Re: [SR-Users] Not seeing ACK to 200 as existing transaction

Hello all, If there is someone that can help with this, that would be rather good! I can supply further details upon anyone having seen this before....... In NORMAL operation, I get an INVITE and process this (My Kamailio is processing SS7 CAMEL, and does an MRSN lookup) It then makes the INVITE onwards to the MSRN collected. When answered, I see the 200. I pass the 200 back to the source of the INVITE I get the ACK, and use "t_check_trans()" and if that returns true will route the ACK onward (I am in the ACK path having set Record-route on the 200) This all works. FAILURE occurs when I get INVITE from another source (one step removed from the generating gateway) All works as it should *BUT* the t_check_trans() returns FALSE, so I guess there is a matching issue to the live transaction? I have tried a "work around" and routed the ACK anyway. This *does* work, and is routed correctly forward to the original "contact" and the 200 messages stop being repeated.. *BUT* the Kamailio instance here times out the dialog as the ACK was not seen correctly. I also popped a "t_check_trans()" into the reply route to test the 200 OK that I get, and that returns TRUE. Here is the 200 message that I send back to the INVITE sender: SIP/2.0 200 OK Via: SIP/2.0/UDP 52.1.2.3;rport=5060;branch=z9hG4bK7bb7.62ce43198ddade3212c715259623a75f.1 Via: SIP/2.0/UDP 149.1.2.3:5060;received=149.1.2.3;rport=5060;branch=z9hG4bK905c7e6e Record-Route: <sip:3.1.2.3;lr;ftag=81dd;did=08b.0141;vsf=AAAAAB8ABQMKCAgMAwUCCHgzWEQXXl5WSUNYQ14cUF5t> Record-Route: <sip:52.1.2.3:5060;lr> From: "+44128084XXXX" <sip:+44128084XXXX@149.1.2.3>;tag=81dd To: "+44739790XXXX" <sip:+44739790XXXX@52.1.2.3>;tag=XXgZtQND1H8rD Call-ID: 9bbd818c-d1c8-4c62-9fed-7878ed05cf721 CSeq: 1 INVITE Contact: <sip:+44739790XXXX@185.1.2.3:5060;transport=udp> User-Agent: Ziron Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 216 P-Asserted-Identity: "Outbound Call" <sip:44780204XXXX@52.1.2.3> v=0 o=MSX53 11014726 11014726 IN IP4 88.1.2.3 s=sip call c=IN IP4 88.1.2.3 t=0 0 a=sendrecv m=audio 14810 RTP/AVP 8 99 a=rtpmap:8 PCMA/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-15 a=ptime:20 And this is the ACK I get back: ACK sip:+44739790XXXX@185.1.2.3:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 52.1.2.3;branch=z9hG4bK7bb7.cca0ec5a2c96b8fe82344d9835163595.0 Via: SIP/2.0/UDP 149.1.2.3:5060;received=149.1.2.3;rport=5060;branch=z9hG4bK6b8f7781 Contact: sip:+44128084XXXX@149.1.2.3 To: "+44739790XXXX" <sip:+44739790XXXX@52.1.2.3>;tag=XXgZtQND1H8rD From: "+44128084XXXX" <sip:+44128084XXXX@149.1.2.3>;tag=81dd Call-ID: 9bbd818c-d1c8-4c62-9fed-7878ed05cf721 CSeq: 1 ACK User-Agent: PartitionwareTM SIP Toolkit v4.0.30319 Content-Length: 0 Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, INFO, OPTIONS, PRACK Supported: timer,100rel Allow-Events: telephone-event P-hint: outbound Any thoughts would be great, Steve _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users