Our Kamailio statefully forwards requests to our Asterisk cluster. Sip session timers fail, whether our Asterisk servers are UAS or UAC.
Kamailio receives the re(INVITE), and just dispatches it out using the default algorithm. I see the record-route header, so I'm not sure what else is missing.. (aside from the last sip packet showing the packet dispatched to a new asterisk peer) Anybody have some troubleshooting tips for us? Loose_route should match the transaction based on record-route, is my understanding. Same thing happens whether carrier re(INVITE)'s or Asterisk does. Just gets dispatched to the next peer in the list. U kamailio:5060 -> Carrier:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP Carrier:5060;rport=5060;branch=z9hG4bK0cB8c2920fa87e47e82..Record-Route: <sip:kamailio;lr=on<sip:192.168.9.130;lr=on>>..From: <sip:From_User@carrier:50 60>;tag=gK0c5010b1..To: <sip:To_User@kamailio:5060<sip:4704430714@192.168.9.130:5060>>;tag=as7651768b..Call-ID: 638356677_31693...@carrier..cseq<mailto:638356677_31693...@66.62.60.238..cseq>: 10795 INVITE..Server: Asterisk PBX 11.2.1..Allow: IN VITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..Supported: replaces, timer..Session-Expires: 1800;refresher=uac..Contact: <sip:10.10.10.10;redact1=abc fhCOikg7YDp7YWcOYDg7wWgOnkg3YeGKYWf9Yuphcp**>..Content-Type: application/sdp..Require: timer..Content-Length: 255....v=0..o=root 1728271566 1728271566 IN IP4 kamailio..s=A sterisk PBX 11.2.1..c=IN IP4 kamailio..t=0 0..m=audio 61004 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ptime:20..a=sendrecv ..a=nortpproxy:yes.. U Carrier:5060 -> kamailio:5060 ACK sip:10.10.10.10;redact1=abcfhCOikg7YDp7YWcOYDg7wWgOnkg3YeGKYWf9Yuphcp** SIP/2.0..Via: SIP/2.0/UDP Carrier:5060;branch=z9hG4bK0cB8c3710b32c17351f..From: <sip:From_User 6@Carrier:5060>;tag=gK0c5010b1..To<mailto:6@66.62.60.238:5060%3e;tag=gK0c5010b1..To>: <sip:To_User@kamailio:5060<sip:4704430714@192.168.9.130:5060>>;tag=as7651768b..Call-ID: 638356677_31693...@carrier..cseq<mailto:638356677_31693...@66.62.60.238..cseq>: 10795 ACK..Max-Forwards: 70..Route: <sip:kamailio:5060;lr=on<sip:192.168.9.130:5060;lr=on>>..Content-Length: 0.... U kamailio:5060 -> Asterisk:5060 ACK sip:1To_User@Asterisk:5060<sip:14704430714@10.1.6.247:5060> SIP/2.0..Via: SIP/2.0/UDP kamailio;branch=z9hG4bKcydzigwkX..Via: SIP/2.0/UDP 10.10.10.10;redact2=xyzA7CwnD23cQBsv6prYkU3Yk23Ykp3ckcGikA OYkpzfRN1fRwBYuphcWZAHW4IUMUzURVT.EYSP8SLJgf7UIaOU72GUDcMcuNecDVkcufDYu6E..From: <sip:From_User@Carrier:5060<sip:13039972016@66.62.60.238:5060>>;tag=gK0c5010b1..To: <sip:To_User@kamailio:5060<sip:4704430714@192.168.9.130:5060>>;tag =as7651768b..Call-ID: 638356677_31693...@carrier..cseq<mailto:638356677_31693...@66.62.60.238..cseq>: 10795 ACK..Max-Forwards: 69..Content-Length: 0.... U Carrier:5060 -> kamailio:5060 INVITE sip:10.10.10.10;redact1=abcfhCOikg7YDp7YWcOYDg7wWgOnkg3YeGKYWf9Yuphcp** SIP/2.0..Via: SIP/2.0/UDP Carrier:5060;branch=z9hG4bK0cBb96217452c17351f..From: <sip:From_User @Carrier:5060>;tag=gK0c5010b1..To<@Carrier:5060%3e;tag=gK0c5010b1..To>: <sip:To_User@kamailio:5060<sip:4704430714@192.168.9.130:5060>>;tag=as7651768b..Call-ID: 638356677_31693...@carrier..cseq<mailto:638356677_31693...@66.62.60.238..cseq>: 10796 INVITE..Max-Forwards: 70.. Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS..Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, mu ltipart/mixed..Contact: <sip:From_User@Carrier:5060<sip:13039972016@66.62.60.238:5060>>..Route: <sip:kamailio:5060;lr=on<sip:192.168.9.130:5060;lr=on>>..Supported: timer..Session-Expires: 1800;refresher=uac..Min-SE: 90..Content-L ength: 234..Content-Disposition: session; handling=required..Content-Type: application/sdp....v=0..o=Sonus_UAC 3540 9911 IN IP4 carrier..s=SIP Media Capabilities..c=IN IP 4 Carrier_MGP..t=0 0..m=audio 22406 RTP/AVP 0 100..a=rtpmap:0 PCMU/8000..a=rtpmap:100 telephone-event/8000..a=fmtp:100 0-15..a=sendrecv..a=maxptime:20.. Matt Scott
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