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New Member

Issue - Cisco Voice Router and Lync SIP Integration

Hi,

Australian Carrier services.

We have integrated a Cisco ISR with a Lync SBA running on a SRE module using SIP.  The ISR is configured to provide E1 PRI ISDN access for the Lync SBA.  Dial Peer has been configured to forward all calls out the PSTN throught the Serial Interface.  All local, national and international and 1800 calls are working fine.  However, certain 1300 numbers are not working and some 1300 are working.  We have tried to isolate a common pattern but cannot understand why some 1300 numbers are not working and others are.  We have used Dilogic gateway on the Lync server connected to the same ISDN service all 1300 numbers work.  This would suggest that the Cisco Voice router is passing certain malformed information that the SIP trunk is passing to the Lync SBA and cancelling the call.

All 1300 numbers are begin sent from the Lync server to the Cisco Router but some 1300 calls are not establishing. 

Debuging the dial peers show a match to the correct dial peer. Debuging the ISDN Q931 show the call leaving and that the is Normally cleared.  Debuging CCSIP suggests that the call is ACK after that the call fails.  This suggests that a codec mismatch could be occuring. The call is being sent to the carrier but I think that something at the recipent end is sending bad information through the SIP trunk to the Lync server which terminates the call.

Can anyone assist in further troubleshooting or may have come across a CUBE ITSP issue which required addtional debuging.

CME or UCM is not used in this topology.  A CUBE licence has been installed.

Cheers

1 ACCEPTED SOLUTION

Accepted Solutions
New Member

Issue - Cisco Voice Router and Lync SIP Integration

Hi,

When you issue a debug q931 and monitor a call, what is the desination number within the call being sent to the pstn? Have you compared a working and a non-working call for anything within the q931 messages?

Also, look into embedded packet capture on the Cisco router. Set up a capture of your working and non working call and look at the plain sip messages to see if anything stands out.

Hopefully this points in the right direction.

5 REPLIES
New Member

Issue - Cisco Voice Router and Lync SIP Integration

Hi,

When you issue a debug q931 and monitor a call, what is the desination number within the call being sent to the pstn? Have you compared a working and a non-working call for anything within the q931 messages?

Also, look into embedded packet capture on the Cisco router. Set up a capture of your working and non working call and look at the plain sip messages to see if anything stands out.

Hopefully this points in the right direction.

New Member

Re: Issue - Cisco Voice Router and Lync SIP Integration

Thanks Tony, I will get some captures and upload If I cannot identify the issue.

Sent from Cisco Technical Support iPhone App

New Member

Issue - Cisco Voice Router and Lync SIP Integration

Does this help?

 
Call Flow: Lync server>SIP>2911>PRI>PSTN
 
--> We collected the following debugs for both non-working and working call.

 

 
Here is the debug analysis:
 
Incoming setup
 
Received: 
INVITE sip:1300546538@10.3.201.40;user=phone SIP/2.0
FROM: "Lync03
Testing"<sip:+61732512200@SRVVALSBA03.workpac.com;user=phone>;epid=E1E486546D;tag=31

2fe624e0
TO: <sip:1300546538@10.3.201.40;user=phone>
CSEQ: 30473 INVITE
CALL-ID: 3c9e14d3-d2dc-4e23-8009-02e0f6b07cc0
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 10.3.201.32:52122;branch=z9hG4bK7878dde8
CONTACT:


be6c0 >
CONTENT-LENGTH: 339
SUPPORTED: 100rel
USER-AGENT: RTCC/4.0.0.0 MediationServer
CONTENT-TYPE: application/sdp
ALLOW: ACK
Allow: CANCEL,BYE,INVITE,PRACK,UPDATE
 
v=0
o=- 11654 1 IN IP4 10.3.201.32
s=session
c=IN IP4 10.3.201.32
b=CT:1000
t=0 0
m=audio 49924 RTP/AVP 97 101 13 0 8
c=IN IP4 10.3.201.32
a=rtcp:49925
a=label:Audio
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
 
Outgoing ISDN setup
 
May 22 08:42:46.444: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8  callref = 0x2368 
        Bearer Capability i = 0x8090A3 
                Standard = CCITT 
                Transfer Capability = Speech  
                Transfer Mode = Circuit 
                Transfer Rate = 64 kbit/s 
        Channel ID i = 0xA98381 
                Exclusive, Channel 1 
        Calling Party Number i = 0x0180, '+61732512200' 
                Plan:ISDN, Type:Unknown 
        Called Party Number i = 0x81, '1300546538' 
                Plan:ISDN, Type:Unknown
                                                             
 
Cancel message received from Lync server
 
Received: 
CANCEL sip:1300546538@10.3.201.40;user=phone SIP/2.0
FROM: "Lync03
Testing"<sip:+61732512200@SRVVALSBA03.workpac.com;user=phone>;tag=312fe624e0;epid=E1

E486546D
TO: <sip:1300546538@10.3.201.40;user=phone>
CSEQ: 30473 CANCEL
CALL-ID: 3c9e14d3-d2dc-4e23-8009-02e0f6b07cc0
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 10.3.201.32:52122;branch=z9hG4bK7878dde8
CONTACT: <sip:SRVVALSBA03.workpac.com:5068;transport=Tcp;maddr=10.3.201.32>
CONTENT-LENGTH: 0
USER-AGENT: RTCC/4.0.0.0 MediationServer
 
 
Disconnect message sent to PSTN
 
May 22 08:42:55.984: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8  callref = 0x2368 
        Cause i = 0x8090 - Normal call clearing
                               
New Member

Re: Issue - Cisco Voice Router and Lync SIP Integration

Issue with the Lync timers and carrier headers.

Sent from Cisco Technical Support iPhone App

New Member

Hi,

Hi,

Did you ever find out what this was.

I have a similar issue.

Lync >> SIP >> CUCM >> H323 >> Cisco 2811 >> ISDN E1 >> Telco

all sites work like this except one, where lync send a cancel message before 183 session progress from CUCM to Lync.

Totally stumped why it is only doing this for certain numbers in one site all of a sudden

I have tried with new trunk, reset gateway/e1 controller but cant work out.

Can't work out whether it is cucm garbling the SIP message or Lync just timing out, but all timers in Lync are set correctly as works with other site no worries.

I have raised with provider but they say everything works correctly, and the same calls work on the same gateway is just

Cisco phone >> CUCM >> h323 >> 2811 >> ISDN E1 >> Telco which leads me to believe it is cucms translation from H323 to SIP on that particular trunk but no idea why.

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