compression codecs and their associated settings, for inclusion in a Media List. Different codecs provide varying levels of compression, allowing one to reduce bandwidth requirements at the expense of voice quality.

Go to Media -> Media Profiles -> Create Media Profile -> Voice Codec Profile. Select the codec and the payload size you want to use.

Media Profiles.

 

Media Lists allow you to specify a set of codecs and fax profiles that are allowed on a given SIP Signaling Group. They contain one or more Media Profiles, which must first be defined in Media Profiles. These lists allow you to accommodate specific transmission requirements, and SIP devices that only implement a subset of the available voice codecs.

Go to Media -> Media Profiles. Click the create () icon to add a new Media List.

Select the Media Profiles you want to add to the Media Profile List.

Media Profiles.

3. SIP Profile

SIP Profiles control how the SBC Edge communicates with SIP devices. They control important characteristics such as session timers, SIP header customization, SIP timers, MIME payloads, and option tags.

Go to SIP -> SIP ProfilesClick the create (icon to add a new SIP Profile Entry.

Configure the SIP Profile according to your topology and  click Apply.

SIP Profile.

 

4. Sip Server Table 

SIP Server Tables contain information about the SIP devices connected to the SBC Edge. The entries in the tables provide information about the IP Addresses, ports, and protocols used to communicate with each server. The Table Entries also contain links to counters that are useful for troubleshooting.

Go to SIP -> SIP Server Tables. Click the create () icon to add a new SIP Server Table.

Type a Description and click OK.

Sip Server Table

 

 

Go to SIP -> SIP Server Tables. Select the SIP Server Table you have just created and click on Create SIP Server.

Configure the SIP Server Entry according to your topology and click OK.

Note.
    The Monitor option specifies the method to monitor server.
  • None: No monitoring of this server occurs.
  • SIP options: An OPTIONS message is sent to the server. When this option is selected, additional configuration items are displayed. These are noted below.

Sip Server Table

5. Signaling Groups

Signaling groups allow telephony channels to be grouped together for the purposes of routing and shared configuration. They are the entity to which calls are routed, as well as the location from which Call Routes are selected. They are also the location from which Tone Tables and Action Sets are selected.

a. ISDN Signaling Group

Go to Signaling Groups -> Create Signaling Group. Click the ISDN Signaling Group icon to create a new Signaling Group.

Configure the Signaling Group according to your topology and  click OK .

ISDN Signaling Group.

 

b. SIP Signaling Group

Go to Signaling Groups -> Create Signaling Group. Click the SIP Signaling Group icon to create a new Signaling Group.

Configure the Signaling Group according to your topology and click OK . We will modify this Signaling Group once the Routing is done in the SBC. 

SIP Signaling Group.

6. Call Routing

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

a. Transformation Table

Transformation Tables facilitate the conversion of names, numbers and other fields when routing a call. They can, for example, 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 they are selected from there. In addition, Transformation tables are configurable as a reusable pool that Action Sets can reference.

Go to Call Routing -> TransformationClick the create () icon to add a new Transformation Table.

Type a Description and  click OK.

Transformation Table.

 

 

Go to Call Routing -> Transformation. Click the name of your transformation table and click the create () icon to add a new Transformation Entry.

Fill in all the fields according to your numbering plan.

Note.

Use a different Transformation Table for every Call Routing Table you want to useIn our case we will use two Transformation Tables (inbound and outbound traffic).

Transformation Entry.

b. Call Routing Table

Go to Call Routing -> Call Routing Table. Click the create () icon to add a new Call Routing Table.

Call Routing Table.

 

 

Go to Call Routing -> Call Routing Table. Click the name of your Call Routing Table and click the create () icon to add a new Call Route Entry.

Fill in the fields in red according to your topology.

Note.

Use a different Call Routing Table for every Signaling Group you want to use. In our case we will use two Call Routing Tables (traffic from the Avaya CM and from Bell Canada).

Call Route Entry.

 

7. Remote Authorization Tables

Remote Authorization Tables and their entries contain information used to respond to request message challenges by an upstream server.

Go to SIP -> Remote Authorization Tables. Click the create () icon to add a new SIP Remote Authorization.

Fill in the fields according to the information shared by the carrier and click Apply.

Remote Authorization Tables.

 

8. Message Manipulation

The SIP Message Manipulation feature is used by a SIP Signaling Group to manipulate the incoming or outgoing messages. This feature is intended to enhance interoperability with different vendor equipment and applications, and for correcting any fixable protocol errors in SIP messages on fly without any changes to firmware/software.

See Appendix A for further information regarding Messages Manipulation

Go to SIP -> Message Manipulation -> Message Rule Tables. Click the create () icon to add a new Message Rule Table.

Fill in the fields in red and  click OK.

Message Rule Table.

 

 

Go to SIP -> Message Manipulation -> Message Rule Tables. Click the name of the Message Rule Table you have just created. Go to Create Rule and Click the Raw Message Rule icon to add a new Message Rule entry. 

Message Rule.

 

 

Fill in the fields in red. This rule replaces the IP address with a domain name. 

Rule 1

 

 

Go to SIP -> Message Manipulation -> Message Rule Tables. Click the name of the Message Rule Table you want to modify and click the Header Rule icon to add a new Message Rule entry. 

This rule modifies the Contact header by replacing the IP address with a domain name.

Rule 2

 

 

Create a new Message Rule entry.

This rule modifies the From header by replacing the IP address with a domain name.

Rule 3

 

 

Create a new Message Rule entry.

This rule modifies the P-Asserted-Identity header by replacing the IP address with a domain name.

Create a new Message Rule entry.

This rule modifies the To header by removing the URI port and the user parameter.

Rule 5

 

 

Create a new Message Rule entry.

This rule modifies the Max-Forwards header by setting the value to 0. It only applies to the OPTIONS messages.

Rule 6

 

9. Modify the Signaling Groups

At this point we have all the necessary items needed to configure our Signaling Groups.

Select the call Routing Table you want to use in your ISDN Signaling Group.

ISDN Signaling Group.

 

Select the call Routing Table, SIP profile, SIP Server Table, Media List ID, Port, Protocol, Federated IPs and Message Table List you want to use in your SIP Signaling Group.

SIP Signaling Group.

Private and Unknown Calls

A Message Manipulation was created to adapt the header format.

Go to SIP -> Message Manipulation -> Message Rule Tables. Click the name of the Message Rule Table you want to modify and click the Header Rule icon to add a new Message Rule entry. 

This rule modifies the From header by replacing the Display Name, the Username and Host as showing in the following picture; it also removes the user parameter.

Assign this Message Rule to the SIP Signaling Group you want to use.

Anonymous rule.


Failover

Terminating ISDN calls return a Q.850 Cause Code when they end. We can use these codes to determine whether or not to reroute the call to another signalling group via additional configured entries in the Call Route Table. A Cause Code Reroute table contains one or more Q.850 Cause Codes which, when matched, trigger a reroute.

See Appendix A for further information regarding Reroute Mapping tables.

Go to Telephone Mapping Tables -> Cause Code Reroute. Click the create () icon to add a new Cause Code Reroute Table.

Cause Code Reroutes.

 

 

Go to Call Routing -> Call Routing Table. Click the Call Routing Table you want to modify. and select the Cause Code Reroute you want to use.

Call Routing Table.

 

Dynamic Outgoing Name & Number Display (ONND)

The SIP Trunk Group Table enables you to configure trunk groups in a profile that can be associated with specific SIP Signaling Groups. The SBC provides interoperability with different servers by transporting, identifying, and processing the trunk group (list of signaling groups) via enterprise trunking parameters. The trunk group helps provides a way to identify/choose the carrier at the gateway for a proxy or service provider.

Go to SIP -> Trunk Groups. Click the create () icon to add a new Trunk Group.

Fill in the fields in red according to the information shared by the carrier and click Apply.

Call Routing Table.


Test Results

Test Results

1100SIP Connectivity
ID. NoProcedureResultComment
1101Validate syntax of OPTIONS messages sent to service provider.Pass 
1102Validate syntax of OPTIONS messages sent from service provider.Pass 
1103Validate in service response codes to OPTIONS messages from provider.Pass 
1104Validate in service response codes to OPTIONS messages to provider.Pass 
1105Validate OPTIONS messages are not sent more than once every 10 seconds to provider.Pass 
2100Initial Calls To/From External Phones
2101Inbound call from an external phone to an enterprise extension. Hang-up at called party (enterprise extension). Wait for calling party to disconnect. Validate proper SIP header syntax, ringback tone, two-way audio and proper call clearance.Pass 
2102Inbound call from an external phone to an enterprise extension. Hang-up at calling party (PSTN phone). Wait for called party to disconnect. Validate proper SIP header syntax, ringback tone, two-way audio and proper call clearance.Pass 
2103Outbound call from an enterprise extension to an external phone. Hang-up at called party (PSTN phone). Wait for calling party to disconnect. Make sure originating party is properly identified (Diversion/History-Info, PAI or From- in that order), using exactly 10 digits for the user part and the domain matching this TN's "PBX" (to which its TG is assigned). Also validate "tgrp/trunk-context" in Contact, if doing explicit TG selection (usually for Toll-bypass). Validate ringback tone, two-way audio and proper call clearance.Pass 
2104Outbound call from an enterprise extension to an external phone. Hang-up at calling party (enterprise extension). Wait for called party to disconnect. Make sure originating party is properly identified (Diversion/History-Info, PAI or From- in that order), using exactly 10 digits for the user part and the domain matching this TN's "PBX" (to which its TG is assigned). Also validate "tgrp/trunk-context" in Contact, if doing explicit TG selection (usually for Toll-bypass). Validate ringback tone, two-way audio and proper call clearance.Pass 
2105Trunk Group Selection: test absence of explicit trunk group selection.Pass 
2106Trunk Group Selection: test trunk group selection with tgrp tag.Pass 
2107Trunk Group Selection: test trunk group selection with otg tag.Pass 
3100Incomplete Call Attempts
3101Inbound call from an external phone to an enterprise extension. Hang-up before far-end answers.Pass 
3102Outbound call from an enterprise extension to an external phone. Hang-up before far-end answers.Pass 
3103No Answer of inbound call from an external phone to an enterprise extension. (No explicit rules on Customer Premise Equipment (CPE). Let extension ring.)Pass 
3104No Answer of outbound call from an enterprise extension to an external phone.Pass 
3105Inbound call from an external phone to an enterprise extension that is “Busy”.Pass 
3106Outbound call from an enterprise extension to an external phone that is “Busy”.Pass 
3107Inbound call from an external phone to an unassigned enterprise extension.Pass 
3108Outbound call from an enterprise extension to an invalid external number (Note that this also happens to test CPE support for early media).Pass 
3109Validation of explicit treatments/terminating responses to basic conditions (busy, no circuit avail, bldn etc).Pass 
4100Codec Support and Negotiation with Hard Phones
4101Whenever the CPE sends out SDP, the Content-Type must be "application/sdp".Pass 
4102Validate inbound G.729 calls.PassCall will still work even if this case fails.
4103Validate outbound G.729 calls (annexb=no is required).Pass 
5100Voicemail and DTMF Tone Support
5101Inbound call from an external phone to an enterprise extension, transfer to voicemail. Leave a message.Pass 
5102Inbound call from an external phone to an enterprise extension, let ring for close to 2 minutes, then transfer to voicemail. Leave a message.Pass 
5103Login to enterprise voicemail and retrieve message from 5102.Pass 
5104Outbound call to an external number, transfer to voicemail. (Ex. Call office or cell phone with voicemail). Leave a message.Pass 
5105Login to external voicemail and retrieve message from 5104.Pass 
5106Test sending a fax (T.30 over G.711, up to 14.4 kbps - V.17).Pass 
5107Test receiving a fax (T.30 over G.711, up to 14.4 kbps - V.17).Pass 
5108RFC2833 DTMF sent from the CPE outbound to an external device are recognized by the receiving equipment.Pass 
5109RFC2833 DTMF sent from an external device inbound to the CPE are recognized by the receiving equipment.Pass 
5110Inband (Q.24) DTMF sent from the CPE outbound to an external device are recognized by the receiving equipment.PassCall from the PSTN was simulated using an external device.
5111Inband (Q.24) DTMF sent from an external device inbound to the CPE are recognized by the receiving equipment.Pass 
5112Test sending a fax (T.38 over UDPTL, 14.4 kbps - V.17).Pass 
5113Test receiving a fax (T.38 over UDPTL, 14.4 kbps - V.17).Pass 
5114Test sending a fax (T.38 over UDPTL, 14.4 kbps - V.17) - T.38 in initial INVITE.Not RunOptional, as not all vendor/customer supports this feature.
5115Test receiving a fax (T.38 over UDPTL, 14.4 kbps - V.17) - T.38 in initial INVITE.Not RunOptional, as not all vendor/customer supports this feature.
5116Test sending a T.38 fax (T.38 over UDPTL, 14.4 kbps - V.17) - originating from T.38 sender.PassOptional, as not all vendor/customer supports this feature Vendor originates audio call, and has to send reINVITE with T.38. Fax is transcoded.
5117Test receiving a T.38 fax (T.38 over UDPTL, 14.4 kbps - V.17) - originating from T.30 sender.PassBell sends T.30 Fax, vendor sends reINVITE with T.38. Fax is transcoded.
6100 PSTN Numbering Plans
6101It was not defined in the Test Plan shared by the customer.Not Run 
6102Outbound Toll-Free Call.Pass 
6103Outbound Local Call.Pass 
6104Outbound International Calls (011)961-865-0650.Passmust be number in the test case.
6105Operator call (0).Pass 
6106Operator Assisted Calls (e.g. 0+10 digits in US).Pass 
6107Validation of e.164 handling on DID.Pass 
6108It was not defined in the Test Plan shared by the customer.Not Run 
6109Operator Assisted International Call (e.g. 0+1 8 to 35 digits).Pass 
6110Casual Dial: 101+xxxx+NDC call (from 13 to 40 digits).Pass 
6111n11 call (e.g. 211).PassTake a call trace for each n11 call.
6112911 call.Pass 
61131-xxx-555-1212 call.PassE.g. 1-613-555-1212.
6114310-xxxx call.PassE.g. 310-2355 (310-BELL).
61151-700-xxx-xxxx call.PassE.g. 700-555-4141.
6116(Optional) 1-900-xxx-xxxx call.Pass 
6117(Optional) 1-976 -xxx-xxxx call.Not RunWe were not able to find a number available on this segment to call it up.
6118Operator-assisted long-distance call (00).Pass 
7100Static ONND
7101Outbound call with Static ONND - using only the From header and a pre-provisioned number (with user=phone).PassFrom: sip:[10-digit-caller-number]@[pbx-customer-domain];user=phone.
7102Outbound call with Static ONND - using the P-Asserted-Identify header and a pre-provisioned number (with user=phone).PassP-Asserted-Identity: sip:[10-digit-caller-number]@[pbx-customer-domain];user=phone.
7103Outbound call with Static ONND - using explicit trunk group selection (with user=phone).Pass"Contact: <sip:[user-part];tgrp=[trunk-ID];trunk-context=siptrunking.bell.ca@[CPE/PBX-IP-address]> OR From/PAI: <sip:[10-digit-caller-number]@[customer-domain];user=phone;otg=[trunkID]>".
7104Outbound call with Static ONND - using the Diversion header without PAI (with user=phone).PassDiversion: sip:[10-digit-redirector-or-transferer-number]@[pbx-customer-domain];user=phone.
7105Outbound call with Static ONND - using the Diversion header (valid Bell number) with PAI (with user=phone).PassDiversion: sip:[10-digit-redirector-or-transferer-number]@[pbx-customer-domain];user=phone.
7106Outbound call with Static ONND - using the Diversion header (external number) with PAI (with user=phone and implicit trunk group selection).PassDiversion: sip:[10-digit-redirector-or-transferer-number]@[pbx-customer-domain];user=phone.
7107Outbound call with Static ONND - using the Diversion header (external number) with PAI (with user=phone and explicit trunk group selection).Pass"Contact: <sip:[user-part];tgrp=[trunk-ID];trunk-context=siptrunking.bell.ca@[CPE/PBX-IP-address]> OR From/PAI/Diversion: <sip:[10-digit-caller-number]@[customer-domain];user=phone;otg=[trunkID]>".
7108Validate proper syntax used in PAI, PPI, From and Diversion for CNAM/CLID display on outbound calls.Pass 
7200Dynamic ONND
7201Outbound call with Dynamic ONND - using the From header (without user=phone).Pass"Contact: <sip:[user-part];tgrp=[trunk-ID];trunk-context=siptrunking.bell.ca@[CPE/PBX-IP-address]> OR From: <sip:[caller-number]@[customer-domain];otg=[trunkID]>".
7202Outbound call with Dynamic ONND - using the P-Asserted-Identify header (without user=phone).Pass"Contact: <sip:[user-part];tgrp=[trunk-ID];trunk-context=siptrunking.bell.ca@[CPE/PBX-IP-address]> OR From/PAI: <sip:[caller-number]@[customer-domain];otg=[trunkID]>".
7203Outbound call with Dynamic ONND - using the Diversion header (with user=phone ) without PAI and using a valid Bell SIP Trunking number in both the Diversion and From.Pass"Contact: <sip:[user-part];tgrp=[trunk-ID];trunk-context=siptrunking.bell.ca@[CPE/PBX-IP-address]> OR From: <sip:[caller-number]@[customer-domain];otg=[trunkID]> Diversion: <sip:[caller-number]@[customer-domain];otg=[trunkID];user=phone>".
7204Outbound call with Dynamic ONND - using the Diversion header (with user=phone ) without PAI and using an external number in either the Diversion or From.Pass"Contact: <sip:[user-part];tgrp=[trunk-ID];trunk-context=siptrunking.bell.ca@[CPE/PBX-IP-address]> OR From: <sip:[caller-number]@[customer-domain];otg=[trunkID]> Diversion: <sip:[caller-number]@[customer-domain];otg=[trunkID];user=phone>".
7205Outbound call with Dynamic ONND - using the Diversion header (with user=phone) with PAI and using a valid Bell SIP Trunking number in both the Diversion and PAI.Pass"Contact: <sip:[user-part];tgrp=[trunk-ID];trunk-context=siptrunking.bell.ca@[CPE/PBX-IP-address]> OR From/PAI: <sip:[caller-number]@[customer-domain];otg=[trunkID]> Diversion: <sip:[caller-number]@[customer-domain];otg=[trunkID];user=phone>".
7206Outbound call with Dynamic ONND - using the Diversion header (with user=phone) with PAI and using an external number in the Diversion.Pass"Contact: <sip:[user-part];tgrp=[trunk-ID];trunk-context=siptrunking.bell.ca@[CPE/PBX-IP-address]> OR From/PAI: <sip:[caller-number]@[customer-domain];otg=[trunkID]> Diversion: <sip:[External Number]@[customer-domain];otg=[trunkID];user=phone>".
7207Outbound call with Dynamic ONND to party A, transfer via tromboning to party B.PassBlind transfer or Attended Transfer is OK (same as test case 8306 or 8311).
7208Outbound call with Dynamic ONND to party A, transfer via REFER to party B.Not RunPBX Call offloading. (same as Test case 8303).
7209Validate proper syntax used in PAI, PPI, From and Diversion for CNAM/CLID display on outbound calls.Pass 
7300Private and Unknown Calls
7301

Place an outbound private call. Validate privacy header syntax and interworking on outbound private call against Bell spec and document differences.

Pass

 

7302

Place an inbound private call. Validate privacy header syntax and interworking on inbound private call against Bell spec and document differences. CPE must respect the privacy header.

Pass

Requires Bell participation. Anonymous call was simulated using an ingress SMM to modify the FROM header according to the format specified in section 16.2.

7303

Validate handling of incoming unknown calls

Pass

From header is unknown@unknown.invalid Requires Bell participation. Unknown call was simulated using an ingress SMM to modify the FROM header according to the format specified in section 9.1.

7304

Validate handling of incoming calls when not subscribed to Calling Line ID Delivery

Pass

From header is sip:siptrunking.bell.ca Requires Bell participation. An SMM was used to to modify the FROM header according to the format specified in section 9.1.

8100Supplementary Features – Call Hold
8101

Inbound Call – PBX Hold and Resume (No music) – Short Hold Duration.

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8102

Inbound Call – PBX Hold and Resume (With music) – Short Hold Duration.

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8103

Outbound Call – PBX Hold and Resume No music) – Short Hold Duration.

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8104

Outbound Call – PBX Hold and Resume (With music) – Short Hold Duration.

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8105

Inbound Call – PSTN Hold and Resume (No music) – Short Hold Duration.

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8106

Inbound Call – PSTN Hold and Resume (With music) – Short Hold Duration.

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8107

Outbound Call – PSTN Hold and Resume (No music) – Short Hold Duration.

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8108

Outbound Call – PSTN Hold and Resume (With music) – Short Hold Duration.

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8109

Inbound Call - PBX Hold and Resume (No music) – Long Hold Duration that exceeds the SIP session timers (~10 min).

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8110

Inbound Call - PBX Hold and Resume (With music) – Long Hold Duration that exceeds the SIP session timers (~10 min).

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8111

Outbound Call - PBX Hold and Resume (No music) – Long Hold Duration that exceeds the SIP session timers (~10 min).

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8112

Outbound Call - PBX Hold and Resume (With music) – Long Hold Duration that exceeds the SIP session timers (~10 min).

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8113

Inbound Call - PSTN Hold and Resume (No music) – Long Hold Duration that exceeds the SIP session timers (~10 min).

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8114

Inbound Call - PSTN Hold and Resume (With music) – Long Hold Duration that exceeds the SIP session timers (~10 min).

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8115

Outbound Call - PSTN Hold and Resume (No music) – Long Hold Duration that exceeds the SIP session timers (~10 min).

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8116

Outbound Call - PSTN Hold and Resume (With music) – Long Hold Duration that exceeds the SIP session timers (~10 min).

PassValidate directionality of rtp flow is defined/honoured during hold and resume function.
8200Supplementary Features – Call Forward (CFD)
8201

Call Forwarding (All) to External Number (Off-net) - 302.

Not RunValidate diversion and rpid headers are properly formatted on CFD calls.  Validate CNAM/CLID display on outbound when call is forwarded.
8202

Call Forwarding (All) to External Number (Off-net) - Refer.

Not RunValidate diversion and rpid headers are properly formatted on CFD calls.  Validate CNAM/CLID display on outbound when call is forwarded.
8203

Call Forwarding (All) to External Number (Off-net) - Tromboning.

PassValidate diversion and rpid headers are properly formatted on CFD calls.  Validate CNAM/CLID display on outbound when call is forwarded.
8204

Call Forwarding (No Answer) to External Number (Off-net) – 302.

Not RunValidate diversion and rpid headers are properly formatted on CFD calls.  Validate CNAM/CLID display on outbound when call is forwarded.
8205

Call Forwarding (No Answer) to External Number (Off-net) – Refer.

Not RunValidate diversion and rpid headers are properly formatted on CFD calls.  Validate CNAM/CLID display on outbound when call is forwarded.
8206

Call Forwarding (No Answer) to External Number (Off-net) – Tromboning.

PassValidate diversion and rpid headers are properly formatted on CFD calls.  Validate CNAM/CLID display on outbound when call is forwarded.
8207

Call Forwarding (Busy) to External Number (Off-net) – 302.

Not RunValidate diversion and rpid headers are properly formatted on CFD calls.  Validate CNAM/CLID display on outbound when call is forwarded.
8208

Call Forwarding (Busy) to External Number (Off-net) – Refer.

Not RunValidate diversion and rpid headers are properly formatted on CFD calls.  Validate CNAM/CLID display on outbound when call is forwarded.
8209

Call Forwarding (Busy) to External Number (Off-net) – Tromboning.

PassValidate diversion and rpid headers are properly formatted on CFD calls.  Validate CNAM/CLID display on outbound when call is forwarded.
8300Supplementary Features – Call Transfer, Conference
8301

Blind Call Transfer of inbound call: Transfer to External Number (Refer).

Not Run 
8302

Blind Call Transfer of inbound call: Transfer to External Number (Tromboning).

Pass 
8303

Blind Call Transfer of inbound call: Transfer to Internal Number (Refer).

Not Run 
8304

Blind Call Transfer of inbound call: Transfer to Internal Number (Tromboning).

Pass 
8305

Blind Call Transfer of outbound call: Transfer to External Number (Refer).

Not Run 
8306

Blind Call Transfer of outbound call: Transfer to External Number (Tromboning).

Pass 
8307

Blind Call Transfer of outbound call: Transfer to Internal Number (Refer).

Not Run 
8308

Blind Call Transfer of outbound call: Transfer to Internal Number (Tromboning).

Pass 
8309

Attended Transfer of inbound call: Transfer to External Number (Tromboning).

Pass 
8310

Attended Transfer of inbound call: Transfer to Internal Number (Tromboning).

Pass 
8311

Attended Transfer of outbound call: Transfer to External Number (Tromboning).

Pass 
8312

Attended Transfer of outbound call: Transfer to Internal Number (Tromboning).

Pass 
8313

Validate call park and unpark.

Pass 
8314

Attended Transfer of inbound call: Transfer to External Number (Refer).

Not Run 
8315

Attended Transfer of inbound call: Transfer to Internal Number (Refer).

Not Run 
8316

Attended Transfer of outbound call: Transfer to External Number (Refer).

Not Run 
8317

Attended Transfer of outbound call: Transfer to Internal Number (Refer).

Not Run 
9100Failover
9101

Validate handling of ICMP unreachable messages on a new call, by pointing CPE primary IP to unreachable IP.

Not Run

Destination Unreachable PSTN failover. The SBC only reroute on cause codes.

9102

Validate handling of bell SBC silently discarding packets on a new call, by pointing to 207.236.202.114:50505.

Pass

Geo redundancy. We are using SIP OPTIONS to know the status of the far end; if there is no answer, the SBC assumes that the peer is down and try using the second route to forward the call.

9103

Validate handling of SIP 503 responses on a new call, by pointing to 207.236.202.114:50503.

Pass

Geo redundancy.

9104

Validate Handling of out service response codes to OPTIONS pings, out of service codes are anything other then 200 and 483 by pointing to 207.236.202.114:50504.

Pass

Geo redundancy. It was tested using an external device, it is responding to the SIP OPTIONS with 503 Service Unavailable.

9105

Validate traffic to CPE from multiple Bell IPs in order to simulate SBC failover. Requires Bell participation.

Pass

Geo redundancy. We simulated inbound calls from different IP addresses using private IP segment. Calls are completed at the Avaya handsets.

9106

(Optional) Validate failover between multiple CPEs.

Not Run

Inbound failover. We only have one CPE in the current topology.

1100Miscellaneous
1101

Validate handling of multiple concurrent calls for the same number.

Pass 
1102

Long Duration Calls - Inbound.

Pass 
1103

Long Duration Calls - Outbound.

Pass 
1104

Outgoing call with wrong DID number or wrong PBX domain.

Pass 
1105

(Optional) Validate handling of outbound call to full TG (403 Forbidden).

Pass 
1106

Validate handling of session audits every 5 or 10 min (UPDATE or re-INVITE).

PassValidated during 11102-11103 or 8109-8116 tests.
1107

Validate handling of CPE-initiated session audits.

PassValidated during 11102-11103 or 8109-8116 tests.

Conclusion

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

Appendix A

Understanding Regular Expressions

Regular Expressions for Number Matching and Transformation

Q.850 to SIP Reroute Mapping Table

Adding and Modifying Entries to Message Rule Tables

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