sorry, of course, here are the invites:
*Invite from asterisk --> opensips* U 62.66.66.67:5060 -> 62.66.66.66:5060 INVITE sip:[email protected] SIP/2.0. Via: SIP/2.0/UDP 62.66.66.67:5060;branch=z9hG4bK16ff74c2;rport. Max-Forwards: 70. From: "49302332434343" <sip:[email protected]>;tag=as1fcd8c32. To: <sip:[email protected]>. Contact: <sip:[email protected]>. Call-ID: [email protected]. CSeq: 102 INVITE. User-Agent: INES. Date: Thu, 04 Feb 2010 10:33:10 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 290. . v=0. o=root 992341641 992341641 IN IP4 62.66.66.67. s=Asterisk PBX 1.6.0-beta9. c=IN IP4 62.66.66.67. t=0 0. m=audio 32482 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. *Invite from opensips --> pstngw: * U 62.66.66.66:5060 -> 213.20.11.11:5060 INVITE sip:[email protected] SIP/2.0. Record-Route: <sip:62.66.66.66;lr=on;ftag=as1fcd8c32;did=a1.547e8c16>. Via: SIP/2.0/UDP 62.66.66.66;branch=z9hG4bKbee2.9124c5b3.0. Via: SIP/2.0/UDP 62.66.66.67:5060;received=62.66.66.67;branch=z9hG4bK16ff74c2;rport=5060. Max-Forwards: 69. From: "49302332434343" <sip:[email protected]>;tag=as1fcd8c32. To: <sip:[email protected]>. Contact: <sip:[email protected]>. Call-ID: [email protected]. CSeq: 102 INVITE. User-Agent: INES. Date: Thu, 04 Feb 2010 10:33:10 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 308. Session-Expires: 1800. . v=0. o=root 992341641 992341641 IN IP4 62.66.66.66. s=Asterisk PBX 1.6.0-beta9. c=IN IP4 62.66.66.66. t=0 0. m=audio 55408 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. a=nortpproxy:yes. Hope you can find something useful. Thanks. <mailto:sip%[email protected]>Brett Nemeroff schrieb: > >From my experience, this usually happen either from a configuration > file error, or from the terminating UAS who sends the BYE not sending > it to the RURI in the contact header from the original INVITE. Can we > see the original INVITE as it hits OpenSIPs? > -Brett > > > On Thu, Feb 4, 2010 at 4:56 AM, Max Mühlbronner <[email protected] > <mailto:[email protected]>> wrote: > > Hello everyone, > > i have a problem when a call is hangup by the callee, i think i > probably > have some general routing logic Problem and i cant find any way to > solve it. > > caller --> asterisk (62.66.66.67) --> opensips(62.66.66.66) (+rtpproxy > on the same machine) --> pstngw (213.20.11.11) > > Everything seems to be working fine, i have been testing a long time > but i recognized some problem. When the callee rejects the call , > (486 > busy) the busy is fine, transmitted back to the caller. > But if the call is established and the callee hangs up, the BYE is not > received by the original calling side so it stays connected. > My opensips knowledge is still very basic, so please excuse if it is > some dumb routing mistake made by me. > > 62.66.66.66 --> opensips > 62.66.66.67 --> asterisk > 213.20.11.11 --> pstngw > > > The busy is fine, and transmitted correctly (and callattempt is > stopped), but the bye is not received on the side where the call was > originating from (asterisk). > > > U 213.20.11.11:5060 <http://213.20.11.11:5060> -> 62.66.66.66:5060 > <http://62.66.66.66:5060> > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 62.66.66.66;branch=z9hG4bKf41a.b1b6523.0. > Via: SIP/2.0/UDP > 62.66.66.67:5060;received=62.66.66.67;branch=z9hG4bK79996cc9;rport=5060. > From: "49302332434343" <sip:[email protected] > <mailto:sip%[email protected]>>;tag=as67e89fcd. > To: <sip:[email protected] > <mailto:sip%[email protected]>>;tag=255533104. > Call-ID: [email protected] > <mailto:[email protected]>. > CSeq: 102 INVITE. > Contact: <sip:[email protected]:5060 > <http://sip:[email protected]:5060>>. > Content-Length: 0. > > > --------------------- > > > the not working BYE, followed by 404 not here, which is sent by the > basic routing block (like in most of the example scripts / > sl_send_reply("404","Not here");) > > > U 213.20.11.11:5060 <http://213.20.11.11:5060> -> 62.66.66.66:5060 > <http://62.66.66.66:5060> > BYE sip:[email protected] > <mailto:sip%[email protected]> SIP/2.0. > Via: SIP/2.0/UDP > 213.20.11.11:5060;branch=z9hG4bK623vjs00c0q1bggou101sd0000g00 > .1. > From: <sip:[email protected] > <mailto:sip%[email protected]>>;tag=960392687. > To: "49302332434343" <sip:[email protected] > <mailto:sip%[email protected]>>;tag=as32838038. > Call-ID: [email protected] > <mailto:[email protected]>. > CSeq: 1 BYE. > Max-Forwards: 9. > Supported: timer. > Content-Length: 0. > Route: <sip:62.66.66.66;lr=on;ftag=as32838038;did=fee.ee87c634>. > > > U 62.66.66.66:5060 <http://62.66.66.66:5060> -> 213.20.11.11:5060 > <http://213.20.11.11:5060> > SIP/2.0 404 Not here. > Via: SIP/2.0/UDP > > 213.20.11.11:5060;rport=5060;received=213.20.11.11;branch=z9hG4bKk4ksmf10eosgjf41m3k0sd0000g00.1. > From: <sip:[email protected] > <mailto:sip%[email protected]>>;tag=1126364538. > To: "49302332434343" <sip:[email protected] > <mailto:sip%[email protected]>>;tag=as1fcd8c32. > Call-ID: [email protected] > <mailto:[email protected]>. > CSeq: 1 BYE. > Server: OpenSIPS (1.6.1-notls (i386/linux)). > Content-Length: 0. > > > > > Thanks very much for any help, really appreciated. :) > > > Best Regards > > Max M. > > > _______________________________________________ > Users mailing list > [email protected] <mailto:[email protected]> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
