[SR-Users] Session Timers Problem

[Scott, Matt]

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 at carrier:50
  60>;tag=gK0c5010b1..To: <sip:To_User at kamailio:5060<sip:4704430714 at 192.168.9.130:5060>>;tag=as7651768b..Call-ID: 638356677_31693090 at carrier..CSeq<mailto:638356677_31693090 at 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 at Carrier:5060>;tag=gK0c5010b1..To<mailto:6 at 66.62.60.238:5060%3e;tag=gK0c5010b1..To>: <sip:To_User at kamailio:5060<sip:4704430714 at 192.168.9.130:5060>>;tag=as7651768b..Call-ID: 638356677_31693090 at carrier..CSeq<mailto:638356677_31693090 at 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 at Asterisk:5060<sip:14704430714 at 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 at Carrier:5060<sip:13039972016 at 66.62.60.238:5060>>;tag=gK0c5010b1..To: <sip:To_User at kamailio:5060<sip:4704430714 at 192.168.9.130:5060>>;tag
  =as7651768b..Call-ID: 638356677_31693090 at carrier..CSeq<mailto:638356677_31693090 at 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 at kamailio:5060<sip:4704430714 at 192.168.9.130:5060>>;tag=as7651768b..Call-ID: 638356677_31693090 at carrier..CSeq<mailto:638356677_31693090 at 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 at Carrier:5060<sip:13039972016 at 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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