Problem with SIP to SIP connections in CME 9.5

c.holloway · ‎03-03-2014

Good morning everyone!

I have a problem with a new deployment of CME that I'm hoping y'all can help me with. We're running CME 9.5 and using Exchange 2013 for voicemail and auto-attendant services. We have a SIP trunk for calls to and from the PSTN and then another trunk to Exchange 2013 for its UM services.

Currently I am unable to call in from the outside and reach any SIP destination from the PSTN. I have tried calling the DID when pointed to the voicemail pilot, the AA pilot, and a SIP DN on a softphone on the network. If I point the DID at a SCCP DN, the call completes without issue. I can also call out from the SIP softphone over the trunk to a SIP endpoint hanging off of our system here at my office.

The behavior of the failed call is the same whether the destination is Exchange or the SIP softphone. Basically I get a loop of SIP Invite, Trying, Moved Temporarily, and ACK until the call times out and goes busy. Below are the debugs from "debug ccsip messages" on these calls. I have also attached pertinent parts of the voice configuration from the router. Some names and numbers have been altered but otherwise this is what we are running.

The router is currently in production but the voice services have yet to be deployed as I'm waiting until I can get voicemail working to roll out the new phone system so I'm free to make any voice changes needed.

Any help/advise would be greatly appreciated.

This debug is for an inbound call on the Provider Trunk to the Auto Attendant Pilot

*Feb 28 21:53:57.509: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:XXXXXXXXXX@10.1.129.254:5060 SIP/2.0

Via: SIP/2.0/UDP 209.209.172.231:5060;branch=z9hG4bKa82dvo209gog57d4h690.1

From: <sip:ZZZZZZZZZZZ@172.16.6.171;user=phone>;tag=1639169081-1393620731871-

To: "Client"<sip:XXXXXXXXXX@SIPTrunkProvider.com>

Call-ID: BW155211871280214-2041632561@172.16.6.171

CSeq: 1025663984 INVITE

Contact: <sip:ZZZZZZZZZZ@209.209.172.231:5060;transport=udp>

Supported: 100rel

Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY

Accept: application/media_control+xml,application/sdp,multipart/mixed

Max-Forwards: 9

Content-Type: application/sdp

Content-Length: 280

v=0

o=Filler 1840125 1 IN IP4 209.209.172.231

s=-

c=IN IP4 209.209.172.231

t=0 0

m=audio 62230 RTP/AVP 0 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

a=bsoft: 1 image udptl t38

SIP: Trying to parse unsupported attribute at media level

*Feb 28 21:53:57.517: //34971/A8928C04B7FB/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 209.209.172.231:5060;branch=z9hG4bKa82dvo209gog57d4h690.1

From: <sip:ZZZZZZZZZZ@172.16.6.171;user=phone>;tag=1639169081-1393620731871-

To: "Client"<sip:XXXXXXXXXX@SIPTrunkProvider.com>

Date: Fri, 28 Feb 2014 21:53:57 GMT

Call-ID: BW155211871280214-2041632561@172.16.6.171

CSeq: 1025663984 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.3.2.T

Content-Length: 0

*Feb 28 21:53:57.517: //34971/A8928C04B7FB/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 209.209.172.231:5060;branch=z9hG4bKa82dvo209gog57d4h690.1

From: <sip:ZZZZZZZZZZ@172.16.6.171;user=phone>;tag=1639169081-1393620731871-

To: "Client"<sip:XXXXXXXXXX@SIPTrunkProvider.com>;tag=8FDC57C8-1C69

Date: Fri, 28 Feb 2014 21:53:57 GMT

Call-ID: BW155211871280214-2041632561@172.16.6.171

CSeq: 1025663984 INVITE

Allow-Events: telephone-event

Warning: 399 10.1.129.254 "Transcoder Not Configured"

Server: Cisco-SIPGateway/IOS-15.3.2.T

Reason: Q.850;cause=47

Content-Length: 0

*Feb 28 21:53:57.565: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:XXXXXXXXXX@10.1.129.254:5060 SIP/2.0

Via: SIP/2.0/UDP 209.209.172.231:5060;branch=z9hG4bKa82dvo209gog57d4h690.1

CSeq: 1025663984 ACK

From: <sip:ZZZZZZZZZZ@172.16.6.171;user=phone>;tag=1639169081-1393620731871-

To: "Client"<sip:XXXXXXXXXX@SIPTrunkProvider.com>;tag=8FDC57C8-1C69

Call-ID: BW155211871280214-2041632561@172.16.6.171

Max-Forwards: 9

Content-Length: 0

Suresh Subramanian · ‎03-03-2014

Hi,

Reason: Q.850;cause=47 indicates Codec mismatch. Please cross check all the Inbound & Outbound dialpeers and its Codec configuration.

What codec does your provider support?

//Suresh Please rate all the useful posts.

Chad Stachowicz · ‎03-03-2014

On top of that... very clearly it also tried to use a transcoder but none was available. I agree, check regions in CUCM + all dial-peers on GW to ensure they are using the proper codec OR voice-class codec for the ability to switch as needed.

You could also hack and toss in a transcoder to be sure... not optimal though, you would rather have good codec matching if possible.

Feb 28 21:53:57.517: //34971/A8928C04B7FB/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 209.209.172.231:5060;branch=z9hG4bKa82dvo209gog57d4h690.1

From: ;tag=1639169081-1393620731871-

To: "Client"<>XXXXXXXXXX@SIPTrunkProvider.com>;tag=8FDC57C8-1C69

Date: Fri, 28 Feb 2014 21:53:57 GMT

Call-ID: BW155211871280214-2041632561@172.16.6.171

CSeq: 1025663984 INVITE

Allow-Events: telephone-event

Warning: 399 10.1.129.254 "Transcoder Not Configured"

Server: Cisco-SIPGateway/IOS-15.3.2.T

Reason: Q.850;cause=47

Content-Length: 0

Chad

c.holloway · ‎03-03-2014

The codec issue is unrelated. I fixed it with the addition of the "voice-class codec 1" command on the SIP dial peers but the same thing happens when calling in

Suresh Subramanian · ‎03-03-2014

Please collect the below debugs for a test call.

- debug voice ccapi inout

- Debug ccsip message

//Suresh Please rate all the useful posts.

c.holloway · ‎03-04-2014

The output of "Debug ccsip message" is above in the original post. The output has not changed with the exception of the codec and transcoder errors no longer being present. I have attached the output of "Debug voice ccapi inout" to the original post. Thank you.

bascheew · ‎09-12-2014

Colin,

I just wanted to report that I was having the same issue and saw that when Exchange was sending the redirect from 5062 to 5065 that it was using the FQDN of Exchange instead of the IP address. I added a host record on the router for the name and everything worked fine after that.

c.holloway · ‎03-06-2014

Suresh,

I was mistaken about which debugs the ones above were. The above CCSIP debugs are from a call from the PSTN to a SIP softphone which are much less important than getting the connection to Exchange working. I have added debugs from a call to Exchange below as per your request. From what I can tell (Im still not very SIP savvy) The SIP conversation between the provider trunk and the router never completes before the SIP conversation between the inbound and Exchange starts. As I said I'm still getting familiar with SIP so maybe this is nothing but it looked a little odd to my eye. Debugs are below

INVITE sip:XXXXXXXXXX@10.1.129.254:5060 SIP/2.0

Via: SIP/2.0/UDP 209.209.172.231:5060;branch=z9hG4bKu4t4v03048p0061dc110.1

From: ;tag=1770100571-1394112257109-

To: "SW Child Dev"<>XXXXXXXXXX@SIPTrunkProvider.com>

Call-ID: BW082417109060314-679806382@172.16.6.141

CSeq: 197684779 INVITE

Contact:

Supported: 100rel

Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY

Accept: application/media_control+xml,application/sdp,multipart/mixed

Max-Forwards: 9

Content-Type: application/sdp

Content-Length: 282

v=0

o=FillerRealm 111344245 1 IN IP4 209.209.172.231

s=-

c=IN IP4 209.209.172.231

t=0 0

m=audio 52898 RTP/AVP 0 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

a=bsoft: 1 image udptl t38

SIP: Trying to parse unsupported attribute at media level

Mar 6 13:25:51.801: //1439/AC26808287F1/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 209.209.172.231:5060;branch=z9hG4bKu4t4v03048p0061dc110.1

From: ;tag=1770100571-1394112257109-

To: "SW Child Dev"<>XXXXXXXXXX@SIPTrunkProvider.com>

Date: Thu, 06 Mar 2014 13:25:51 GMT

Call-ID: BW082417109060314-679806382@172.16.6.141

CSeq: 197684779 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.3.2.T

Content-Length: 0

Mar 6 13:25:51.813: //1440/AC27B96A87F6/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:601@10.1.1.11:5062 SIP/2.0

Via: SIP/2.0/TCP 10.1.129.254:5060;branch=z9hG4bK5B61050

Remote-Party-ID: ;party=calling;screen=no;privacy=off

From: <>ZZZZZZZZZZ@SIPTrunkProvider.com>;tag=CBB8924-2203

To: <601>

Date: Thu, 06 Mar 2014 13:25:51 GMT

Call-ID: AC285592-A46911E3-87FAC945-249872F9@10.1.129.254

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 2888284522-2758349283-2281097541-0613970681

User-Agent: Cisco-SIPGateway/IOS-15.3.2.T

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1394112351

Contact:

Diversion: ;privacy=off;reason=unconditional;counter=1;screen=no

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 8

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 270

v=0

o=CiscoSystemsSIP-GW-UserAgent 5118 5186 IN IP4 10.1.129.254

s=SIP Call

c=IN IP4 10.1.129.254

t=0 0

m=audio 17346 RTP/AVP 18 101

c=IN IP4 10.1.129.254

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

Mar 6 13:25:51.813: //1440/AC27B96A87F6/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

FROM: <>ZZZZZZZZZZ@SIPTrunkProvider.com>;tag=CBB8924-2203

TO: <601>

CSEQ: 101 INVITE

CALL-ID: AC285592-A46911E3-87FAC945-249872F9@10.1.129.254

VIA: SIP/2.0/TCP 10.1.129.254:5060;branch=z9hG4bK5B61050

CONTENT-LENGTH: 0

TIMESTAMP: 1394112351

Mar 6 13:25:52.013: //1440/AC27B96A87F6/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 302 Moved Temporarily

FROM: <>ZZZZZZZZZZ@SIPTrunkProvider.com>;tag=CBB8924-2203

TO: <601>;epid=9D14031D42;tag=38224e65e4

CSEQ: 101 INVITE

CALL-ID: AC285592-A46911E3-87FAC945-249872F9@10.1.129.254

VIA: SIP/2.0/TCP 10.1.129.254:5060;branch=z9hG4bK5B61050

CONTACT: <>601@Exchange-UM.swcdcinc.org:5065;transport=Tcp>

CONTENT-LENGTH: 0

SERVER: RTCC/5.0.0.0

Diversion: ;privacy=off;reason=unconditional;counter=1;screen=no

Mar 6 13:25:52.013: //1440/AC27B96A87F6/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:601@10.1.1.11:5062 SIP/2.0

Via: SIP/2.0/TCP 10.1.129.254:5060;branch=z9hG4bK5B61050

From: <>ZZZZZZZZZZ@SIPTrunkProvider.com>;tag=CBB8924-2203

To: <601>;epid=9D14031D42;tag=38224e65e4

Date: Thu, 06 Mar 2014 13:25:51 GMT

Call-ID: AC285592-A46911E3-87FAC945-249872F9@10.1.129.254

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

Suresh Subramanian · ‎03-06-2014

The call to 601 is negotiating G729 codec.

Sent:

INVITE sip:601@10.1.1.11:5062 SIP/2.0

Via: SIP/2.0/TCP 10.1.129.254:5060;branch=z9hG4bK5B61050

Remote-Party-ID: ;party=calling;screen=no;privacy=off

From: <>ZZZZZZZZZZ@SIPTrunkProvider.com>;tag=CBB8924-2203

To: <601>

Date: Thu, 06 Mar 2014 13:25:51 GMT

Call-ID: AC285592-A46911E3-87FAC945-249872F9@10.1.129.254

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 2888284522-2758349283-2281097541-0613970681

User-Agent: Cisco-SIPGateway/IOS-15.3.2.T

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1394112351

Contact:

Diversion: ;privacy=off;reason=unconditional;counter=1;screen=no

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 8

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 270

v=0

o=CiscoSystemsSIP-GW-UserAgent 5118 5186 IN IP4 10.1.129.254

s=SIP Call

c=IN IP4 10.1.129.254

t=0 0

m=audio 17346 RTP/AVP 18 101

c=IN IP4 10.1.129.254

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

!

could you please configure the voice class codec command on the dial-peer voice 1 and check the behaviour?

//Suresh

Please rate all the useful posts.

//Suresh Please rate all the useful posts.

c.holloway · ‎03-06-2014

Suresh,

I have the voice class codec 1 command on the inbound dial peer and see no change in behavior.

Carlo Poggiarelli · ‎03-06-2014

Hi.

Please apply voice class codec to dialpeer 3 too and check the behaviour if is the same.

After that change please post a debug ccsip message again.

Thanks

Regards

Carlo

Please rate all helpful posts

"The more you help the more you learn"

Please rate all helpful posts "The more you help the more you learn"

c.holloway · ‎03-06-2014

Carlo,

Thank you for taking a look. I updated that dial-peer previously. The last debug ccsip messages I posted includes output with this applied to the dial-peer voice 3

Suresh Subramanian · ‎03-06-2014

Please collect the debug ccsip message and debug voice ccapi inout for a test call. Please collect these debugs together in the buffer and attach the files here.

Sent from Cisco Technical Support Android App

//Suresh Please rate all the useful posts.

c.holloway · ‎03-06-2014

Most recent debugs are now attached as debugs.log

Suresh Subramanian · ‎03-06-2014

Hello,

we see the call is redirected to AA-601 from the dial-peer 3 with codec G711ulaw & G729 and destination port: 5062.

instead of getting 180/183, we are getting 302 Moved Temporarily from AA. could you please crosscheck if that number 601 is properly configured in AA?

Mar 6 17:42:54.917: //2286/94EC31A5954F/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 302 Moved Temporarily

FROM: <>8282858882@morrisbroadband.com>;tag=DA6DECC-122A

TO: <601>;epid=9D14031D42;tag=bfa3d4b9c5

CSEQ: 101 INVITE

CALL-ID: 94ECCDCD-A48D11E3-9553C945-249872F9@10.1.129.254

VIA: SIP/2.0/TCP 10.1.129.254:5060;branch=z9hG4bK6CF1199

CONTACT: <>601@Exchange-UM.swcdcinc.org:5065;transport=Tcp>

CONTENT-LENGTH: 0

SERVER: RTCC/5.0.0.0

Diversion: <8283540105>;privacy=off;reason=unconditional;counter=1;screen=no

Also, the contact field of the above 302 message is showing the port number 5065, but in the dial-peer we have configured 5062 to AA. why is that difference?

//Suresh

Please rate all the useful posts.

//Suresh Please rate all the useful posts.