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
