> On 26 May 2016, at 18:47, Jeremy Betts <jeremy.be...@freevoiceusa.com> wrote: > > Hello, > > I'm having an issue where kamailio is not sending the failure reply as I > would expect when receiving calls from a certain provider. Kamailio sends the > 403 response on calls from other providers. The big difference I see is that > the "problem" provider is using compact headers. I am trying to force the 403 > response to be sent for testing purposes but I just can't get it to reply to > this one provider. > > I've included traces and configuration excerpts below. Any help is much > appreciated! > > Problem Call (No Reply Sent): There must be something missing here. Without a response, there would be no ACK.
/O > > U 63.79.178.192:5060 -> 184.171.164.100:5060 > INVITE sip:+19727289377@184.171.164.100;transport=UDP;user=phone > <sip:+19727289377@184.171.164.100;transport=UDP;user=phone> SIP/2.0. > v: SIP/2.0/UDP > 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. > Record-Route: <sip:63.79.178.192;lr> <sip:63.79.178.192;lr>. > f: <sip:+17143258018@199.173.94.144:5060;user=phone> > <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. > t: <sip:+19727289377@63.79.178.192:5060;user=phone> > <sip:+19727289377@63.79.178.192:5060;user=phone>. > i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. > CSeq: 1 INVITE. > Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. > v: SIP/2.0/UDP > SCR9:5060;maddr=199.173.94.144;branch=z9hG4bK-41c7bce-f439f11-21b78f96;received=199.173.94.144. > Max-Forwards: 27. > m: <sip:199.173.94.144:5060;transport=UDP> > <sip:199.173.94.144:5060;transport=UDP>. > k: 100rel, resource-priority, replaces. > c: application/sdp. > l: 235. > P-Asserted-Identity: <sip:+17143258018@63.79.178.192;user=phone> > <sip:+17143258018@63.79.178.192;user=phone>. > Privacy: none. > . > v=0. > o=PVG 1464277233580 1464277233580 IN IP4 199.173.68.106. > s=-. > p=+1 6135555555. > c=IN IP4 199.173.68.106. > t=0 0. > m=audio 55380 RTP/AVP 18 0 8 101. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > a=ptime:20. > a=fmtp:18 annexb=no. > > # > U 63.79.178.192:5060 -> 184.171.164.100:5060 > ACK sip:+19727289377@184.171.164.100;transport=UDP;user=phone > <sip:+19727289377@184.171.164.100;transport=UDP;user=phone> SIP/2.0. > v: SIP/2.0/UDP > 63.79.178.192:5060;branch=z9hG4bK9a2b97bc805e6a3b73b43e3de4150da5.1d819013. > f: <sip:+17143258018@199.173.94.144:5060;user=phone> > <sip:+17143258018@199.173.94.144:5060;user=phone>;tag=-45026-41c7bce-729616d4-41c7bce. > t: <sip:+19727289377@63.79.178.192:5060;user=phone> > <sip:+19727289377@63.79.178.192:5060;user=phone>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.1aac. > i: b03a96e8905eadc713c441c7bcef439f1124b7ca791c2679c0-0086-5719. > CSeq: 1 ACK. > l: 0. > Max-Forwards: 27. > . > > Successful Call (Reply Sent): > > # > U 64.136.173.31:5060 -> 184.171.164.100:5066 > INVITE sip:7146466334@184.171.164.100:5066 > <sip:7146466334@184.171.164.100:5066> SIP/2.0. > Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. > Record-Route: > <sip:sansay3408006464rdb14345@64.136.173.31:5060;lr;transport=udp> > <sip:sansay3408006464rdb14345@64.136.173.31:5060;lr;transport=udp>. > To: <sip:7146466334@184.171.164.100> <sip:7146466334@184.171.164.100>. > From: <sip:17143258018@64.136.173.31> > <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. > Call-ID: 1089333243-0-3123809354@64.136.173.226 > <mailto:1089333243-0-3123809354@64.136.173.226>. > CSeq: 1 INVITE. > Contact: <sip:17143258018@64.136.173.31:5060> > <sip:17143258018@64.136.173.31:5060>. > Supported: timer. > Session-Expires: 1800;refresher=uac. > Min-SE: 90. > P-Asserted-Identity: <sip:17143258018@192.168.20.76> > <sip:17143258018@192.168.20.76>. > Privacy: none. > Expires: 120. > Max-Forwards: 67. > Content-Type: application/sdp. > Content-Length: 274. > . > v=0. > o=Sansay-VSXi 188 1 IN IP4 64.136.173.31. > s=Session Controller. > c=IN IP4 69.85.185.142. > t=0 0. > m=audio 37970 RTP/AVP 0 18 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:18 G729/8000. > a=fmtp:18 annexb=no. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > a=sendrecv. > a=ptime:20. > > > # > U 184.171.164.100:5066 -> 64.136.173.31:5060 > SIP/2.0 403 DID Lookup Failed. > Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. > To: <sip:7146466334@184.171.164.100> > <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. > From: <sip:17143258018@64.136.173.31> > <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. > Call-ID: 1089333243-0-3123809354@64.136.173.226 > <mailto:1089333243-0-3123809354@64.136.173.226>. > CSeq: 1 INVITE. > Server: x-Freevoice SIP Proxy 4.21. > Content-Length: 0. > . > > # > U 64.136.173.31:5060 -> 184.171.164.100:5066 > ACK sip:7146466334@184.171.164.100:5066 <sip:7146466334@184.171.164.100:5066> > SIP/2.0. > Via: SIP/2.0/UDP 64.136.173.31:5060;branch=z9hG4bK1sansay3408006464rdb14345. > To: <sip:7146466334@184.171.164.100> > <sip:7146466334@184.171.164.100>;tag=b1eb89aa72a4b2a406f6fb21bbd3e03f.d2ec. > From: <sip:17143258018@64.136.173.31> > <sip:17143258018@64.136.173.31>;tag=sansay3408006464rdb14345. > Call-ID: 1089333243-0-3123809354@64.136.173.226 > <mailto:1089333243-0-3123809354@64.136.173.226>. > CSeq: 1 ACK. > Max-Forwards: 70. > Content-Length: 0. > > Kamailio Config: > request_route { > > # per request initial checks > route(REQINIT); > > # NAT detection > #route(NATDETECT); > > # handle requests within SIP dialogs > route(WITHINDLG); > > ### only initial requests (no To tag) > > # CANCEL processing > if (is_method("CANCEL")) > { > setflag(FLT_ACCFAILED); > if (t_check_trans()) > t_relay(); > exit; > } > > t_check_trans(); > > if(is_method("INVITE")||is_method("MESSAGE")){ > route(ORIGINATE); > #route(FROMAST); > } > > > route[ORIGINATE] { > > sl_send_reply("403", "DID Lookup Failed"); > exit; > } > > > Jeremy Betts > > <https://www.youtube.com/user/FreevoiceLLC> > > <https://www.youtube.com/user/FreevoiceLLC>_______________________________________________ > 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
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