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Document Overview

This document provides a configuration guide for Ribbon EdgeMarc 2900A when connecting to Cisco Unified Communication Manager CUCM.

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

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound calls flows between Ribbbon EdgeMarc 2900A and the Cisco Unified Communication Manager CUCM platform.

Audience

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

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 provided:

Table : Requirements


Equipment

Software Version

Ribbon Communications

Ribbon EdgeMarc 2900A

V15.8.0

Third-party Equipment

Cisco Unified Communication Manager

12.0.1.21900-7

Kapanga Softphone1.00.2182d

MicroSIP Softphone3.19.29

Cisco IP Communicator Softphone8.6.2.0

VentaFax Fax Machine7.6.243.616

NGT Litev.1.51

Reference Configuration

The following reference configuration shows connectivity between the third-party product and Ribbon EdgeMarc 2900A.

Figure : Reference Configuration

Support

For any questions regarding this document or the content herein, 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/outbound call
  • Hold and Resume
  • Call Forwarding
  • FAX
  • DTMF
  • Conference Call
  • Action on eSBC outage (restart of eSBC)
  • Action on Loss of Virgin Media primary SBC

Cisco UCM 12 Configuration

The following new configurations are included in this section:

  1. SIP Profile
  2. SIP Trunk Security Profile
  3. Trunk
  4. Route Group
  5. Route List
  6. Route Pattern

1. SIP Profile

Select Device > Device Settings > SIP Profile

Figure : SIP Profile

2. SIP Trunk Security Profile

Select System> Security > SIP Trunk Security Profile

Figure : SIP Trunk Security Profile


3. Trunk

Select Device > Trunk

Figure : Trunk


4. Route Group

Select Call Routing > Route/Hunt > Route Group

Figure : Route Group

5. Route List

Select Call Routing > Route/Hunt > Route List

Figure : Route List


6. Route Pattern

Select Call Routing > Route/Hunt > Route Pattern

Note

Use this procedure to create any Route Pattern configuration.


Figure : Route Pattern


EdgeMarc Configuration

Network

VoIP

Network

LAN and WAN Interfaces

Login to EdgeMarc as root user and go to Network to configure the LAN and WAN interfaces.

Figure : EdgeMarc Network LAN Interface

 

Figure : EdgeMarc Network WAN Interface

Static Routes

Navigate to Network > Static Routesto configure the routes.

Figure : Static Routes

VoIP

VoIP Settings

  1. Login as root user and navigate to VoIP to configure the VoIP features.

Figure : VoIP

SIP Settings

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

Figure : SIP

 

 

B2BUA

  1. Navigate to VoIP > B2BUA
  2. Configure LAN Part with the next form.

Figure : B2BUA

Test Results

The following table provides information about the tests that Ribbon performed to complete all scenarios that Virgin Media needs for customers.

S.NoProcedureObservationResultComment
IOP1

Vendors eSBC response to SIP OPTIONS messages from SBC

No calls are required for this test. SIP trace to be captured for approximately 60 seconds and checked for correct signaling.

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

e.g. 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. SIP trace to be captured for approximately 60 seconds (depending on agreement) and checked for correct signaling.

Vendors eSBC setup for Solution IP.Addr Mode
eSBC configured to send OPTIONS messages to the SBC on a periodic basis. The SBC responds with SIP response 200OK, for example:
"OPTIONS sip:ping@192.168.1.10:5060 SIP/2.0"

Verify that the eSBC can simultaneously send SIP OPTIONS messages to both the solution SBC addresses.

Pass
IOP4Basic test call from IP-PBX to PSTN line through SBC-A (using SBC-A IPV4 ip address). 

IP-PBX line initiates call, Call is answered, IP-PBX line terminates call.

Vendors eSBC setup for Solution IP.Addr Mode
Call from the IP-PBX. Invite seen from eSBC to SBC-A, proxy authentication challenge returned to eSBC, re-invite with correct credentials from eSBC and call progresses as expected.
For 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 and confirm that G.711 A law codec with 10 or 20ms packetisation is used.
Also check to see if INVITE contains the Session-Expires header and that the INVITE is syntactically correct. Check for Supported Header to see if 'timer' is supported. Ensure that the response in the 200 OK is compatible with the INVITE and verify that the Required Header contains 'timer'. (x-ref IOP9)

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 test completed eSBC to be configure to use Primary SBC-A for calls to route to.

IP-PBX line initiates call, Call is answered, IP-PBX line terminates call.

Vendors eSBC setup for Solution IP.Addr Mode
Call from the IP-PBX. Invite seen from eSBC to SBC-B, proxy authentication challenge returned to eSBC, re-invite with correct credentials from eSBC and call progresses as expected.
e.g.
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 and confirm that G.711 A law codec with 10ms or 20ms packetisation is being used.

Pass
IOP7b

Called Number format - vendors 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)

SBC to be configured for Global calling plan.

IP-PBX line initiates call to PSTN line, Call is answered.
IP-PBX line terminates call.

Configure the eSBC to present the called number in the user part of the Request & To URIs to be sent 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 - vendors eSBC to soft switch number normalization - Global Dial Plan

Test eSBC capability to send 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)

SBC to be configured for Global calling plan.

IP-PBX line initiates call to PSTN line, Call is answered.
IP-PBX terminates call.

Configure the eSBC to present the calling number in the user part of the From & PAI URIs to be sent 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)

SBC to be configured for Global calling plan.

PSTN line initiates call to IP-PBX line, Call is answered.
PSTN line terminates call.

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)

Also check to see that the INVITE contains Session-Expires header and that it is syntactically correct. Check for Supported Header and ensure 'timer' is supported. Ensure response in 200 OK is compatible with INVITE and check for Required Header and if it contains '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)

SBC to be configured for Global calling plan.

PSTN line initiates call to IP-PBX line, Call is answered.
PSTN line terminates call.

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

Call made from IP-PBX line to the Emergency services using 999. Call answered.
Either party terminates 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

Call made from IP-PBX line to the Emergency services using 112. Call answered,
Either party terminates 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

Call made from IP-PBX line using a text direct set to the Emergency services using 18000. Call answered.
Either party terminates 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

Call made from IP-PBX line to PSTN line, Answer Call.
IP-PBX line terminates call.

Pass
IOP15PSTN calls SIP #1, SIP #1 conferences in SIP #2

Call made from IP-PBX line to PSTN line, Answer Call.
PSTN line terminates call

Pass
IOP16IP-PBX Line to PSTN - Busy subscriber

Call made from IP-PBX line to a busy PSTN line (without divert on busy)
Wait for soft switch to return busy response. Ensure that eSBC does not recurse and setup call via secondary SIP trunk.

Pass
IOP17IP-PBX Line to PSTN - No answer timeout test

Call made from IP-PBX line to a PSTN line (without divert on no answer)
Do not answer call.
Wait for soft switch to return no answer timeout response. Ensure that eSBC does not recurse and setup call via secondary SIP trunk.

Pass With CaveatCancel message is sent by SfB 2015 server and there is not an option to change the timer for this.
IOP18

IP-PBX Line to PSTN - Subscriber not reachable

Vendor to call 01189111111

Call made from IP-PBX line to an invalid number.
Wait for soft switch to return response. Ensure that eSBC does not recurse and setup call via secondary SIP trunk.

Pass
IOP19PSTN Line to IP-PBX - call answer - Originator disconnect. 

Call made from a PSTN line to an IP-PBX line, Answer Call.
Originator disconnects call.

Pass.
IOP20PSTN Line to IP-PBX - call answer - Terminator disconnect

Call made from a PSTN line to an IP-PBX line, Answer Call.
IP-PBX line terminates call.

Pass
IOP21PSTN Line to IP-PBX - busy subscriber

Call made from PSTN line to a busy IP-PBX line (without divert on busy)
Wait for IP-PBX to return busy response.

De-ScopedSfB 2015/Lync does not support Busy line due to a permanent call waiting service. If a UM/Voicemail service is activated call goes there.
IOP22PSTN Line to IP-PBX - No answer timeout test, Invoked by PBX

Call made from a PSTN line to an IP-PBX line (without divert on no answer).

Wait for the IP-PBX to return no answer timeout response

De-ScopedSfB2015/Lync does not support No answer time out. If a UM/Voicemail service is activated call goes there.
IOP23PSTN Line to IP-PBX - subscriber not reachable

Call made from a PSTN line to an invalid number/unprogrammed DDI on the IP-PBX.
Wait for IP-PBX to return response.

Pass
IOP24Verify CLIP service on IP-PBX line (incoming call from PSTN) 

Call made from PSTN line to IP-PBX line. PSTN line is set to allow CLI presentation.
Check that CLI is delivered as expected.
Either party terminates call.

Pass
IOP25Verify CLIR service on IP-PBX line (incoming call from PSTN)

Call made from PSTN line to IP-PBX line. PSTN line is set to restrict CLI presentation.
Check that CLI is not delivered as expected.
Either party terminates call.

Pass
IOP26Verify CLIP service on PSTN line (outgoing call from IP-PBX, From)

Ensure number used in From header is agreed with Virgin Media and entered into the soft switch database for screening purposes.

Call made from an IP-PBX line to a PSTN line.
Ensure that the eSBC is configured such that the IP-PBX line sends From header containing Calling Line ID (CLI) in the INVITE.

Ensure that the eSBC allows presentation of its CLI using privacy-header (Privacy: none or privacy-header not present)

Ensure that the expected CLI is presented to the PSTN line.
Either party terminates 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

Ensure number used in PAI/PPI header is agreed with Virgin Media and entered into the soft switch database for screening purposes.

Call made from an IP-PBX line to a PSTN line.
Ensure that the eSBC is configured such that the IP-PBX line sends PAI/PPI header containing Calling Line ID (CLI) in the INVITE.
If PAI header is populated this will be used in preference to the From header.
Ensure that the eSBC allows presentation of its CLI using privacy-header (Privacy: none or privacy-header not present)

Ensure that the expected CLI is presented to the PSTN line.
Either party terminates call.

Pass
IOP28Verify CLIR service on PSTN line (outgoing call from IP-PBX)

Ensure number used in From/PAI header is agreed with Virgin Media and entered into the soft switch database for screening purposes.

Call made from an IP-PBX line to a PSTN line.
Ensure that the eSBC is configured such that the IP-PBX line sends From and/or PAI header containing either the Calling Line ID or obscured information in the INVITE.
e.g.
From: "user751000" <sip:+441256751000@192.168.1.10>;tag=12345
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=12345

Ensure that the eSBC restricts presentation of its CLI using privacy-header (Privacy: id or Privacy: user or Privacy: user;id)

Ensure that CLI is NOT presented to the PSTN line.
Either party terminates call.

Pass
IOP29Verify Call Forward Immediate (unconditional) on a IP-PBX line (Incoming call from PSTN, call forward terminates within IP-PBX)

Call made from a PSTN line to an IP-PBX line with call forward to a line within the same IP-PBX, Answer Call.
Either party terminates call.

Does the IP-PBX have configuration settings to send SIP status 181 messages to the soft switch?

Pass
IOP30Verify Call Forward Immediate (unconditional) on a IP-PBX line (Incoming call from PSTN, call forward terminates PSTN)

Call made from a PSTN line to an IP-PBX line with call forward to a line in the PSTN, answer call.
Either party terminates call.

Pass
IOP31Verify Call Forward Busy on IP-PBX line (Incoming call from PSTN, call forward terminates within IP-PBX)

Call made from a PSTN line to an IP-PBX line with Call Forward Busy (or equivalent) to a line within the IP-PBX, answer call.
Either party terminates call.

Not-Exec
IOP32Verify Call Forward No-answer on IP-PBX line (Incoming call from PSTN, call forward terminates within IP-PBX)

Call made from a PSTN line to an IP-PBX line with Call Forward No-answer (or equivalent) to a line within the IP-PBX, Answer Call.
Either party terminates call.

Pass
IOP33Verify Call Hold Service on IP-PBX (Incoming call from PSTN)

Call made from a PSTN line to an IP-PBX line with Call Hold, answer call.
IP-PBX line puts the call on hold.
Leave call on hold for 30 seconds and then retrieve call. Ensure speech path is re-established in both directions.
Either party terminates call.

Pass
IOP34Verify 3-party conference service on IP-PBX (Incoming call from PSTN, 3rd party within IP-PBX)

Call made from a PSTN line to an IP-PBX line with 3-party conference, answer call.
IP-PBX line uses the 3-party conference facility to put the PSTN line on hold while dialing 3rd party. (another IP-PBX line)
Once the 3rd party has answered the call, place the three parties in a conference.
Ensure that all parties have a two-way speech path.
Keep the speech path open for at least 20 seconds.
Either party terminates call.

PassConference is created on the SfB 2015 server as another room/place where all other users are connected. All users must release the call to be disconnected from this conference call.
IOP35Verify 3-party conference service on IP-PBX (Incoming call from PSTN, 3rd party PSTN)

Call made from a PSTN line to an IP-PBX line with 3-party conference, answer call.
IP-PBX line uses the 3-party conference facility to put PSTN line on hold whilst dialing 3rd party. (another PSTN line)
Once the 3rd party has answered the call, place the 3 parties in a conference.
Ensure that all parties have a two-way speech path.
Keep the speech path open for at least 20 seconds.
Either party terminates call.

PassConference is created on the SfB 2015 server as another room/place where all other users are connected. All users must release the call to be disconnected from this conference call.
IOP36Verify do-not-disturb service on IP-PBX line (Incoming call from PSTN)

Does not ring.
PSTN line receives an appropriate announcement or tone.

Record the SIP status received from IP-PBX.

Pass

IOP37Verify Call park service on IP-PBX line (Incoming call from PSTN)

Call made from a PSTN line to IP-PBX line A with Call Park (or equivalent) feature active, answer call.
Place the call in the Park condition.
After 10 seconds, retrieve call from IP-PBX line B using the Call Park pick-up code.
Ensure speech path is re-established in both directions.
Either party terminates call.

Pass
IOP38Verify Call Waiting on an IP-PBX line, involving a PSTN line

Call made from PSTN line A to an IP-PBX line with Call Waiting active, answer call.
Call made from PSTN line B to the same IP-PBX line which should receive an indication that a second call is waiting.
PSTN line B receives ringback tone.
IP-PBX line answers the call from PSTN line B.
PSTN line A should receive an appropriate indication that they are now on hold.
IP-PBX line toggles the call back to PSTN line A
Ensure speech path is re-established in both directions and that PSTN line B should receive an appropriate indication that they are now on hold.
Either party terminates call.

Pass
IOP39Verify DTMF transmission from/to IP-PBX - Inband

Configure the IP-PBX/eSBC to send DTMF transmission in-band.

Call made from IP-PBX line to a PSTN line, answer call.
PSTN line presses each of the keys on the number pad in turn. Note the far end experience.
IP-PBX line presses each of the keys on the number pad in turn. Note the far end experience.

Was the received DTMF tone reflective the length of time the key was pressed?

Pass
IOP40Verify DTMF transmission from/to IP-PBX - RFC 2833 - telephone-event 

Configure the IP-PBX/eSBC to send DTMF transmission using RFC 2833 - telephone-event.

Call made from IP-PBX line to a PSTN line, Answer call.
PSTN line presses each of the keys on the number pad in turn. Note the far end experience.
IP-PBX line presses each of the keys on the number pad in turn. Note the far end experience.

Was the received DTMF tone reflective the length of time the key was pressed?

Pass
IOP41T.38 Fax transmission mode - PSTN to IP-PBX origination

Configure the ATA/IP-PBX/eSBC such that Fax transmission is sent using T.38 Version 0 Fax transmission mode.
Call made from PSTN line to an IP-PBX line, answer call.
Fax transmission is completed and call is terminated by either of the end terminal devices.

Ensure Wireshark trace shows that T.38 Fax Transmission is used. Check that the fax is transmitted and received as expected.

Pass
IOP42T.38 Fax transmission mode - IP-PBX to PSTN origination

Configure the ATA/IP-PBX/eSBC such that Fax transmission is sent using T.38 Version 0 Fax transmission mode.
Call made from IP-PBX line to a PSTN line, answer call.
Fax transmission is completed and call is terminated by either of the end terminal devices.

Ensure Wireshark trace shows that T.38 Fax Transmission is used. Check that the fax is transmitted and received as expected.

Pass
IOP43In-band G.711 Fax transmission mode - PSTN to IP-PBX origination

Configure the ATA/IP-PBX/eSBC such that Fax transmission is sent using in-band G.711 Fax transmission mode.
Call made from PSTN line to an IP-PBX line, answer call.
Fax transmission is completed and call is terminated by either of the end terminal devices.

Ensure Wireshark trace shows that in-band G.711 Fax Transmission is used. Check that the fax is transmitted and received as expected.

Pass
IOP44In-band G.711 Fax transmission mode - IP-PBX to PSTN origination

Configure the ATA/IP-PBX/eSBC such that Fax transmission is sent using in-band G.711 Fax transmission mode.
Call made from IP-PBX line to a PSTN line, Answer call.
Fax transmission is completed and call is terminated by either of the end terminal devices.

Ensure Wireshark trace shows that in-band G.711 Fax Transmission is used. Check that the fax is transmitted and received as expected.

Pass

IOP45Test of Call in progress audit function (response to in-call OPTIONS from soft switch to eSBC) & session refresh & response to UPDATE messages.

Call made from an IP-PBX line to a PSTN line, answer call.
Leave the two parties in conversation for 35 minutes.
Ensure Session-expires setting is 3600 or less.
Ensure both parties have two-way speech at beginning and end of call.
Either party terminates call.

Check the Wireshark trace to ensure that in-call OPTIONS are sent by the soft switch and that the eSBC responds with status 200OK.
Check to see if the eSBC sends any in-call audit SIP messages.
Check for session refresh Update or Re-Invite and correct response.

Pass
IOP46

Test of 4 simultaneous calls, 2 inbound, 2 outbound calls

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

Configure the eSBC so that successive calls route to alternate SBCs (round robin, cyclic, etc.)
Make 4 simultaneous calls 2 inbound, 2 outbound calls. Answer calls and ensure two-way speech path for each call.

Pass
IOP47Test of eSBC endpoint restart-recoveryRestart the eSBC and ensure that after recovery, inbound and outbound calls are successful.Pass
IOP48Test of eSBC loss of Ethernet link and reconnectionRemove the Ethernet link between the eSBC and CE router. Leave in this condition for at least 3 minutes. Reconnect the Ethernet link and ensure that after approximately 2 minutes inbound and outbound calls are successful.Pass
IOP49Test of Primary SBC loss 

** Contact MSL engineer to carry out the following **
On the Primary SBC carry out the ALLSTOP command to disable the SBC.

Call made from IP-PBX line to a PSTN Line.
Call should attempt to route to Primary SBC. On non-response to INVITE, eSBC re-routes the call to the Secondary SBC.
Wait for call answer.
Either party terminates call.

** Contact MSL engineer to carry out the following **
Restart the Primary SBC

Pass
IOP51Test of verify call forward Internal Busy

Additional test to cover when vendors are using Microsoft Skype for Business 2015.

PBX Subscriber 1 to make call to another PBX Subscriber 2 so that PSTN to call PBX subscriber 1 is Busy.

PSTN call PBX user 1. The call should automatically go to voicemail after 10 seconds when forwarding is off.

VM is on another PBX Internal Line call should go to Voice Mail.

If voicemail PSTN to listen VM announcement if another PBX user, check speech is clear in both directions.

If forwarded to voicemail PSTN terminated call after hearing VM announcement.

If forwarded to another user either party terminate the call after checking speech is clear in both directions.

Pass
IOP52Test of Call forward internal on No Answer

Additional test to cover when vendors are using Microsoft Skype for Business 2015.

PSTN call PBX user 1. PBX User 1 is not to answer the call.

The call should automatically go to voicemail (VM) which is in another internal PBX line if call forwarding is turned off.

Call automatically goes to voicemail after 10 seconds.

PSTN terminated call after hearing VM announcement.

If forwarded is ON call is forwarded to another PBX user internal.

Check speech quality, terminate the call after checking speech is clear in both directions.

Pass
IOP53Test Call from PBX to PSTN
  1. Configure eSBC to offer T.38 in addition to G711A-law and G711-U law.
  2. Call made from PBX to PSTN.
  3. Call established and two dialogs for 10 minutes.
  4. Check Wireshark output. You should not see T.38 being reflected in the protocol column after the call has been established for 7 minutes.
  5. If T.38 is reflected in the protocol column, make a note of this.
Pass