Ribbon EdgeMarc 2900A configuration Cisco Unified Communication Manager CUCM

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

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.

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)

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.

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)

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)

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'.

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)

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>

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>

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>

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

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

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.

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.

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.

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

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

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

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

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

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.

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.

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.

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.

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.

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?

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.

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.

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.

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.

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.

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.

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

Record the SIP status received from IP-PBX.

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.

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.

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?

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?

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.

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.

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.

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.

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.

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.

** 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

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.

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.

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.