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Info

This Application Note is offered as a convenience to Ribbon customers. The specifications and information regarding the product in this document are subject to change without notice. All statements, information, and recommendations contained in this document 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 contained here.

Info

The links are only internally to Ribbon partners and employees. They don´t work outside from the Ribbon Network.

Requirements

The following table lists the hardware and software used in the reference configuration.

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1Reference Configuration Topology
3Topology

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Support

 

For questions about information in this document, contact Ribbon Support in either of the following ways:

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1Define FQDN
3Define FQDN

 

 

 

 

 

 

 

 

 

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1Define IP Address Type
3Define IP Address Type

 

 

 

 

 

 

 

 

 

 

 

 

 

 

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1Define Root Trunk
3Define Root Trunk

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Voice Policy
Voice Policy
2. Voice Policy

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Select Control Panel > Voice Routing >Route to access the Route configuration screen.

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1Route
Route3Route

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Trunk Configuration
Trunk Configuration
5. Trunk Configuration

Select Control Panel > Voice Routing >Trunk Configuration to access the trunk configuration screen.

 

 

 

 

 

 

 

 

 

 

 

 

 

 

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1Trunk Configuration
3Trunk Configuration

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1Virgin Media SIP Profile
3Virgin Media SIP Profile

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

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1Skype 2015 SIP Profile
3Skype 2015 SIP Profile

 

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11Fax SIP Profile
3Fax SIP Profile

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1Virgin Media SIP Servers
3Virgin Media SIP Servers

 

 

 

 

 

 

 

 

 

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1Skype 2015 SIP Server
3Skype 2015 SIP Server

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1Virgin Media Signaling Group
3Virgin Media Signaling Group

 

 

 

 

 

 

 

 

 

 

 

 

 

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1Skype 2015 Signaling Group
3Skype 2015 Signaling Group

 

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1Fax Signaling Group
3Fax Signaling Group

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Tranformation
Tranformation
8. Transformation

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1Skype Transformation
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1Fax-Tenor Transformation
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1Virgin Media Call Routing
3Virgin Media Call Routing

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

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1Skype Call Routing
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1Fax Call Routing
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Test Results
Test Results
Interoperability Test Results

The following table provides test results for interoperability compliance testing between Ribbon SBC 1000/2000 and Skype for Business 2015.

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1Interoperability Compliance Test Results
3Test Results
Test NumberTest ScenarioSetup / Test ResultsStatusComment
IOP1vendors eSBC response to SIP OPTIONS messages from SBC

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

For each eSBC, the SBC will periodically send 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 will be placed (or remain) 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 eSBCNo calls are required for this test. SIP trace to be captured for approx 60 seconds (depending on agreement) and checked for correct signalling.

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

Check 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.
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 being used.
Also check to see if INVITE contains Session-Expires header and that it is syntatically correct. Check for Supported Header to see if 'timer' is supported. Ensure response in 200 OK is compatible with INVITEand check for Required Header and if it contains 'timer'. (x-ref IOP9)
Pass 
IOP5Basic test call from IP-PBX to PSTN line through SBC-B (using SBC-B 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-B, proxy authentication challenge returned to eSBC, re-invite with correct credentials from eSBC and call progresses as expected.

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 and confirm that G.711 A law codec with 10ms packetisation is being used.

Pass 
OP7bIOP7bCalled Number format - vendors eSBC to soft switch number normalisation - 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 
IOP8bCalling Number format - vendors eSBC to soft switch number normalisation - 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 the 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 
IOP9bCalled Number format - soft switch to eSBC number normalisation - 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 
IOP10bCalling Number format - soft switch to eSBC number normalisation - 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 
IOP12Emergency 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 
IOP13Emergency 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 disconnectCall made from IP-PBX line to PSTN line, Answer Call.
IP-PBX line terminates call.
Pass 
IOP15IP-PBX Line to PSTN - call answer - Terminator disconnectCall made from IP-PBX line to PSTN line, Answer Call.
PSTN line terminates call
Pass 
IOP16IP-PBX Line to PSTN - Busy subscriberCall 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 testCall 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 caveat. 
IOP18IP-PBX Line to PSTN - Subscriber not reachableCall 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 disconnectCall made from a PSTN line to an IP-PBX line, Answer Call.
IP-PBX line terminates call.
Pass 
IOP23PSTN Line to IP-PBX - subscriber not reachableCall 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 
IOP27Verify CLIP service on PSTN line (outgoing call from IP-PBX, PAI/PPI)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.

Example:
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 has 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.
NoExecSkype server does not support busy line.
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 PSTN line on hold whilst dialling 3rd party (another IP-PBX 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.
Pass 
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 dialling 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.
Pass 
IOP36Verify do-not-disturb service on IP-PBX line (Incoming call from PSTN)Call made from a PSTN line to an IP-PBX line with do-not-disturb feature active. Ensure IP-PBX line 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 the 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 lineCall 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 - InbandConfigure 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 originationConfigure 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 originationConfigure 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 originationConfigure 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 originationConfigure 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.Make a call from an IP-PBX line to a PSTN line. Answer the call.
Leave the  two parties in conversation for 10 minutes.
Ensure both parties have two way speech.
Either party terminates call.

Check 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.
Pass 
IOP46Test of 4 simultaneous calls, 2 inbound, 2 outbound callsConfigure the eSBC such 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 approx 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  with CaveatWe had the duplication of 200 OK
IOP51Test of verify call forward Internal Busy

Additional test to cover when vendors is using Microsoft Skype for Business

sent to Voice Message after 10 seconds
IP-PBX has two numbers on call and PSTN called first number, Skype for Business sent to Voice Message after 10 seconds.Pass IOP52PSTN called IP-PBX user after 10 seconds the call goes to Voice MessagePSTN called IP-PBX user after 10 seconds Skype for Business sent to Voice Message.Pass

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 shoud automatically go to voicemail after 10 sec 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 another either party terminate the call after checking speech is clear in both directions.

Pass with CaveatWe had the duplication of 200 OK
IOP52Test of Call forward internal on No Answer

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

PSTN call PBX user 1. PBX User 1 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 with CaveatWe had the duplication of 200 OK
IOP53Test Call from PBX to PSTN
  1. eSBC to be configured to offer T.38 in addition to G711A-law and G711-U law
  2. Call made from PBX to PSTN
  3. Call to be established and two dialog for 10 minutes.
  4. Check Wireshark output. You should not see T.38 being reflected in the protocol column after call having been established for 7 minutes.
  5. If T.38 is reflected in the protocol column make a note of this.
Pass with Caveat We had the duplication of 200 OK 

 

Conclusion

This Application Note describes the steps required to configure the Ribbon SBC 1000/2000 to successfully interoperate with Skype for Business 2015 and Virgin Media SIP Trunk. All feature and serviceability test cases have been completed. The majority of test cases passed with noted exceptions and observations provided in Interoperability Test Results.

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Appendix A

  • For test case IOP13, we used the software NGT Lite to enabled Text direct to Emergency services:

  • For removed the "Message Authorization Digest" we applied the next "Message Manipulation":

Image Added

 

  • For test case IOP13, we used the software NGT Lite to enabled Text direct to Emergency services:

Image Added

 

  • For test case IOP27, it is required to add PAID with the next "Message Rule":

Image Added

 

To enabled this "Message Manipulation", you need to select the "Signaling Group" and enable there the next flag and the "Message Manipulation":

Image Added

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  • For test case IOP28, it is required that Enable forward P-Asserted-Identity data is selected in order to set CLIR.

Select Control Panel > Voice Routing > Trunk Configuration

  • in order to set CLIR.

Select Control Panel > Voice Routing > Trunk Configuration

Image Added

 

  • For test case IOP29, IOP30, IOP32, to configure any kind of Call Forward in Microsoft Skype for Business Client you need to select the number to forward the call:

Caption
0Figure
1IOP32
3IOP32

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  • For test case IOP36, it is required to configure in Microsoft Exchange Admin the service "Unified Messaging". After that, Skype for Business will send message 480 to Ribbon SBC 2000:

 

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  • For test case IOP39, the following configuration needs to be changed on the SBC Ribbon in order to send the DTM inband:

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  • For test cases IOP43 and IOP44, the following configuration needs to be changed on the SBC in order to send the Fax inband:

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