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Document History

DateNameComment
06.09.2016Roman KokesInitial Draft
   

Introduction

The interoperability compliance testing focuses on verifying inbound and outbound call flows between Sonus SBC 1000/2000, Skype 2015 and AT&T IP Flexible Reach SIP Trunk.

...

Third-party Product Features

The following third-party product features are supported:

  • Basic originated and terminated calls
  • Basic inbound/outbound call
  • Hold and Resume
  • Call Forwarding
  • FAX
  • DTMF
  • Conference Call

Anchor
Not Supported Features
Not Supported Features
Not Supported Features

  • Network-Based Blind Call Transfer with REFER
  • Network-Based Consultative Call Transfer with REFER (Attended)
  • Network-Based Consultative Call Transfer with REFER (Unattended)

Verify License

No special licensing is required for this test.

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Back to Top

 

...

To configure Message Manipulation, select Settings > SIP > Message Manipulation > Condition Rule Table, as shown in the following figures.

Caption
0Figure
1AT&T Message Manipulation

Caption
0Figure
1Skype 2015 Message Manipulation

 

 

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Conclusion

This Application Note

AnchorTest ResultsTest ResultsInteroperability Test ResultsThe following table provides test results for interoperability compliance testing between Sonus SBC 1000/2000 and Skype for Business 2015. Caption
0Table
1Interoperability Compliance Test Results
 Category

Test

Test Case ID

Abstract

Call

Trace

Call

Type

Test Result Comment

Basic Call Tests

Hop-off to PSTN

(G.729 offered as first choice codec, except where specified)

Hop-off to PSTN20001Call Ringback
From a CPE Phone, make a call to a PSTN phone;
Do not answer the call on PSTN phone;
Verify the following:
1)  Calling Party hears ringback.
Hang up the CPE Phone, while the call is still ringing at Called Party (PSTN phone)
3)  Verify that Called Party Phone (PSTN phone) stops ringing and the call is cleared.NOIPLD-PSTNPASS 

Basic Call Tests

Hop-off to PSTN

(G.729 offered as first choice codec, except where specified)

Hop-off to PSTN20002Negotiate and use G.729 Codec
From a CPE Phone, make a call to a PSTN phone, and answer the call;
Verify voice cut through on connect;
Check for voice Quality; Use your judgment ...
Negotiate G.729 as the Codec;
Verify Call Ringback heard at Calling Party;
Pass RTP/DTMF both ways;
Keep the call up for at least 90 seconds;
Disconnect the call from Called Party
(Confirm that the Vendor CPE uses RTP Port in the range of 16384 through 32767, for receiving RTP)YESIPLD-PSTNPASS 

Basic Call Tests

Hop-off to PSTN

(G.729 offered as first choice codec, except where specified)

Hop-off to PSTN20003Negotiate and use G.711 Codec
From a CPE Phone, make a call to a PSTN phone, and answer the call;
Verify voice cut through on connect;
Check for voice Quality; Use your judgment ...
Negotiate G.711 as the Codec;
Verify Call Ringback heard at Calling Party;
Pass RTP/DTMF both ways;
Keep the call up for at least 90 seconds;
Disconnect the call from Called Party
(Confirm that the Vendor CPE uses RTP Port in the range of 16384 through 32767, for receiving RTP)YESIPLD-PSTNPASS 

Basic Call Tests

Hop-off to PSTN

(G.729 offered as first choice codec, except where specified)

Hop-off to PSTN20004RTCP  for Round Trip Delay Calculation
From a CPE Phone, make a call to a PSTN phone, and answer the call;
Mute call from CPE Phone; Wait for 30 seconds; Un-mute;
Wait for another 30 seconds; Disconnect the call from Calling Party
Include SIP, RTP and RTCP in the trace
NOTE:  This test is to check for Round Trip Delay, so please make sure that the RTCP Sender Reports are sent in both directions.YESIPLD-PSTNPASS 

Basic Call Tests

Hop-off to PSTN

(G.729 offered as first choice codec, except where specified)

Hop-off to PSTN20005Long duration call:
From a CPE Phone, make a call to a PSTN phone, and answer the call;
Keep the call up for at least one hour;
Check for RTP/DTMF working both ways every 10 to 15 minutesNOIPLD-PSTNPASS 

Basic Call Tests

Hop-off to PSTN

(G.729 offered as first choice codec, except where specified)

Hop-off to PSTN20006No ptime from CPE
CPE sends no ptime in invite;
AT&T sends maxptime of 20 in 18x and OK;
Verify 20 msec G729 payload is sent in both directionsYESIPLD-PSTNNOT SUPPORTED 

Basic Call Tests

Hop-off to PSTN

(G.729 offered as first choice codec, except where specified)

Hop-off to PSTN20007ptime=20 from CPE
CPE sends ptime of 20 in invite;
AT&T sends maxptime of 20 in 18x and OK;
Verify 20 msec G729 payload is sent in both directionsYESIPLD-PSTNPASS 

Basic Call Tests

Hop-off to PSTN

(G.729 offered as first choice codec, except where specified)

Hop-off to PSTN20008ptime=30 from CPE
CPE sends ptime of 30 in invite;
AT&T sends maxptime of 30 in 18x and OK;
Verify 30 msec G729 payload is sent in both directionsYESIPLD-PSTNPASS 

Basic Call Tests

Hop-off to PSTN

(G.729 offered as first choice codec, except where specified)

Hop-off to PSTN20009Confirm Voice Traffic (RTP & Signaling) is assigned to Class Of Service 1 (COS1)
Work with your AT&T AVPN Test CoordinatorNOIPLD-PSTNNOT REQUIRED 

Basic Call Tests

Hop-off to PSTN

(G.729 offered as first choice codec, except where specified)

Hop-off to PSTN20010Run the test call with compressed RTP (cRTP) turned on
Work with your AT&T AVPN Test CoordinatorYESIPLD-PSTNNOT REQUIRED 

Basic Call Tests

Hop-off to PSTN

(G.729 offered as first choice codec, except where specified)

Hop-off to PSTN20011Determine bandwidth per call (RTP & Signaling) with data congestion;
Run test with cRTP turned on
Work with your AT&T AVPN Test CoordinatorNOIPLD-PSTNNOT REQUIRED 

Basic Call Tests

Hop-off to PSTN

(G.729 offered as first choice codec, except where specified)

Hop-off to PSTN20012Determine bandwidth per call (RTP & Signaling) with data congestion;
Run test with cRTP turned off
Work with your AT&T AVPN Test CoordinatorNOIPLD-PSTNNOT REQUIRED 

Basic Call Tests

Hop-on from PSTN

(G.729 offered as first choice codec, except where specified)

Hop-on from PSTN20021Call Ringback
From a PSTN phone, make a call to a CPE Phone;
Do not answer Called Party Phone.  Verify the following:
1)  Calling Party hears ringback.
Hang up the PSTN phone while the call is still ringing at Called Party (CPE Phone)
2)  Verify Called Party Phone (CPE Phone) stops ringing and the call is cleared.NOLOCALPASS 

Basic Call Tests

Hop-on from PSTN

(G.729 offered as first choice codec, except where specified)

Hop-on from PSTN20022Negotiate and use G.729 Codec
From a PSTN phone, make a call to a CPE Phone, and answer the call;
Verify voice cut through on connect;
Check for voice Quality; Use your judgment ...
Negotiate G.729 as the Codec;
Verify Call Ringback heard at Calling Party;
Pass RTP/DTMF both ways;
Keep the call up for at least 90 seconds;
Disconnect the call from Called Party
(Confirm that the Vendor CPE uses RTP Port in the range of 16384 through 32767, for receiving RTP)NOLOCALPASS 

Basic Call Tests

Hop-on from PSTN

(G.729 offered as first choice codec, except where specified)

Hop-on from PSTN20023Negotiate and use G.711 Codec
From a PSTN phone, make a call to a CPE Phone, and answer the call;
Verify voice cut through on connect;
Check for voice Quality; Use your judgment ...
Negotiate G.711 as the Codec;
Verify Call Ringback heard at Calling Party;
Pass RTP/DTMF both ways;
Keep the call up for at least 90 seconds;
Disconnect the call from Called Party
(Confirm that the Vendor CPE uses RTP Port in the range of 16384 through 32767, for receiving RTP)NOLOCALPASS 

Basic Call Tests

Hop-on from PSTN

(G.729 offered as first choice codec, except where specified)

Hop-on from PSTN20024RTCP  for Round Trip Delay Calculation
From a PSTN phone, make a call to a CPE Phone, and answer the call;
Mute call from CPE Phone; Wait for 30 seconds; Un-mute;
Wait for another 30 seconds; Disconnect the call from Calling Party
Include SIP, RTP and RTCP in the trace
NOTE:  This test is to check for Round Trip Delay, so please make sure that the RTCP Sender Reports are sent in both directions.YESLOCALPASS 

Basic Call Tests

Hop-on from PSTN

(G.729 offered as first choice codec, except where specified)

Hop-on from PSTN20025Long duration call:
From a PSTN phone, make a call to a CPE Phone, and answer the call;
Keep the call up for at least one hour;
Check for RTP/DTMF working both ways every 10 to 15 minutesNOLOCALPASS 

Basic Call Tests

Hop-on from PSTN

(G.729 offered as first choice codec, except where specified)

Hop-on from PSTN20026No ptime from CPE
AT&T sends maxptime of 30 in invite
CPE sends no ptime in 18x and OK;
Verify 20 msec G729 payload is sent in both directionsYESLOCALNOT SUPPORTED 

Basic Call Tests

Hop-on from PSTN

(G.729 offered as first choice codec, except where specified)

Hop-on from PSTN20027ptime=20 from CPE
AT&T sends maxptime of 30 in invite;
CPE sends ptime of 20 in 18x and OK;
Verify 20 msec G729 payload is sent in both directionsYESLOCALPASS 

Basic Call Tests

Hop-on from PSTN

(G.729 offered as first choice codec, except where specified)

Hop-on from PSTN20028ptime=30 from CPE
AT&T sends maxptime of 30 in invite;
CPE sends ptime of 30 in 18x and OK;
Verify 30 msec G729 payload is sent in both directionsYESLOCALPASS 

Basic Call Tests

Hop-on from PSTN

(G.729 offered as first choice codec, except where specified)

Hop-on from PSTN20029Confirm Voice Traffic (RTP & Signaling) is assigned to Class Of Service 1 (COS1)
Work with your AT&T AVPN Test CoordinatorNOLOCALNOT REQUIRED 

Basic Call Tests

Hop-on from PSTN

(G.729 offered as first choice codec, except where specified)

Hop-on from PSTN20030Run the test call with compressed RTP (cRTP) turned on
Work with your AT&T AVPN Test CoordinatorYESLOCALNOT REQUIRED 

Basic Call Tests

Hop-on from PSTN

(G.729 offered as first choice codec, except where specified)

Hop-on from PSTN20031Determine bandwidth per call (RTP & Signaling) with data congestion;
Run test with cRTP turned on
Work with your AT&T AVPN Test CoordinatorNOLOCALNOT REQUIRED 

Basic Call Tests

Hop-on from PSTN

(G.729 offered as first choice codec, except where specified)

Hop-on from PSTN20032Determine bandwidth per call (RTP & Signaling) with data congestion;
Run test with cRTP turned off
Work with your AT&T AVPN Test CoordinatorNOLOCALNOT REQUIRED International CallHop-off to PSTN20091TESTING USING AT&T PRODUCTION CIRCUIT:
You may use any International Phone Number.
TESTING USING AT&T VIT LAB CIRCUIT:
The only International Number you may dial is the IBM Support Line in Europe (011 41 583330158)
Make the International and Verify / perform the following:
NOTE:  For Enhanced IP Flex CPE Site, you will hear prompt to enter Authorization Code.  Enter valid Authorization Code followed by #.
1) Caller hears Ringback
2) Voice cut through on connect
3) Check for Voice Quality; Use your judgment …
4) Send RTP/DTMF from CPE phone
5) Keep the call up for at least 30 seconds
6) Disconnect the callYESIPLD-PSTNPASS Simultaneous CallsHop-off to PSTN20101Setup at least two OUTBOUND (CPE to PSTN) calls;
Keep calls on for 2 min;
Initiate disconnect for one call from Called Party;
and for the other call, from Calling Party.NOIPLD-PSTNPASS Simultaneous CallsHop-on from PSTN20102Setup at least two INBOUND (PSTN to CPE) calls;
Keep calls on for 2 min;
Initiate disconnect for one call from Called Party;
and for the other call, from Calling Party.NOLOCALPASS Simultaneous CallsHop-off to PSTN
Hop-on from PSTN20103Setup at least one OUTBOUND and one INBOUND calls;
Keep calls on for 2 min;
Initiate disconnect for one call from Called Party;
and for the other call, from Calling Party.NOLOCALPASS Calling Name DeliveryHop-off to PSTN20111From a CPE Phone make a call to some PSTN pone;
Pass Display Name;
Verify display at calling and called partiesNOIPLD-PSTNPASS Calling Name DeliveryHop-on from PSTN20112From a PSTN phone make a call to a CPE Phone;
Pass display name;
Verify display at calling and called partiesNOLOCALPASS 

Calling Number

Privacy

Hop-off to PSTN20121From a CPE Phone make a call to some PSTN phone;
Pass Calling Party Number (CPN), marked private;
Verify display at called party phoneYESIPLD-PSTNPASS 

Calling Number

Privacy

Hop-on from PSTN20122From some PSTN Phone make a call to a CPE phone;
Pass Calling Party Number (CPN), marked private;
Verify display at called party phoneNOLOCALPASS Call Hold and ResumeHop-off to PSTN20141From a CPE Phone make a call to some PSTN phone;
Perform Hold and Resume at both ends
If the PBX supports Music On Hold (MOH), test the HOLD with MOHNOIPLD-PSTNPASS Call Hold and ResumeHop-on from PSTN20142From a PSTN phone make a call to a CPE Phone;
Perform Hold and Resume at both ends
If the PBX supports Music On Hold (MOH), MUST test the HOLD with MOHYESLOCALPASS Voicemail TestsHop-off to PSTN20151From a CPE Phone make a call to some PSTN phone with voicemail set up, and let the call go to voicemail;
Leave voicemail / Pass DTMF;
Retrieve voicemail, by making a 2nd callYESIPLD-PSTNPASS Voicemail TestsHop-on from PSTN20152From a PSTN phone make a call to a CPE Phone with voicemail set up, and let the call go to voicemail;
Leave voicemail / Pass DTMF;
Retrieve voicemail, by making a 2nd callYESLOCALPASS 

PBX-Based

3-Way Call Conference

Intra-Site, Inter-Site and involving PSTN

Hop-off to PSTN20161From a CPE Phone1, make a call to CPE Phone2;
From the CPE Phone1 conference (add) a PSTN phone;
Keep the 3-way confrence up for at least two minutes;
Verify that all participants can hear each otherNOIPLD-PSTNPASS 

PBX-Based

3-Way Call Conference

Intra-Site, Inter-Site and involving PSTN

Hop-off to PSTN20162From a CPE Phone1, make a call to a PSTN phone;
From the CPE Phone1 conference (add) CPE Phone2;
Keep the 3-way confrence up for at least two minutes;
Verify that all participants can hear each otherNOIPLD-PSTNPASS 

PBX-Based

3-Way Call Conference

Intra-Site, Inter-Site and involving PSTN

Hop-on from PSTN20163From a PSTN phone make a call to a CPE Phone1;
From the CPE Phone1 conference (add) CPE Phone2;
Keep the 3-way confrence up for at least two minutes;
Verify that all participants can hear each otherYESLOCALPASS 

PBX-Based

Unattended

Call Transfer

(No REFER from CPE)

Hop-off to PSTN20171From a CPE Phone1, make a call to CPE Phone2;
Initiate the call transfer from CPE Phone1 to some PSTN phone, AFTER Ringback is heard but BEFORE the call is answered at the PSTN phone;
Verify RTP/DTMF both ways;
Keep the call up for at least 60 seconds;
Initiate disconnect from the CPE Phone1NOIPLD-PSTNPASS 

PBX-Based

Unattended

Call Transfer

(No REFER from CPE)

Hop-off to PSTN20172From a CPE Phone1, make a call to a PSTN phone;
Initiate the call transfer from CPE Phone1 to CPE Phone2, AFTER Ringback is heard but BEFORE the call is answered at CPE Phone2;
Verify RTP/DTMF both ways;
Keep the call up for at least 60 seconds;
Initiate disconnect from the PSTN phoneNOIPLD-PSTNPASS 

PBX-Based

Unattended

Call Transfer

(No REFER from CPE)

Hop-on from PSTN20173From a PSTN phone make a call to a CPE Phone1;
Initiate the call transfer from CPE Phone1 to CPE Phone2, AFTER Ringback is heard but BEFORE the call is answered at the CPE Phone2;
Verify RTP/DTMF both ways;
Keep the call up for at least 60 seconds;
Initiate disconnect from the PSTN phoneYESLOCALPASS 

PBX-Based

Attended

Call Transfer

(No REFER from CPE)

Hop-off to PSTN20181From a CPE Phone1, make a call to CPE Phone2;
Initiate the call transfer from CPE Phone1 to some PSTN phone, AFTER the call is answered at the PSTN phone;
Verify RTP/DTMF both ways;
Keep the call up for at least 60 seconds;
Initiate disconnect from the PSTN phoneNOIPLD-PSTNPASS 

PBX-Based

Attended

Call Transfer

(No REFER from CPE)

Hop-off to PSTN20182From a CPE Phone1, make a call to a PSTN phone;
Initiate the call transfer from CPE Phone1 to CPE Phone2, AFTER the call is answered at CPE Phone2;
Verify RTP/DTMF both ways;
Keep the call up for at least 60 seconds;
Initiate disconnect from the CPE Phone2NOIPLD-PSTNPASS 

PBX-Based

Attended

Call Transfer

(No REFER from CPE)

Hop-on from PSTN20183From a PSTN phone make a call to a CPE Phone1;
Initiate the call transfer from CPE Phone1 to CPE Phone2, AFTER the call is answered at the CPE Phone2;
Verify RTP/DTMF both ways;
Keep the call up for at least 60 seconds;
Initiate disconnect from the CPE Phone2YESLOCALPASS 

PBX-Based

Call Forwarding

Unconditional

(CFU)

Hop-on from PSTN
Hop-off to PSTN20191Set up one of the TN with PBX-Based Call Forward Unconditional (CFU) advanced feature;  Make sure the Call Forward To number is set to AT&T BVoIP Customer Care - 877-288-8362;
From a PSTN phone, make a call to the CPE Phone (the one that is set up with CFU);
Verify that the Call Forwards to the PSTN endpoint (AT&T BVoIP Customer - 877-288-8362);
Interact with the Voice Prompt using DTMF for at least 30 seconds, then disconnect the call.
If the CPE supports/includes Diversion header, make sure it is a 10-digit TN recognizable by the AT&T IP Flexible Reach Network.  If Diversion header is present it takes presedence over PAI and From headers.YESLOCALPASS 

PBX-Based

Auto Attendant

Hop-on from PSTN20201From a PSTN phone, make a call to CPE Auto Attendant;
Connect to extention via DTMF;
Verify Voice cut through on connect.  Send RTP/DTMF both waysYESLOCALPASS 

PBX-Based

Meet-Me Conference Bridge

Hop-off to PSTN202111st call: From a PSTN phone, make call to
             the CPE Bridge Number
2nd call: From another PSTN phone, make a call to
             the CPE Bridge Number
3rd call: From a CPE Phone, make a call to
             the CPE Bridge Number
Keep the confrence bridge up for at least two minutes;
Verify that all participants can hear each otherNOLOCALPASS 

AT&T IP Teleconferencing

(IPTC)

To AT&T IPTC20221From a CPE Phone1 call to AT&T IP Teleconferencing (IPTC) number (mentioned above);
Enter the Conference as the HOST;
From a CPE Phone2 call to AT&T IP Teleconferencing (IPTC) number (mentioned above);
Enter the conference as PARTICIPANT;
Verify Audio;  Verify Voice Quality - Use your judgment
Verify that callers from CPE Phone1 and Phone2 can hear each other
Execute this test twice, once per each IPTC Numbers mentioned aboveNOIPLD-nonPSTNPASS 

AT&T IP Teleconferencing

(IPTC)

To AT&T IPTC20222From a CPE Phone1 call to AT&T IP Teleconferencing;
Perform Hold and Resume on Phone1NOIPLD-nonPSTNPASS 

AT&T IP Teleconferencing

(IPTC)

To AT&T IPTC20223From a CPE Phone1, make a call to CPE Phone2;
Perform PBX-Based Attended Transfer from CPE Phone1 to IPTC numberNOIPLD-nonPSTNPASS 

AT&T IP Teleconferencing

(IPTC)

To AT&T IPTC20224From a CPE Phone1, make a call to IPTC number;
Perform PBX-Based Attended Transfer from CPE Phone1 to CPE Phone2NOIPLD-nonPSTNPASS 

AT&T IP Teleconferencing

(IPTC)

To AT&T IPTC20225From a CPE Phone1, make a call to CPE Phone2;
Perform PBX-Based 3-way Call Conference, by adding the IPTC numberNOIPLD-nonPSTNNOT SUPPORTEDSkype 2015 does not senf DTMF event to the conference.

AT&T IP Teleconferencing

(IPTC)

To AT&T IPTC20226From a CPE Phone1, make a call to IPTC number;
Perform PBX-Based 3-way Call Conference, by adding CPE Phone2NOIPLD-nonPSTNPASS Advanced Call PrompterHop-off to PSTN20231Make a call to AT&T BVoIP Customer Care @ 877-288-8362
Verify /perform the following:
1) Caller hears Ringback
2) Voice cut through on connect
3) Check for Voice Quality; Use your judgment …
4) Send RTP/DTMF from CPE phone;  Interact with the Voice Prompt using DTMF
5) Keep the call up for at least 30 seconds
6) Disconnect the callYESIPLD-PSTNPASS Operator / N11 DialingHop-off to PSTN20241Testing using AT&T VIT Lab Circuit:  Operator Call (Call to 0) does not work.  Mark as CONDITIONAL PASS.
Testing using AT&T Production Circuit:  For Operator Call (Call to 0), user will hear recorded operator prompt.YESLOCALPASS Operator / N11 DialingHop-off to PSTN20242Testing using AT&T VIT Lab Circuit:  Call to 911 will terminate to a voice response system, within the AT&T Labs.
Testing using AT&T Production Circuit:  Do not run this test until arrangements are made with AT&T product management.YESLOCALPASS Failover TestsHop-off to PSTN20251Make a call that fails on the Primary AT&T IPBE;
Call route through the Secondary AT&T IPBE.
(If the Certification Testing is for AT&T Business In a Box offer, please mark this test as CONDITIONAL PASS, and mention "BIB Testing" in the Comments field)
Instructions on how to test:
Set up Primary AT&T IPBE to some unreachable IP Address (i.e. 1.2.3.4), and Secondary IPBE to the actual AT&T VIT Lab IPBE IP Address.NOIPLD-PSTNPASS Failover TestsHop-on from PSTN20252Acceps incoming calls from Secondary AT&T IPBE
Testing using AT&T VIT Lab Circuit:  AT&T VIT Lab Network does not have 2nd IPBE.  So, no need to run this test.  Mark as CONDITIONAL PASS.
Testing using AT&T Production Circuit:  It is run (executed) by default in Production Network.  Verify from the Call Traces that are collected from the various tests, that some calls are received from the Primary AT&T IPBE and some are received from the Secondary AT&T IPBE. NOIPLD-nonPSTNPASS Failover TestsSIP Trunk Monitoring20253AT&T network sends SIP Options to customer IP trunk;
If you are not seeing periodic SIP OPTIONS messages coming from AT&T Network, please contact your AT&T Representative.
(If the Certification Testing is for AT&T Business In a Box offer, please mark this test as CONDITIONAL PASS, and mention "BIB Testing" in the Comments field)NOIPLD-nonPSTNPASS 

FAX Tests with T.38

(G3 FAX Machine

at CPE Site)

Hop-off to PSTN20301Customer IP Trunk G3 to PSTN  G3YESIPLD-PSTNPASS 

FAX Tests with T.38

(G3 FAX Machine

at CPE Site)

Hop-off to PSTN20302Customer IP Trunk G3 to PSTN  SG3NOIPLD-PSTNCONDITIONAL PASSFax machine has V.34 enabled(SG3) but SBC does not support T.38 version 3

FAX Tests with T.38

(G3 FAX Machine

at CPE Site)

Hop-on from PSTN20303PSTN  G3 to Customer IP Trunk G3YESLOCALPASS 

FAX Tests with T.38

(G3 FAX Machine

at CPE Site)

Hop-on from PSTN20304PSTN  SG3 to Customer IP Trunk G3NOLOCALCONDITIONAL PASSFax machine has V.34 enabled(SG3) but SBC does not support T.38 version 3

FAX Tests with T.38

(SG3 FAX Machine

at CPE Site)

Hop-off to PSTN20323Customer IP Trunk SG3 to PSTN  G3NOIPLD-PSTNCONDITIONAL PASSFax machine has V.34 enabled(SG3) but SBC does not support T.38 version 3

FAX Tests with T.38

(SG3 FAX Machine

at CPE Site)

Hop-off to PSTN20324Customer IP Trunk SG3 to PSTN  SG3NOIPLD-PSTNCONDITIONAL PASSFax machine has V.34 enabled(SG3) but SBC does not support T.38 version 3

FAX Tests with T.38

(SG3 FAX Machine

at CPE Site)

Hop-on from PSTN20325PSTN  G3 to Customer IP Trunk SG3NOLOCALCONDITIONAL PASSFax machine has V.34 enabled(SG3) but SBC does not support T.38 version 3

FAX Tests with T.38

(SG3 FAX Machine

at CPE Site)

Hop-on from PSTN20326PSTN  SG3 to Customer IP Trunk SG3NOLOCALCONDITIONAL PASSFax machine has V.34 enabled(SG3) but SBC does not support T.38 version 3

FAX Tests with G.711

(G3 FAX Machine

at CPE Site)

Hop-off to PSTN20343Customer IP Trunk G3 to PSTN  G3YESIPLD-PSTNPASS 

FAX Tests with G.711

(G3 FAX Machine

at CPE Site)

Hop-off to PSTN20344Customer IP Trunk G3 to PSTN  SG3NOIPLD-PSTNPASS 

FAX Tests with G.711

(G3 FAX Machine

at CPE Site)

Hop-on from PSTN20345PSTN  G3 to Customer IP Trunk G3YESLOCALPASS 

FAX Tests with G.711

(G3 FAX Machine

at CPE Site)

Hop-on from PSTN20346PSTN  SG3 to Customer IP Trunk G3NOLOCALPASS 

FAX Tests with G.711

(SG3 FAX Machine

at CPE Site)

Hop-off to PSTN20363Customer IP Trunk SG3 to PSTN  G3NOIPLD-PSTNPASS 

FAX Tests with G.711

(SG3 FAX Machine

at CPE Site)

Hop-off to PSTN20364Customer IP Trunk SG3 to PSTN  SG3NOIPLD-PSTNPASS 

FAX Tests with G.711

(SG3 FAX Machine

at CPE Site)

Hop-on from PSTN20365PSTN  G3 to Customer IP Trunk SG3NOLOCALPASS 

FAX Tests with G.711 (SG3 FAX Machine at CPE Site)

Hop-on from PSTN20366PSTN  SG3 to Customer IP Trunk SG3NOLOCALPASS 

Network-Based

Locate Me

(Sequential Ringing)

Hop-on from PSTN20401From a PSTN phone, make a call to the CPE Phone which is enabled with Sequential Ring feature. 
Verify the following:
1) Announcement on PSTN phone (Calling Party) is heard that your party is is being located;
2) CPE Phone Rings - Don't answer call at the CPE Phone.  Verify that the CPE Phone stops ringing after 3 to 4 rings, and the call at the CPE Phone is cleared, but the Calling party keeps hearing ring / announcement;
3) Call rings at anothe CPE Phone - Answer the phone at this CPE Phone.  Verify that you are prompted to enter any key of the keypad, to confirm the answer of the call at the CPE Phone on which the call is answered;
4) Verify RTP/DTMF both waysYESEnhancedPASS 

Network Based

Simultaneous Ringing

Hop-on from PSTN20411From a PSTN phone, make a call to the CPE Phone which is enabled with Simultaneous Ring feature.
Verify the following:
1) Ringback is heard on the PSTN phone w/o any announcement;
2) More than one CPE Phone(s) (called CPE Phone + any other CPE Phone(s) that are provisioned to ring simultaneously for the called CPE Phone) Ring - Answer call at the CPE Phone other than the one provisioned with Simultaneous Ringing feature;  Verify that the other CPE Phone(s) stop ringing;
3) Verify RTP/DTMF both waysYESEnhancedPASS 

Network Based

Simultaneous Ringing

Hop-on from PSTN20412From a PSTN phone, make a call to the CPE Phone which is enabled with Simultaneous Ring feature.
Verify the following:
1) Ringback is heard on the PSTN phone w/o any announcement;
2) More than one CPE Phone(s) (called CPE Phone + any other CPE Phone(s) that are provisioned to ring simultaneously for the called CPE Phone) Ring - Answer call at the CPE Phone that is provisioned with Simultaneous Ringing feature;  Verify that the other CPE Phone(s) stop ringing;
3) Verify RTP/DTMF both waysYESEnhancedPASS 

Network-based

Call Forward Unconditional(CFU),  a.k.a.

Call Forward - Always (CFA)

Call gets forwarded always (unconditionally)

Feature Activation20421On CPE Phone 1, enter the Network-based CFA Activation Feature Access Code (*72) and the Call Fwd Destination TN (any PSTN TN).  Example:  *727325556789YESEnhancedPASS 

Network-based

Call Forward Unconditional(CFU),  a.k.a.

Call Forward - Always (CFA)

Call gets forwarded always (unconditionally)

Hop-on from PSTN
Hop-off to PSTN 20422For Testing using Production Circuit:  Enable Ring Splash (Play Ring Reminder on Call Forward) using the Web Portal.  Contact your AT&T Representative, if you need help with this.
For Testing using AT&T VIT Lab Circuit: Ring Splash (Play Ring Reminder on Call Forward) is enabled.
From a PSTN Phone (other than the Call Fwd Destination) call CPE Phone 1.
Verify:  1) Phone 1 hears a short burst of ringing (indicating the call got forwarded); 2) Call rings at  PSTN phone (Call Fwd Destination); 3) Answer the call at PSTN phone; 4) Verify RTP/DTMF both ways.YESEnhancedCONDITIONAL PASS

Play Ring Reminder is activated but PSTN Phone doesn't hear any burst of ringing 

 

Network-based Call Forward Unconditional(CFU),  a.k.a.

Call Forward - Always (CFA)

Call gets forwarded always (unconditionally)

Feature Dectivation20423On CPE Phone 1, enter the Network-based CFA Deactivation Feature Access Code (*73).
Verify user gets a confirmation indication.
Place a call to CPE Phone 1; Verify call rings at Phone 1.YESEnhancedPASS 

Network-Based

Call Forwarding - Busy

(CFB)

Call get forwarded on Busy Response (486 SIP Response from the CPE)

Feature Activation20431On CPE Phone 1, enter the Network-based CFB Activation Feature Access Code (*90) and the Call Fwd Destination TN (any PSTN TN).  Example:  *907325556789NoEnhancedNOT SUPPORTEDSkype 2015 does not send Busy line.

Network-Based

Call Forwarding - Busy

(CFB)

Call get forwarded on Busy Response (486 SIP Response from the CPE)

Hop-on from PSTN
Hop-off to PSTN20432Keep CPE Phone 1 off hook (busy); 
From a PSTN Phone (other than the Call Fwd Destination) call CPE Phone 1.
Verify:  1) Verify that the CPE sends 486 Busy response to AT&T Network;  2) Call rings at  PSTN phone (Call Fwd Destination); 3) Answer the call at PSTN phone; 4) Verify RTP/DTMF both ways.YESEnhancedNOT SUPPORTEDSkype 2015 does not send Busy line.

Network-Based

Call Forwarding - Busy

(CFB)

Call get forwarded on Busy Response (486 SIP Response from the CPE)

Feature Deactivation20433On CPE Phone 1, enter the Network-based CFB Deactivation Feature Access Code (*91).
Verify user gets a confirmation indication.
Place a call to CPE Phone 1; Verify call rings at Phone 1.NoEnhancedNOT SUPPORTEDSkype 2015 does not send Busy line.

Network-Based Call Forwarding - Ring No Answer(CFRNA)

Call get forwarded on No Answer

Feature Activation20441On CPE Phone 1, enter the Network-based CFRNA Activation Feature Access Code (*92) and the Call Fwd Destination TN (any PSTN TN).  Example:  *927325556789NoEnhancedPASS 

Network-Based Call Forwarding - Ring No Answer(CFRNA)

Call get forwarded on No Answer

Hop-on from PSTN
Hop-off to PSTN20442From a PSTN Phone (other than the Call Fwd Destination) call CPE Phone 1.
Verify:  1) Call rings at  CPE Phone 1;  2) DO NOT Answer the call on CPE Phone 1;  3) CPE Phone 1 stops ringing after 3 or 4 rings;  4) PSTN phone (Call Fwd Destination) rings; 5) Answer the call at PSTN phone; 6) Verify RTP/DTMF both ways.YESEnhancedPASS 

Network-Based Call Forwarding - Ring No Answer(CFRNA)

Call get forwarded on No Answer

Feature Deactivation20443On CPE Phone 1, enter the Network-based CFB Deactivation Feature Access Code (*93).
Verify user gets a confirmation indication.
Place a call to CPE Phone 1; Verify call rings at Phone 1.NoEnhancedPASS 

Network-Based Call Forwarding - Not Reachable(CF-NR)

Call gets forwarded on any 4xx, 5xx or 6xx other than 486 or 600 SIP Response from the CPE

Feature Activation20451On CPE Phone 1, enter the Network-based CF-NR Activation Feature Access Code (*94) and the Call Fwd Destination TN (any PSTN TN).  Example:  *947325556789NoEnhancedPASS 

Network-Based Call Forwarding - Not Reachable(CF-NR)

Call gets forwarded on any 4xx, 5xx or 6xx other than 486 or 600 SIP Response from the CPE

Hop-on from PSTN
Hop-off to PSTN20452Perform necessary set up on CPE to return 403 or 603 for any calls to CPE Phone 1. 
From a PSTN Phone (other than the Call Fwd Destination) call CPE Phone 1.
Verify:  1) CPE sends one of 4xx, 5xx, or 6xx SIP Response, other than 486 or 600; 2) Call rings at  PSTN phone (Call Fwd Destination); 3) Answer the call at PSTN phone; 4) Verify RTP/DTMF both ways.YESEnhancedCONDITIONAL PASSSkype Server is sending SIP Ringing/Session progress before it sends SIP 603

Network-Based Call Forwarding - Not Reachable(CF-NR)

Call gets forwarded on any 4xx, 5xx or 6xx other than 486 or 600 SIP Response from the CPE

Feature Deactivation20453On CPE Phone 1, enter the Network-based CF-NR Deactivation Feature Access Code (*95).
Verify user gets a confirmation indication.
Place a call to CPE Phone 1; Verify call rings at Phone 1.NoEnhancedPASS 

Network-Based Blind Call Transfer

CPE must support sending REFER for the Network-Based Call Transfer

Hop-on from PSTN20461Review AT&T SIP Interface Specification Document for details regarding the SIP Signaling Flow.
Make a call from a PSTN phone to a CPE Phone;
Initiate a blind transfer from the CPE Phone to another (different) PSTN phone (Target Party), by sending REFER SIP Request (with Refer-To header) to AT&T Network.
NOTE:  Please note that for this test to PASS, the transfer request has to be be initiated by the CPE, by sending a REFER request to AT&T Network.  The AT&T Network will perform the actual transfer.YESEnhancedNOT SUPPORTED 

Network-Based Consultative Call Transfer(Attended)

CPE must support sending REFER for the Network-Based Call Transfer

Hop-on from PSTN
Hop-off to PSTN20471Review AT&T SIP Interface Specification Document for details regarding the SIP Signaling Flow.
Make a call from a PSTN phone to a CPE Phone;
Initiate a call from the CPE Phone to another (different) PSTN phone (Target Party);
Initiate transfer AFTER the call is answered by the Target Party, by sending REFER SIP Request (with Refer-To header with Replaces parameter) to AT&T Network.
NOTE:  Please note that for this test to PASS, the transfer request has to be be initiated by the CPE, by sending a REFER request to AT&T Network.  The AT&T Network will perform the actual transfer.YESEnhancedNOT SUPPORTED 

Network-Based Consultative Call Transfer(Unattended)

CPE must support sending REFER for the Network-Based Call Transfer

Hop-on from PSTN
Hop-off to PSTN20481Review AT&T SIP Interface Specification Document for details regarding the SIP Signaling Flow.
Make a call from a PSTN phone to a CPE Phone;
Initiate a call from the CPE Phone to another (different) PSTN phone (Target Party);
Initiate transfer AFTER hearing the Ringback but BEFORE the call is answered by the Target Party, by sending REFER SIP Request (with Refer-To header with Replaces parameter) to AT&T Network.
NOTE:  Please note that for this test to PASS, the transfer request has to be be initiated by the CPE, by sending a REFER request to AT&T Network.  The AT&T Network will perform the actual transfer.YESEnhancedNOT SUPPORTED 

Account Code and Authorization Code

Hop-off to PSTN20491Make a call to a PSTN phone, from the CPE Phone that is provisioned with Mandatory for Account Code
You will hear prompt to enter Account Code.  Enter valid Account Code.
Verify / perform the following:
1) Caller hears Ringback
2) Voice cut through on connect
3) Check for Voice Quality; Use your judgment …
4) Send RTP/DTMF from CPE phone
5) Keep the call up for at least 30 seconds
6) Disconnect the callYESEnhancedPASS Account Code and Authorization CodeHop-off to PSTN20492Make a call to a PSTN phone, from the CPE Phone that is provisioned with Optional for Account Code
Verify / perform the following:
1) Caller hears Ringback
2) Voice cut through on connect
3) Check for Voice Quality; Use your judgment …
4) Send RTP/DTMF from CPE phone
5) Keep the call up for at least 30 seconds
6) Disconnect the callNOEnhancedPASS Account Code and Authorization CodeHop-off to PSTN20493From the CPE Phone that is provisioned with Optional for Account Code, enter Feature Access Code (*71) to receive Prompt to enter Account Code;
Enter the valid Account Code;
Once the valid Account Code is entered, you will hear a confirmation, followed by 2nd dial-tone;
Enter a PSTN phone number to make a call
Verify / perform the following:
1) Caller hears Ringback
2) Voice cut through on connect
3) Check for Voice Quality; Use your judgment …
4) Send RTP/DTMF from CPE phone
5) Keep the call up for at least 30 seconds
6) Disconnect the callYESEnhancedPASS Account Code and Authorization CodeHop-off to PSTN20494From the CPE Phone that is provisioned with Optional for Account Code, enter Feature Access Code (*71) plus valid Account Code plus a PSTN phone number.  For example, *711007325556789.
Verify / perform the following:
1) Caller hears Ringback
2) Voice cut through on connect
3) Check for Voice Quality; Use your judgment …
4) Send RTP/DTMF from CPE phone
5) Keep the call up for at least 30 seconds
6) Disconnect the callYESEnhancedPASS Account Code and Authorization CodeHop-off to PSTN20495Make an international call to the IBM Support Line in Europe (011 41 583330158), from the CPE Phone that is provisioned with Optional for Account Code
You will hear prompt to enter Authorization Code.  Enter valid Authorization Code followed by #.
Verify / perform the following:
1) Caller hears Ringback
2) Voice cut through on connect
3) Check for Voice Quality; Use your judgment …
4) Send RTP/DTMF from CPE phone
5) Keep the call up for at least 30 seconds
6) Disconnect the callYESEnhancedPASS Account Code and Authorization CodeHop-off to PSTN20496Make an international call to the IBM Support Line in Europe (011 41 583330158), from the CPE Phone that is provisioned with Mandatory for Account Code
You will hear prompt to enter Authorization Code.  Enter valid Authorization Code followed by #.
You will hear prompt to enter Account Code.  Enter valid Account Code.
Verify / perform the following:
1) Caller hears Ringback
2) Voice cut through on connect
3) Check for Voice Quality; Use your judgment …
4) Send RTP/DTMF from CPE phone
5) Keep the call up for at least 30 seconds
6) Disconnect the callYESEnhancedPASS 
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Conclusion

This Application Note describes the configuration steps required for the Sonus SBC 1000/2000 to successfully interoperate with Skype for Business 2015 server and AT&T IP Flexible Reach SIP Trunk. All feature and serviceability test cases were completed and passed with the exceptions/observations noted in InteroperabilityTest Results Not Supported Section.

 

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