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Document Overview

This document provides a configuration guide for Ribbon EdgeMarc 6000 when connecting to the Avaya Communication Manager (CM) 7.1.

This configuration guide supports features in the Virgin Media SIP Trunk Application.

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound call flows between the Ribbon EdgeMarc 6000 and the Avaya CM 7.1 platform.

Audience

This is a technical document intended for telecommunications engineers for configuring both the Ribbon SBCs and the third-party product. Users will perform steps to navigate the third-party product as well as the Ribbon SBC Command Line Interface (CLI). Understanding the basic concepts of TCP/UDP/TLS, IP/Routing, and SIP/RTP is also necessary for completing the configuration and for troubleshooting, if necessary.

Note

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:

Table : Requirements


Equipment

Software Version

Ribbon Communications

Ribbon EdgeMarc 6000

V16.0.0

Third-party Equipment



Avaya Communication Manager

7.1

Avaya one-X Communicator6.2
NGT Litev.1.51

Reference Configuration

The following reference configuration shows the connectivity between the third-party and the Ribbon EdgeMarc 6000.

Figure : Reference Configuration

Support

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

Third-Party Product Features

Ribbon supports the following third-party product features:

  • Basic originated and terminated calls
  • Basic inbound and outbound calls
  • Hold and Resume
  • Call Forwarding
  • DTMF
  • Conference calls
  • Action on eSBC outage (restart of eSBC)
  • Action on Loss of Virgin Media primary SBC

Configure Avaya Communication Manager 7.1

The following new configurations are included in this section:

  1. DS1 Circuit Pack
  2. Signaling Group
  3. Trunk Group
  4. Route Pattern
  5. ARS Digit Analysis Table
  6. Station
  7. Change public-unknown-numbering

1. DS1 Circuit Pack

  1. Use the Site Administration to log into the Avaya CM 7.1.
  2. Type change ds1 001V5 in the command line.
  3. Configure the Signaling Mode field with isdn-pri.
  4. Configure the Connect field with pbx.
  5. Configure the Interface field with user.

Figure : DS1 Circuit Pack



2. Signaling Group

  1. Use the Site Administration to log into the Avaya CM 7.1.
  2. Type add signaling-group next in the command line. 

    Note

    The next switch auto-generates the next available group number for the Signaling Group, which is the most efficient method for creating a new Signaling Group.

  3. Confirm the next available Signaling Group information and press F3 to save the changes. 

Figure : Signaling Group

3. Trunk Group

  1. Use the Site Administration to log into the Avaya CM 7.1.
  2. Type add trunk-group next in the command line.
  3. Enter the trunk group information and press F3 to save the changes.

Figure : Trunk Group


4. Route Pattern

  1. Use the Site Administration to log into the Avaya CM 7.1.
  2. Type list route-pattern in the command line to determine the next available route pattern.
  3. Identify the route-pattern number you are going to use, and then press F1 to exit the current operation.
  4. Type change route-pattern and then type the available route pattern number. Press F3 to save the changes.

Figure : Route Pattern


5. ARS Digit Analysis Table

  1. Use the Site Administration to log into the Avaya CM 7.1.
  2. Type change ars analysis <dialed_number> to add or change the handling for a specified called number.
  3. Confirm the changes and press F3 to save.

Figure : ARS Digit Analysis Table


6.  Station

  1. Use the Site Administration to log into the Avaya CM 7.1.
  2. Type add station next to add a new station. 
  3. Confirm the changes and press F3 to save.

Figure : Station

7.  Change public-unknown-numbering

  1. Use the Site Administration to log into the Avaya CM 7.1.
  2. Type change public-unknown-numbering 0 and add the caller number station. With this command, the caller number will appear for outgoing calls in an ISDN interconnection.
  3. Confirm the changes and press F3 to save.

Figure : Change public-unknown-numbering


EdgeMarc Configuration

Network

VoIP


Network

LAN and WAN Interfaces

  1. Log into the EdgeMarc as a root user.
  2. Click Network to configure the LAN and WAN interfaces.

Figure : EdgeMarc Network LAN Interface

 

Figure : EdgeMarc Network WAN Interface

Static Routes

Select Network > Static Routes to configure the routes.

Figure : Static Routes

T1/E1 Configuration

Select Network > T1/E1 Configuration to configure the T1.

Figure : T1/E1 Configuration

ISDN - Network

Select Network > ISDN > Network to configure the Network Side ISDN PRI configuration.

Figure : ISDN - Network

VoIP

VoIP Settings

  1. Login as a root user.
  2. Click VoIP to configure the VoIP features.

Figure : VoIP

SIP Settings

  1. Select VoIP > SIP to configure the SIP settings.
  2. Configure the SIP servers.

Figure : SIP Settings

 

B2BUA 

  1. Select VoIP > B2BUA to access the B2BUA trunking configuration.
  2. Configure the next parameters.

Figure : B2BUA

Survivability

  1. Select VoIP > Survivability.
  2. Configure the parameters.

Figure : Survivability

Test Results

The following table provides Ribbon's test results of the scenarios that Virgin Media requires for its customers.

S.NoProcedureObservationResultComment
IOP1

Vendor's eSBC response to SIP OPTIONS messages from SBC

No calls are required for this test.

Capture the SIP trace for approximately 60 seconds and check for correct signaling.

For each eSBC, the SBC periodically sends an OPTIONS request to the vendor's eSBC to check if its SIP stack is reachable. If the IP-PBX sends a SIP response 200 OK, the SIP trunk is placed or remains in an In-Service state.

Example:

OPTIONS sip:ping@<ip-pbx_IP_Addr>:5060 SIP/2.0

Pass
IOP2SBC response to SIP OPTIONS messages from vendor eSBC

No calls are required for this test.

Capture SIP trace for approximately 60 seconds (depending on agreement) and check for correct signaling.

Vendor's eSBC Setup for Solution IP.Addr Mode

  1. The eSBC is configured to send OPTIONS messages to the SBC periodically.
  2. The SBC responds with SIP response 200 OK.

Example:
"OPTIONS sip:ping@192.168.1.10:5060 SIP/2.0"

  • Verify that the eSBC can simultaneously send SIP OPTIONS messages to both solution SBC addresses.
Pass
IOP4Basic test call from IP-PBX to PSTN line through SBC-A (using SBC-A IPV4 IP address)
  1. The IP-PBX line initiates the call. 
  2. The call is answered.
  3. The IP-PBX line terminates the call.

Vendor's eSBC Setup for Solution IP.Addr Mode

A call progresses successfully when:

  • A call is received from the IP-PBX.
  • An Invite is seen from the eSBC to SBC-A.
  • Proxy authentication challenge is returned to the eSBC.
  • A Re-invite with correct credentials is received from the eSBC. 

Example:
Request-Line: INVITE sip:<B-party>@<SBC-A ip.addr TBD>:5060 SIP/2.0
To: sip:<B-Party>@<SBC-A ip.addr TBD>

  • Check the Wireshark trace to confirm that the G.711 A law codec with 10 or 20ms packetization is used.
  • Verify that the INVITE contains the Session-Expires header, and the INVITE is syntactically correct.
  • Check the Supported Header to ensure that it supports the timer. Ensure that the response in the 200 OK is compatible with the INVITE. Also, verify that the Required Header contains the timer.
Pass
IOP5

Basic test call from IP-PBX to PSTN line through SBC-B (using SBC-B IPV4 ip address)

Vendor to configure eSBC so that it used secondary SBC (SBC_B) for this test

Once the test completes, configure eSBC to use Primary SBC-A for calls to route to.

  1. The IP-PBX line initiates the call. 
  2. The call is answered.
  3. The IP-PBX line terminates the call.

Vendor's eSBC Setup for Solution IP.Addr Mode

A call progresses successfully when:

  • A call is received from the IP-PBX.
  • An Invite is seen from the eSBC to SBC-B.
  • Proxy authentication challenge is returned to the eSBC.
  • A re-invite with correct credentials is received from the eSBC.

Example:
Request-Line: INVITE sip:<B-party>@<SBC-B ip.addr TBD>:5060 SIP/2.0
To: sip:<B-Party>@<SBC-B ip.addr TBD>

Check the Wireshark trace to confirm that the G.711 A law codec with 10ms or 20ms packetization is used.

Pass
IOP7b

Called Number format - vendor's eSBC to soft switch number normalization - Global Dial Plan

Test eSBC capability to send the called number in one of the following Global number formats (user part of Request & To URIs):

0yyyyyyyyyy (where y refers to any number, calling party = national)
+44yyyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)

Configure the SBC for the Global calling plan.

  1. The IP-PBX line initiates a call to the PSTN line.
  2. The call is answered.
  3. The IP-PBX line terminates the call.
  4. Configure the eSBC to present the called number in the user part of the Request & To URIs and send it in one of the following formats:
    • 0yyyyyyyyyy (where y refers to any number, calling party = national)
    • +44yyyyyyyyyy (where y refers to any number, calling party = national)
    • +yyyyyyyyyy (where y refers to any number, calling party = international)
    • yyyyyyyyyy (where y refers to any number, calling party = unknown)
Pass
IOP8b

Calling Number format - vendor's eSBC to soft switch number normalization - Global Dial Plan

Test eSBC capability to send the calling number in one of the following Global number formats (user part of From & PAI URIs):

0yyyyyyyyyy (where y refers to any number, calling party = national)
+44yyyyyyyyyy (where y refers to any number, calling party = national)
00yyyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)

Configure the SBC for the Global calling plan.

  1. The IP-PBX line initiates a call to the PSTN line.
  2. The call is answered.
  3. The IP-PBX terminates the call.
  4. Configure the eSBC to present the calling number in the user part of the From & PAI URIs and send it in one of the following formats:
    • 0yyyyyyyyyy (where y refers to any number, calling party = national)
    • +44yyyyyyyyyy (where y refers to any number, calling party = national)
    • 00yyyyyyyyyy (where y refers to any number, calling party = international)
    • yyyyyyyyyy (where y refers to any number, calling party = unknown)
Pass

IOP9b

Called Number format - soft switch to eSBC number normalization - Global Dial Plan

Test eSBC capability of accepting the called number in one of the following Global number formats (user part of Request & To URIs):

+44yyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)

Configure the SBC for the Global calling plan.

  1. The PSTN line initiates a call to the IP-PBX line. 
  2. The call is answered.
  3. The PSTN line terminates the call.
  4. Configure the eSBC to accept the called number in the user part of the Request & To URIs in one of the following formats:
    • +44yyyyyyyyy (where y refers to any number, calling party = national)
    • +yyyyyyyyy (where y refers to any number, calling party = international)
    • yyyyyyyyyy (where y refers to any number, calling party = unknown)
  5. Verify that the INVITE contains the Session-Expires header and the INVITE is syntactically correct.
  6. Check the Supported Header to ensure that it supports the timer. Ensure that the response in the 200 OK is compatible with the INVITE. Also, verify that the Required Header contains the timer.
Pass
IOP10b

Calling Number format - soft switch to eSBC number normalization - Global Dial Plan

Test eSBC capability of accepting the calling number in one of the following Global number formats (user part of From & PAI URIs):

+44yyyyyyyyy (where y refers to any number, calling party = national)
+yyyyyyyyy (where y refers to any number, calling party = international)
yyyyyyyyyy (where y refers to any number, calling party = unknown)

Configure the SBC for the Global calling plan.

  1. The PSTN line initiates a call to the IP-PBX line.
  2. The call is answered.
  3. The PSTN line terminates the call.
  4. Configure the eSBC to accept the calling number in the user part of the Request & To URIs in one of the following formats:
    • +44yyyyyyyyy (where y refers to any number, calling party = national)
    • +yyyyyyyyy (where y refers to any number, calling party = international)
    • yyyyyyyyyy (where y refers to any number, calling party = unknown)
Pass
IOP11Emergency Call Handling -IP-PBX Line to PSTN - UK Emergency call 999
  1. Make a call from the IP-PBX line to the Emergency services using 999.
  2. The call is answered.
  3. Either party terminates the call.

Example:
Request-Line: INVITE sip:999@<SBC-A ip.addr TBD>:5060 SIP/2.0
To: <sip:999@<SBC-A ip.addr TBD>>
From: <sip:<A-party>@<IP-PBX IP.Addr>

Pass
IOP12

Emergency Call Handling -IP-PBX Line to PSTN - UK Emergency call 112

  1. Make a call from the IP-PBX line to the Emergency services using 112.
  2. The call is answered.
  3. Either party terminates the call.

Example:
Request-Line: INVITE sip:112@<SBC-A ip.addr TBD>:5060 SIP/2.0
To: <sip:112@<SBC-A ip.addr TBD>>
From: <sip:<A-party>@<IP-PBX IP.Addr>

Pass
IOP13

Emergency Call Handling -IP-PBX Line to PSTN - UK Emergency call 18000 - Text Direct

  1. Make a call from the IP-PBX line using a text direct set to the Emergency services using 18000. 
  2. The call is answered.
  3. Either party terminates the call.

Example:
Request-Line: INVITE sip:18000@<SBC-A ip.addr TBD>:5060 SIP/2.0
To: <sip:18000@<SBC-A ip.addr TBD>>
From: <sip:<A-party>@<IP-PBX IP.Addr>

Pass


IOP14IP-PBX Line to PSTN - Call answer - Originator disconnect
  1. Make a call from the IP-PBX line to the PSTN line.
  2. Answer the call.
  3. The IP-PBX line terminates the call.
Pass
IOP15PSTN calls SIP #1, SIP #1 conferences in SIP #2
  1. Make a call from the IP-PBX line to the PSTN line.
  2. Answer the call.
  3. The PSTN line terminates the call.
Pass
IOP16IP-PBX Line to PSTN - Busy subscriber
  1. Make a call from the IP-PBX line to a busy PSTN line (without divert on busy).
  2. Wait for the soft switch to return the busy response.
  3. Ensure that the eSBC is not recursive.
  4. Set up the call via the secondary SIP trunk.
Pass
IOP17IP-PBX Line to PSTN - No answer timeout test
  1. Make a call from the IP-PBX line to a PSTN line (without divert on no answer).
  2. Do not answer the call.
  3. Wait for the soft switch to return the no answer timeout response.
  4. Ensure that the eSBC is not recursive. 
  5. Set up the call via the secondary SIP trunk.
Pass with CaveatIt is not possible to modify the timer in Avaya CM 7.1 (3 minutes by default) because we use the terminal Avaya One-X Communicator. At the same time Virgin Media sent a CANCEL. Since we have two CANCELs at the same time, VM sent us 481 Call Leg does not Exist.
IOP18

IP-PBX Line to PSTN - Subscriber not reachable

Vendor to call 01189111111

  1. Make a call from the IP-PBX line to an invalid number.
  2. Wait for the soft switch to return a response.
  3. Ensure that the eSBC is not recursive. 
  4. Set up the call via the secondary SIP trunk.
Pass
IOP19PSTN Line to IP-PBX - Call answer - Originator disconnect
  1. Make a call from a PSTN line to an IP-PBX line.
  2. Answer the call. 
  3. The originator disconnects the call.
Pass.
IOP20PSTN Line to IP-PBX - Call answer - Terminator disconnect
  1. Make a call from a PSTN line to an IP-PBX line.
  2. Answer the call.
  3. The IP-PBX line terminates the call.
Pass with CaveatAll calls from PSTN to EdgeMarc show a duplication ptime attribute in SDP responses. This does not appear to impact on call (passed with a caveat) but is bad practice.
IOP21PSTN Line to IP-PBX - Busy subscriber
  1. Make a call from a PSTN line to a busy IP-PBX line (without divert on busy).
  2. Wait for the IP-PBX to return the busy response.
Pass
IOP22PSTN Line to IP-PBX - No answer timeout test, Invoked by PBX
  1. Make a call from a PSTN line to an IP-PBX line (without divert on no answer).
  2. Wait for the IP-PBX to return the no answer timeout response.
Pass
IOP23PSTN Line to IP-PBX - Subscriber not reachable
  1. Make a call from a PSTN line to an invalid number/unprogrammed DDI on the IP-PBX.
  2. Wait for the IP-PBX to return a response.
Pass
IOP24Verify CLIP service on IP-PBX line (incoming call from PSTN) 
  1. Make a call from the PSTN line to the IP-PBX line.
  2. The PSTN line is set to allow the CLI presentation. Check that the CLI is delivered as expected.
  3. Either party terminates the call.
Pass
IOP25Verify CLIR service on IP-PBX line (incoming call from PSTN)
  1. Make a call from the PSTN line to the IP-PBX line.
  2. PSTN line is set to restrict the CLI presentation. Check that CLI is not delivered as expected.
  3. Either party terminates the call.
Pass
IOP26Verify CLIP service on PSTN line (outgoing call from IP-PBX, From)
  1. Ensure Virgin Media agrees with the number used in the From header, and enter the number into the soft switch database for screening.
  2. Make a call from an IP-PBX line to a PSTN line.
  3. Ensure the eSBC configuration enables the IP-PBX line to send the From header containing the Calling Line ID (CLI) in the INVITE.
  4. Ensure that the eSBC allows the presentation of its CLI, using the privacy-header (Privacy: none or privacy-header not present).
  5. Ensure that the expected CLI is presented to the PSTN line.
  6. Either party terminates the call.
Pass
IOP27

Verify CLIP service on PSTN line (outgoing call from IP-PBX, PAI/PPI)

Vendor to ensure PAI number is different to that from which the call originates.

  1. Ensure Virgin Media agrees with the number used in the PAI/PPI header, and enter the number into the soft switch database for screening.
  2. Make a call from an IP-PBX line to a PSTN line.
  3. Ensure that the eSBC configuration enables the IP-PBX line to send the PAI/PPI header containing the Calling Line ID (CLI) in the INVITE. Note that if the PAI header is populated, it will be used in preference to the From header.
  4. Ensure that the eSBC allows the presentation of its CLI, using the privacy-header (Privacy: none or privacy-header not present).
  5. Ensure that the expected CLI is presented to the PSTN line.
  6. Either party terminates the call.
Pass
IOP28Verify CLIR service on PSTN line (outgoing call from IP-PBX)
  1. Ensure Virgin Media agrees with the number used in the From/PAI header, and enter the number into the soft switch database for screening.
  2. Make a call from an IP-PBX line to a PSTN line. 
  3. Ensure that the eSBC configuration enables the IP-PBX line to send the From and/or PAI header, containing either the Calling Line ID or obscured information in the INVITE.

Example:
From: "user751000" <sip:+441256751000@192.168.1.10>;tag=12345
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=12345

  • Ensure that the eSBC restricts the presentation of its CLI, using the privacy-header (Privacy: id or Privacy: user or Privacy: user;id).
  • Ensure that CLI is not presented to the PSTN line.
  • Either party terminates the call.
Pass
IOP29Verify Call Forward Immediate (unconditional) on an IP-PBX line (Incoming call from PSTN, call forward terminates within IP-PBX)
  1. Make a call from a PSTN line to an IP-PBX line with Call Forward to a line within the same IP-PBX.
  2. Answer the call.
  3. Either party terminates the call.

The IP-PBX does not have configuration settings to send SIP status 181 messages to the soft switch.

Pass
IOP30Verify Call Forward Immediate (unconditional) on an IP-PBX line (Incoming call from PSTN, call forward terminates PSTN)
  1. Make a call from a PSTN line to an IP-PBX line with Call Forward to a line in the PSTN.
  2. Answer the call.
  3. Either party terminates the call.
Pass
IOP31Verify Call Forward Busy on IP-PBX line (Incoming call from PSTN, call forward terminates within IP-PBX)
  1. Make a call from a PSTN line to an IP-PBX line with Call Forward Busy (or equivalent) to a line within the IP-PBX.
  2. Answer the call.
  3. Either party terminates the call.
Pass
IOP32Verify Call Forward No-answer on IP-PBX line (Incoming call from PSTN, call forward terminates within IP-PBX)
  1. Make a call from a PSTN line to an IP-PBX line with Call Forward No-answer (or equivalent) to a line within the IP-PBX.
  2. Answer the call.
  3. Either party terminates the call.
Pass
IOP33Verify Call Hold Service on IP-PBX (Incoming call from PSTN)
  1. Make a call from a PSTN line to an IP-PBX line with Call Hold.
  2. Answer the call.
  3. IP-PBX line places the call on hold.
  4. Leave the call on hold for 30 seconds and then retrieve the call.
  5. Ensure speech path is re-established in both directions.
  6. Either party terminates the call.
Pass
IOP34Verify three-party conference service on IP-PBX (Incoming call from PSTN, third party within IP-PBX)
  1. Make a call from a PSTN line to an IP-PBX line with a three-party conference.
  2. Answer the call.
  3. IP-PBX line uses the three-party conference facility to place the PSTN line on hold while dialing the third party (on another IP-PBX line).
  4. Once the third party answers the call, place the three parties in a conference.
  5. Ensure that all parties have a two-way speech path. Keep the speech path open for at least 20 seconds.
  6. Either party terminates the call.
Pass
IOP35Verify three-party conference service on IP-PBX (Incoming call from PSTN, third party PSTN)
  1. Make a call from a PSTN line to an IP-PBX line with a three-party conference.
  2. Answer the call.
  3. IP-PBX line uses the three-party conference facility to place the PSTN line on hold while dialing the third party (on another IP-PBX line).
  4. Once the third party has answered the call, place the three parties in a conference.
  5. Ensure that all parties have a two-way speech path. Keep the speech path open for at least 20 seconds.
  6. Either party terminates the call.
Pass
IOP36Verify do-not-disturb service on IP-PBX line (Incoming call from PSTN)
  1. The call does not ring.
  2. PSTN line receives an appropriate announcement or tone.
  3. Record the SIP status received from the IP-PBX.
Pass

IOP37Verify Call park service on IP-PBX line (Incoming call from PSTN)
  1. Make a call from a PSTN line to an IP-PBX line A with the Call Park (or equivalent) feature active.
  2. Answer the call.
  3. Place the call in the Park condition.
  4. After 10 seconds, retrieve the call from the IP-PBX line B, using the Call Park pick-up code.
  5. Ensure the speech path is re-established in both directions.
  6. Either party terminates the call.
Pass
IOP38Verify Call Waiting on an IP-PBX line, involving a PSTN line
  1. Make a call from PSTN line A to an IP-PBX line with Call Waiting active.
  2. Answer the call. 
  3. Make a call from the PSTN line B to the same IP-PBX line, which should receive an indication that a second call is waiting.
  4. PSTN line B receives the ringback tone.
  5. IP-PBX line answers the call from the PSTN line B.
  6. PSTN line A should receive an appropriate indication that they are now on hold.
  7. IP-PBX line toggles the call back to PSTN line A.
  8. Ensure the speech path is re-established in both directions and that PSTN line B receives an appropriate indication that they are now on hold.
  9. Either party terminates the call.
Pass
IOP39Verify DTMF transmission from/to IP-PBX - Inband
  1. Configure the IP-PBX/eSBC to send the DTMF transmission in-band.
  2. Make a call from an IP-PBX line to a PSTN line.
  3. Answer the call. 
  4. PSTN line presses each of the keys on the number pad in turn. Note the far-end experience.
  5. IP-PBX line presses each of the keys on the number pad in turn. Note the far-end experience.

The received DTMF tone is reflective of the length of time the key was pressed.

Not executedThe SBC team from VM changed this to not executed as IN Band DTMF tones are not currently supported and will require a feature to be added to the code.
IOP40Verify DTMF transmission from/to IP-PBX - RFC 2833 - Telephone-event 
  1. Configure the IP-PBX/eSBC to send the DTMF transmission, using RFC 2833 - telephone-event.
  2. Make a call from an IP-PBX line to a PSTN line.
  3. Answer the call. 
  4. PSTN line presses each of the keys on the number pad in turn. Note the far-end experience.
  5. IP-PBX line presses each of the keys on the number pad in turn. Note the far-end experience.

The received DTMF tone is reflective the length of time the key was pressed.

Pass
IOP41T.38 Fax transmission mode - PSTN to IP-PBX origination
  1. Configure the ATA/IP-PBX/eSBC so that the Fax transmission is sent using the T.38 Version 0 Fax transmission mode.
  2. Make a call from an PSTN line to an IP-PBX line.
  3. Answer the call. 
  4. Fax transmission is completed and the call is terminated by either of the end terminal devices.
  5. Ensure the Wireshark trace shows it is using the T.38 Fax Transmission.
  6. Check that the fax is transmitted and received as expected.
FailedThe EM6000 only sent a Re-INVITE when the fax machine was connected directly to FXS ports to negotiate the T.38.
IOP42T.38 Fax transmission mode - IP-PBX to PSTN origination
  1. Configure the ATA/IP-PBX/eSBC so that the Fax transmission is sent using the T.38 Version 0 Fax transmission mode.
  2. Make a call from an IP-PBX line to a PSTN line.
  3. Answer the call.
  4. Fax transmission is completed and the call is terminated by either of the end terminal devices.
  5. Ensure Wireshark trace shows that the T.38 Fax Transmission is used.
  6. Check that the fax is transmitted and received as expected.
FailedThe EM6000 only sent a Re-INVITE when the fax machine was connected directly to FXS ports to negotiate the T.38.
IOP43In-band G.711 Fax transmission mode - PSTN to IP-PBX origination
  1. Configure the ATA/IP-PBX/eSBC so that Fax transmission is sent using the in-band G.711 Fax transmission mode.
  2. Make a call from a PSTN line to an IP-PBX line.
  3. Answer the call.
  4. Fax transmission is completed and the call is terminated by either of the end terminal devices.
  5. Ensure the Wireshark trace shows that the in-band G.711 Fax Transmission is used.
  6. Check that the fax is transmitted and received as expected.
Pass
IOP44In-band G.711 Fax transmission mode - IP-PBX to PSTN origination
  1. Configure the ATA/IP-PBX/eSBC so that the Fax transmission is sent using the  in-band G.711 Fax transmission mode.
  2. Make a call from an IP-PBX line to a PSTN line.
  3. Answer the call.
  4. Fax transmission is completed and the call is terminated by either of the end terminal devices.
  5. Ensure the Wireshark trace shows that the in-band G.711 Fax Transmission is used.
  6. Check that the fax is transmitted and received as expected.
Pass
IOP45Test for call in progress audit function (response to in-call OPTIONS from soft switch to eSBC) and session refresh and response to UPDATE messages
  1. Make a call from an IP-PBX line to a PSTN line.
  2. Answer the call.
  3. Leave the two parties in conversation for 35 minutes.
  4. Ensure the Session-expires setting is 3600 or less.
  5. Ensure both parties have two-way speech at the beginning and end of call.
  6. Either party terminates the call.
  7. Check the Wireshark trace to ensure that the in-call OPTIONS are sent by the soft switch and that the eSBC responds with the status 200OK.
  8. Check if the eSBC sends any in-call audit SIP messages.
  9. Check for session refresh Update or Re-Invite and correct response.
Pass
IOP46

Test for four simultaneous calls: two inbound, two outbound calls

Vendor to configure eSBC for Round robin to ensure calls go to both primary and secondary SBC.

  1. Configure the eSBC so that successive calls route to alternate SBCs (round robin, cyclic, and so on).
  2. Make four simultaneous calls: two inbound and two outbound calls.
  3. Answer the calls and ensure two-way speech path for each call.
Not executedRound robin outbound calls is not a feature of the EdgeMarc platform. Adding this feature requires a change in the EdgeMarc code. 
IOP47Test for eSBC endpoint restart-recovery
  1. Restart the eSBC.
  2. Ensure that, after recovery, inbound and outbound calls are successful.
Pass
IOP48Test for eSBC loss of Ethernet link and reconnection
  1. Remove the Ethernet link between the eSBC and CE router. Leave it in this condition for at least 3 minutes.
  2. Reconnect the Ethernet link and ensure that after approximately 2 minutes inbound and outbound calls are successful.
Pass
IOP49Test for the Primary SBC loss 

Note: Contact an MSL engineer to carry out the following.

  1. On the Primary SBC, carry out the ALLSTOP command to disable the SBC.
  2. Make a call from the IP-PBX line to a PSTN Line.
  3. Make sure that the call tries to route to the Primary SBC. On a non-response to the INVITE, the eSBC re-routes the call to the Secondary SBC.
  4. Wait for call answer.
  5. Either party terminates the call.

Note: Contact an MSL engineer to carry out the following.

  1. Restart the Primary SBC.
Pass
IOP51Test for Call forward Internal Busy

The following is an additional test to cover when vendors are using the Avaya Communication Manager 7.1:

  1. PBX Subscriber 1 makes a call to PBX Subscriber 2, so that the PSTN to call the PBX subscriber 1 is Busy.
  2. PSTN calls PBX user 1. The call should automatically go to voicemail after 10 seconds when call forwarding is off.
  3. If VM is on another PBX Internal Line, the call should go to voicemail.
  4. PSTN user listens to the voiceMail announcement, and leaves a clear message for PBX Subscriber 1 in VM.
  5. If forwarded to voicemail, the PSTN terminates the call after hearing the VM announcement.
  6. If forwarded to another user, either party terminates the call after checking that speech is clear in both directions.
Not executed
IOP52Test for Call forward internal on No Answer

The following is an additional test to cover when vendors are using the Avaya Communication Manager 7.1:

  1. PSTN calls PBX user 1.
  2. The PBX User 1 should not answer the call.
  3. The call should automatically go to voicemail (VM), which is in another internal PBX line if call forwarding is turned off.
  4. The call automatically goes to voicemail after 10 seconds.
  5. The PSTN terminates the call after hearing the VM announcement.
  6. If call forwarding is ON, the call is forwarded to another PBX user internal.
  7. Check the speech quality and terminate the call after checking that speech is clear in both directions.
Not executed
IOP53Test for making a call from a PBX to a PSTN
  1. Configure the eSBC to offer the T.38 in addition to G711A-law and G711-U law.
  2. Make a call from the PBX to a PSTN.
  3. Ensure the call is connected and dialog takes place for 10 minutes.
  4. Check the Wireshark output. Confirm that the T.38 is not reflected in the protocol column after the call is connected for 7 minutes.
  5. If the T.38 is reflected in the protocol column, take a note.
Pass