Hi Folks,
We are having issue with SIP calls via CUBE. We have SCCP phones and SIP trunk to 2 CUBE routers. Issue is when the call lands on CUBE 1 it goes to CUCM-1 and user answers the phone. On the IP-Phone it answer but on the mobile phone it still keeps on ringing. From the CUBE logs i see CUCM-1 didn't send 200 OK message.
Since we are using Enterprise Trunk then the same call is forwarded by service provider on CUBE 2 and this time the call goes to same CUCM-1 again but this time it responds with 200 OK message.
I looked at the two Invite messages, and i only see a difference in terms of capset, not sure why capset are different, however i don't know if that's actually the issue.
Below are the 2 invite messages.
Non working Invite
==============
INVITE sip:5141@10.31.16.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.129.67.1:5060;branch=z9hG4bK3C614D21E1 From: <sip:00424428812@domain.com>;tag=56E85556-999 To: <sip:5141@10.31.16.2> Date: Tue, 29 May 2018 04:34:39 GMT Call-ID: 6F03E96D-623011E8-B40EAF92-244BDF9C@10.129.67.1 Supported: timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 1862487213-1647317480-3020468114-0608952220 User-Agent: Cisco-SIPGateway/IOS-15.5.3.S4b Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1527568479 Contact: <sip:00424428812@10.129.67.1:5060> Expires: 300 Allow-Events: telephone-event Max-Forwards: 28 P-Asserted-Identity: <sip:00424428812@10.129.67.1> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 308
v=0 o=CiscoSystemsSIP-GW-UserAgent 2577 5910 IN IP4 10.129.67.1 s=SIP Call c=IN IP4 10.129.67.1 t=0 0 m=audio 0 RTP/AVP 8 0 18 97 c=IN IP4 10.129.67.1 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=ptime:20
Working invite
===========
INVITE sip:5141@10.31.16.2:5060 SIP/2.0 Via: SIP/2.0/UDP 10.31.16.10:5060;branch=z9hG4bK2C13AA2001 From: <sip:00424428812@domain.com>;tag=70CB8180-145F To: <sip:5141@10.31.16.2> Date: Tue, 29 May 2018 04:16:22 GMT Call-ID: E0E10CD4-622D11E8-A4F391CE-316F73CA@10.31.16.10 Supported: timer,resource-priority,replaces,sdp-anat Min-SE: 1800 Cisco-Guid: 3772765164-1647120872-2767032782-0829387722 User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1527567382 Contact: <sip:00424428812@10.31.16.10:5060> Expires: 300 Allow-Events: telephone-event Max-Forwards: 28 P-Asserted-Identity: <sip:00424428812@10.31.16.10> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 349
v=0 o=CiscoSystemsSIP-GW-UserAgent 4450 6324 IN IP4 10.31.16.10 s=SIP Call c=IN IP4 10.31.16.10 t=0 0 m=audio 20052 RTP/AVP 8 0 18 100 97 c=IN IP4 10.31.16.10 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15
In the non-working scenario i see below error when AU connect process is going on.
91598591.001 |14:29:22.355 |AppInfo |DET-SDPMsg-setSdpLineInfoMap: MLINE_AUDIO numMlines= 1 91598591.002 |14:29:22.355 |AppInfo |DET-SIPInterface-(83331)::BFCP is disabled:Filter out Presentation Video and BFCP 91598591.003 |14:29:22.355 |AppInfo |!!ERROR!! -SDPMsg::-getIPAddrTypeOfAudio(): AUDIO mLineList has size 0 91598591.004 |14:29:22.355 |AppInfo |!!ERROR!! -SDPMsg- - getAudioMLine, count=0 but idx=0, returning System Default 91598591.005 |14:29:22.355 |AppInfo |DET-SDPMsg-getBWbpsFromSession: 0 0,0,0 91598591.006 |14:29:22.355 |AppInfo |DET-SDPMsg-getBWbpsFromSession: No session level bw modifiers! 91598591.007 |14:29:22.355 |AppInfo |!!ERROR!! -SDPMsg- - getAudioMLine, count=0 but idx=0, returning System Default 91598591.008 |14:29:22.355 |AppInfo |DET-SDPMsg-()::setTIASforDO, sessionBWbps=64000, receivedSessionBWbps=0, calculatedAudioTIAS=64000 91598591.009 |14:29:22.355 |AppInfo |DET-SDPMsg-()::setSessionBandwidthModifiers(bw=64000, action=0) - session(bitmask=0x0,tias=0,as=0,ct=0) 91598591.010 |14:29:22.355 |AppInfo |!!ERROR!! -MediaMLinesList-::getSDPMode, ERROR: Invalid Size or Line Index Size=0, Line Index=0 91598591.011 |14:29:22.355 |AppInfo |DET-SIPInterface-(83331)::waitSDPResponse_SDPOfferInd, ignoring offer with 0 audio m-lines 91598592.000 |14:29:22.355 |SdlSig |SIPMediaToUpdateCc |waitForAnswerorOfferorMXCap |SIPInterface(2,100,69,83331)
Any idea what does the error message corresponds to?
Regards,
Alok
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