Hello, can you show both received 200ok + ACK as well as those sent out? It is important to see how Record-/Route, Contact and r-uri change on the way to spot where the issue is.
Cheers, Daniel On 12/05/15 05:56, Darren Campbell (Primar) wrote: > Hi all > > Experiencing a commonly reported issue where calls drop out after 30 > seconds or so. Mainly because the provider hangs up after not > recognising/receiving ACK in response to 200 OK. > > Unfortunately (or maybe fortunately), I haven't had much experience > with Enswitch so was hoping someone in the community might help guide > me as to which rules Enswitch might be using to match ACKs to calls in > progress. Maybe there is another avenue I should be investigating. > > > Here's a sample of the 200 OK and ACK that repeats. > > 13:44:04.155646 IP PROVIDERIP.5060 > 172.21.0.226.5060: SIP, length: 1058 > E..>.M..?..Ug.v..........*J.SIP/2.0 200 OK^M > Via: SIP/2.0/UDP > 172.21.0.226;rport=5060;branch=z9hG4bKfe94.efbf7fbcaf8bd15243a61fdc9d6d1e78.0^M > Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK65f00a0c;rport=5080^M > Record-Route: <sip:PROVIDERIP;lr=on>^M > Record-Route: <sip:172.21.0.226;r2=on;lr=on;ftag=as65919d92;nat=yes>^M > Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as65919d92;nat=yes>^M > From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as65919d92^M > To: <sip:PHONENUMBER@PROVIDERIP>;tag=as260fefaa^M > Call-ID: 271ac7a174d613cd0b94504353733a2c@PROVIDERIP^M > CSeq: 103 INVITE^M > Server: Enswitch^M > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH^M > Supported: replaces^M > Contact: <sip:PHONENUMBER@PROVIDERMEDIAIP:5060>^M > Content-Type: application/sdp^M > Content-Length: 286^M > ^M > v=0^M > o=root 2110894460 2110894461 IN IP4 PROVIDERMEDIAIP^M > s=Asterisk PBX 11.3.0^M > c=IN IP4 PROVIDERMEDIAIP^M > t=0 0^M > m=audio 15594 RTP/AVP 0 8 3 101^M > a=rtpmap:0 PCMU/8000^M > a=rtpmap:8 PCMA/8000^M > a=rtpmap:3 GSM/8000^M > a=rtpmap:101 telephone-event/8000^M > a=fmtp:101 0-16^M > a=ptime:20^M > a=sendrecv^M > > 13:44:04.164519 IP 172.21.0.226.5060 > PROVIDERIP.5060: SIP, length: 525 > E..)!a...@..v....g.v.......t.ack sip:PHONENUMBER@PROVIDERIP:5060 SIP/2.0^M > Via: SIP/2.0/UDP > 172.21.0.226;branch=z9hG4bKfe94.472e9fc0479de79b4f176cc9585d8880.0^M > Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK752b5264;rport=5080^M > Route: <sip:PROVIDERIP;lr=on>^M > Max-Forwards: 69^M > From: "asterisk" <sip:PROVIDERUSER@PROVIDERIP:5080>;tag=as65919d92^M > To: <sip:PHONENUMBER@PROVIDERIP>;tag=as260fefaa^M > Contact: <sip:PROVIDERUSER@127.0.0.1:5080>^M > Call-ID: 271ac7a174d613cd0b94504353733a2c@PROVIDERIP^M > CSeq: 103 ACK^M > User-Agent: Elastix 3.0^M > Content-Length: 0^M > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com
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