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.
- For additional information on the Avaya Communication Manager 7.1, please visit https://downloads.avaya.com/css/P8/documents/101038598
- For additional information on the Ribbon SBC, please visit https://doc.rbbn.com/display/UXDOC80/Ribbon+SBC+Edge+Documentation
- For additional information on Ribbon, please visit https://ribboncommunications.com/
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 | V8.0.0 |
Third-party Equipment | Avaya Communication Manager Avaya System Manager | CM 7.1.2.0.0.532.24184 7.1 |
| OS |
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Other software |
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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.
- Global Support Assistance Center +1-978-614-8589 or +1-888-391-3434 (English language Support)
- Web: https://ribboncommunications.com/services/ribbon-support-portal-login
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.
Avaya Communication Manager Configuration
The following new configurations are included in this section:
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:
- Port configuration
- Media List
- SIP Profile
- Sip Server Table
- Signaling Groups
- Call Routing
- Remote Authorization Tables
- Message Manipulation
- 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 Profiles. Click 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 -> Transformation. Click 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 use. In 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
| 1100 | SIP Connectivity | ||
| ID. No | Procedure | Result | Comment |
|---|---|---|---|
| 1101 | Validate syntax of OPTIONS messages sent to service provider. | Pass | |
| 1102 | Validate syntax of OPTIONS messages sent from service provider. | Pass | |
| 1103 | Validate in service response codes to OPTIONS messages from provider. | Pass | |
| 1104 | Validate in service response codes to OPTIONS messages to provider. | Pass | |
| 1105 | Validate OPTIONS messages are not sent more than once every 10 seconds to provider. | Pass | |
| 2100 | Initial Calls To/From External Phones | ||
| 2101 | Inbound 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 | |
| 2102 | Inbound 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 | |
| 2103 | Outbound 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 | |
| 2104 | Outbound 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 | |
| 2105 | Trunk Group Selection: test absence of explicit trunk group selection. | Pass | |
| 2106 | Trunk Group Selection: test trunk group selection with tgrp tag. | Pass | |
| 2107 | Trunk Group Selection: test trunk group selection with otg tag. | Pass | |
| 3100 | Incomplete Call Attempts | ||
| 3101 | Inbound call from an external phone to an enterprise extension. Hang-up before far-end answers. | Pass | |
| 3102 | Outbound call from an enterprise extension to an external phone. Hang-up before far-end answers. | Pass | |
| 3103 | No Answer of inbound call from an external phone to an enterprise extension. (No explicit rules on Customer Premise Equipment (CPE). Let extension ring.) | Pass | |
| 3104 | No Answer of outbound call from an enterprise extension to an external phone. | Pass | |
| 3105 | Inbound call from an external phone to an enterprise extension that is “Busy”. | Pass | |
| 3106 | Outbound call from an enterprise extension to an external phone that is “Busy”. | Pass | |
| 3107 | Inbound call from an external phone to an unassigned enterprise extension. | Pass | |
| 3108 | Outbound call from an enterprise extension to an invalid external number (Note that this also happens to test CPE support for early media). | Pass | |
| 3109 | Validation of explicit treatments/terminating responses to basic conditions (busy, no circuit avail, bldn etc). | Pass | |
| 4100 | Codec Support and Negotiation with Hard Phones | ||
| 4101 | Whenever the CPE sends out SDP, the Content-Type must be "application/sdp". | Pass | |
| 4102 | Validate inbound G.729 calls. | Pass | Call will still work even if this case fails. |
| 4103 | Validate outbound G.729 calls (annexb=no is required). | Pass | |
| 5100 | Voicemail and DTMF Tone Support | ||
| 5101 | Inbound call from an external phone to an enterprise extension, transfer to voicemail. Leave a message. | Pass | |
| 5102 | Inbound call from an external phone to an enterprise extension, let ring for close to 2 minutes, then transfer to voicemail. Leave a message. | Pass | |
| 5103 | Login to enterprise voicemail and retrieve message from 5102. | Pass | |
| 5104 | Outbound call to an external number, transfer to voicemail. (Ex. Call office or cell phone with voicemail). Leave a message. | Pass | |
| 5105 | Login to external voicemail and retrieve message from 5104. | Pass | |
| 5106 | Test sending a fax (T.30 over G.711, up to 14.4 kbps - V.17). | Pass | |
| 5107 | Test receiving a fax (T.30 over G.711, up to 14.4 kbps - V.17). | Pass | |
| 5108 | RFC2833 DTMF sent from the CPE outbound to an external device are recognized by the receiving equipment. | Pass | |
| 5109 | RFC2833 DTMF sent from an external device inbound to the CPE are recognized by the receiving equipment. | Pass | |
| 5110 | Inband (Q.24) DTMF sent from the CPE outbound to an external device are recognized by the receiving equipment. | Pass | Call from the PSTN was simulated using an external device. |
| 5111 | Inband (Q.24) DTMF sent from an external device inbound to the CPE are recognized by the receiving equipment. | Pass | |
| 5112 | Test sending a fax (T.38 over UDPTL, 14.4 kbps - V.17). | Pass | |
| 5113 | Test receiving a fax (T.38 over UDPTL, 14.4 kbps - V.17). | Pass | |
| 5114 | Test sending a fax (T.38 over UDPTL, 14.4 kbps - V.17) - T.38 in initial INVITE. | Not Run | Optional, as not all vendor/customer supports this feature. |
| 5115 | Test receiving a fax (T.38 over UDPTL, 14.4 kbps - V.17) - T.38 in initial INVITE. | Not Run | Optional, as not all vendor/customer supports this feature. |
| 5116 | Test sending a T.38 fax (T.38 over UDPTL, 14.4 kbps - V.17) - originating from T.38 sender. | Pass | Optional, as not all vendor/customer supports this feature Vendor originates audio call, and has to send reINVITE with T.38. Fax is transcoded. |
| 5117 | Test receiving a T.38 fax (T.38 over UDPTL, 14.4 kbps - V.17) - originating from T.30 sender. | Pass | Bell sends T.30 Fax, vendor sends reINVITE with T.38. Fax is transcoded. |
| 6100 | PSTN Numbering Plans | ||
| 6101 | It was not defined in the Test Plan shared by the customer. | Not Run | |
| 6102 | Outbound Toll-Free Call. | Pass | |
| 6103 | Outbound Local Call. | Pass | |
| 6104 | Outbound International Calls (011)961-865-0650. | Pass | must be number in the test case. |
| 6105 | Operator call (0). | Pass | |
| 6106 | Operator Assisted Calls (e.g. 0+10 digits in US). | Pass | |
| 6107 | Validation of e.164 handling on DID. | Pass | |
| 6108 | It was not defined in the Test Plan shared by the customer. | Not Run | |
| 6109 | Operator Assisted International Call (e.g. 0+1 8 to 35 digits). | Pass | |
| 6110 | Casual Dial: 101+xxxx+NDC call (from 13 to 40 digits). | Pass | |
| 6111 | n11 call (e.g. 211). | Pass | Take a call trace for each n11 call. |
| 6112 | 911 call. | Pass | |
| 6113 | 1-xxx-555-1212 call. | Pass | E.g. 1-613-555-1212. |
| 6114 | 310-xxxx call. | Pass | E.g. 310-2355 (310-BELL). |
| 6115 | 1-700-xxx-xxxx call. | Pass | E.g. 700-555-4141. |
| 6116 | (Optional) 1-900-xxx-xxxx call. | Pass | |
| 6117 | (Optional) 1-976 -xxx-xxxx call. | Not Run | We were not able to find a number available on this segment to call it up. |
| 6118 | Operator-assisted long-distance call (00). | Pass | |
| 7100 | Static ONND | ||
| 7101 | Outbound call with Static ONND - using only the From header and a pre-provisioned number (with user=phone). | Pass | From: sip:[10-digit-caller-number]@[pbx-customer-domain];user=phone. |
| 7102 | Outbound call with Static ONND - using the P-Asserted-Identify header and a pre-provisioned number (with user=phone). | Pass | P-Asserted-Identity: sip:[10-digit-caller-number]@[pbx-customer-domain];user=phone. |
| 7103 | Outbound 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]>". |
| 7104 | Outbound call with Static ONND - using the Diversion header without PAI (with user=phone). | Pass | Diversion: sip:[10-digit-redirector-or-transferer-number]@[pbx-customer-domain];user=phone. |
| 7105 | Outbound call with Static ONND - using the Diversion header (valid Bell number) with PAI (with user=phone). | Pass | Diversion: sip:[10-digit-redirector-or-transferer-number]@[pbx-customer-domain];user=phone. |
| 7106 | Outbound call with Static ONND - using the Diversion header (external number) with PAI (with user=phone and implicit trunk group selection). | Pass | Diversion: sip:[10-digit-redirector-or-transferer-number]@[pbx-customer-domain];user=phone. |
| 7107 | Outbound 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]>". |
| 7108 | Validate proper syntax used in PAI, PPI, From and Diversion for CNAM/CLID display on outbound calls. | Pass | |
| 7200 | Dynamic ONND | ||
| 7201 | Outbound 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]>". |
| 7202 | Outbound 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]>". |
| 7203 | Outbound 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>". |
| 7204 | Outbound 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>". |
| 7205 | Outbound 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>". |
| 7206 | Outbound 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>". |
| 7207 | Outbound call with Dynamic ONND to party A, transfer via tromboning to party B. | Pass | Blind transfer or Attended Transfer is OK (same as test case 8306 or 8311). |
| 7208 | Outbound call with Dynamic ONND to party A, transfer via REFER to party B. | Not Run | PBX Call offloading. (same as Test case 8303). |
| 7209 | Validate proper syntax used in PAI, PPI, From and Diversion for CNAM/CLID display on outbound calls. | Pass | |
| 7300 | Private 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. |
| 8100 | Supplementary Features – Call Hold | ||
| 8101 | Inbound Call – PBX Hold and Resume (No music) – Short Hold Duration. | Pass | Validate directionality of rtp flow is defined/honoured during hold and resume function. |
| 8102 | Inbound Call – PBX Hold and Resume (With music) – Short Hold Duration. | Pass | Validate directionality of rtp flow is defined/honoured during hold and resume function. |
| 8103 | Outbound Call – PBX Hold and Resume No music) – Short Hold Duration. | Pass | Validate directionality of rtp flow is defined/honoured during hold and resume function. |
| 8104 | Outbound Call – PBX Hold and Resume (With music) – Short Hold Duration. | Pass | Validate directionality of rtp flow is defined/honoured during hold and resume function. |
| 8105 | Inbound Call – PSTN Hold and Resume (No music) – Short Hold Duration. | Pass | Validate directionality of rtp flow is defined/honoured during hold and resume function. |
| 8106 | Inbound Call – PSTN Hold and Resume (With music) – Short Hold Duration. | Pass | Validate directionality of rtp flow is defined/honoured during hold and resume function. |
| 8107 | Outbound Call – PSTN Hold and Resume (No music) – Short Hold Duration. | Pass | Validate directionality of rtp flow is defined/honoured during hold and resume function. |
| 8108 | Outbound Call – PSTN Hold and Resume (With music) – Short Hold Duration. | Pass | Validate 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). | Pass | Validate 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). | Pass | Validate 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). | Pass | Validate 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). | Pass | Validate 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). | Pass | Validate 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). | Pass | Validate 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). | Pass | Validate 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). | Pass | Validate directionality of rtp flow is defined/honoured during hold and resume function. |
| 8200 | Supplementary Features – Call Forward (CFD) | ||
| 8201 | Call Forwarding (All) to External Number (Off-net) - 302. | Not Run | Validate 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 Run | Validate 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. | Pass | Validate 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 Run | Validate 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 Run | Validate 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. | Pass | Validate 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 Run | Validate 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 Run | Validate 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. | Pass | Validate diversion and rpid headers are properly formatted on CFD calls. Validate CNAM/CLID display on outbound when call is forwarded. |
| 8300 | Supplementary 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 | |
| 9100 | Failover | ||
| 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. |
| 1100 | Miscellaneous | ||
| 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). | Pass | Validated during 11102-11103 or 8109-8116 tests. |
| 1107 | Validate handling of CPE-initiated session audits. | Pass | Validated 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
Overview
Content Tools