Hello, I used kamailio to handle SIP line from a provider using uacreg table. Everything was working fine till they add a proxy which do not set (or not propagate) route header in ACK or BYE request. So my routing logic is broken in the "in-dialog" transaction mode, since the loose_route return FALSE. If I try to force a t_relay after the loose_route, my message may not be routing properly if for example my UAC was not connected with the standard 5060 port or use a different protocol like TCP (It works if both are in UDP and use 5060 port). Is there a way to retrieve the record-route previously sets for this session in order to route this message properly ?

Another problem is that the dialog module do not match the BYE transaction, the "did" variable is missing since the route is not there but it should match the request with SIP matching (dlg_match_mode is set to 1). For example my dialog parameters are: dialog:: hash=1214:1340198900 / state:: 3 timestart:: 1333006717 timeout:: 80537132 callid:: 16260-CI-30dd2867-797784ae0(a)my.sip.provider from_uri:: sip:0367023024@my.sip.provider;user=phone from_tag:: 16260-GI-30dd2868-17bb342c3 caller_contact:: sip:13.12.14.20:5060 caller_cseq:: 815264875 caller_route_set:: <sip:13.12.14.20:5060;lr> caller_bind_addr:: udp:130.120.140.131:5060 to_uri:: sip:0974711672@13.12.14.17;user=phone to_tag:: 1878467993 callee_contact:: sip:0974711672@96.57.249.78:1024 callee_cseq:: 815264875 callee_route_set:: callee_bind_addr:: udp:130.120.140.131:5060/ The BYE request: /BYE sip:0974711672@96.57.249.78:1024 SIP/2.0 Call-ID: 16260-CI-30dd2867-797784ae0(a)my.sip.provider CSeq: 815264876 BYE From: "0033482531303" <sip:0367023024@my.sip.provider;user=phone>;tag=16260-GI-30dd2868-17bb342c3 Max-Forwards: 28 Record-Route: <sip:13.12.14.20:5060;lr> To: <sip:0974711672@13.12.14.17;user=phone>;tag=1878467993 Via: SIP/2.0/UDP 13.12.14.20:5060;branch=z9hG4bK-LNVP-1196a924-5f1f495a Reason: q.850;cause=16 User-Agent: Cirpack/v4.42a (gw_sip) Content-Length: 0/ Why does this BYE request is not matched by the dialog modules using SIP parameters (Call-ID, uri, tag seems correct) ?

Hervé _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users