Hi James, I see your call is not actually cancelled - see the negative reply on the CANCEL and the 200 OK on INVITE - so the call does establish and this is the reason for showing up in the dialog and lb module.
Regards, Bogdan James Wiegand wrote: > Hi Bogdan, > > Here's the dialog from a test call. > The remote client is Eyebeam on a PC connected to Asterisk. I made a > call and hung up before answering. The call has been terminated for > some time. I can do an lb_reload to clear out the hung lb session. > > opensipsctl fifo lb_list > Destination:: sip:XXX.XXX.XXX.6 id=1 > Resource:: pstn max=0 load=0 > Destination:: sip:XXX.XXX.XXX.7 id=2 > Resource:: pstn max=0 load=0 > Destination:: sip:XXX.XXX.XXX.8 id=3 > Resource:: pstn max=1 load=1 > Destination:: sip:XXX.XXX.XXX.9 id=4 > Resource:: pstn max=0 load=0 > > opensipsctl fifo dlg_list > dialog:: hash=3498:265315739 > state:: 3 > user_flags:: 0 > timestart:: 1245419911 > timeout:: 99843 > callid:: [email protected] > from_uri:: sip:[email protected] > from_tag:: as14720305 > caller_contact:: sip:[email protected] > caller_cseq:: 102 > caller_route_set:: > caller_bind_addr:: udp:XXX.XXX.XXX.24:5060 > to_uri:: sip:[email protected] > to_tag:: as4042950a > callee_contact:: sip:[email protected] > callee_cseq:: 102 > callee_route_set:: > callee_bind_addr:: udp:XXX.XXX.XXX.24:5060 > dialog:: hash=3895:1205860066 > state:: 3 > user_flags:: 0 > timestart:: 1245419947 > timeout:: 99879 > callid:: [email protected] > from_uri:: sip:[email protected] > from_tag:: as5a726731 > caller_contact:: sip:[email protected] > caller_cseq:: 102 > caller_route_set:: > caller_bind_addr:: udp:XXX.XXX.XXX.24:5060 > to_uri:: sip:[email protected] > to_tag:: as3ac79c83 > callee_contact:: sip:[email protected] > callee_cseq:: 102 > callee_route_set:: > callee_bind_addr:: udp:XXX.XXX.XXX.24:5060 > > > TCP SIP trace, not from the same call, but with the same result: > > 09:08:37.758213 IP (tos 0x0, ttl 45, id 34347, offset 0, flags > [none], proto: UDP (17), length: 855) YYY.YYY.YYY.12.sip > > XXX.XXX.XXX.24.sip: SIP, length: 827 > INVITE sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP YYY.YYY.YYY.12:5060;branch=z9hG4bK1fe6bb8d;rport > From: "device" <sip:[email protected]>;tag=as0fb4ac11 > To: <sip:[email protected]> > Contact: <sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Fri, 19 Jun 2009 14:00:50 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 284 > > v=0 > o=root 3848 3848 IN IP4 YYY.YYY.YYY.12 > s=session > c=IN IP4 YYY.YYY.YYY.12 > t=0 0 > m=audio 6962 RTP/AVP 0 3 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 09:08:37.759853 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], > proto: UDP (17), length: 345) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip: > SIP, length: 317 > SIP/2.0 100 Giving a try > Via: SIP/2.0/UDP YYY.YYY.YYY.12:5060;branch=z9hG4bK1fe6bb8d;rport=5060 > From: "device" <sip:[email protected]>;tag=as0fb4ac11 > To: <sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 INVITE > Server: VistaVox SIP Service > Content-Length: 0 > > > 09:08:40.113592 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], > proto: UDP (17), length: 874) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip: > SIP, length: 846 > SIP/2.0 183 Session Progress > Via: SIP/2.0/UDP > YYY.YYY.YYY.12:5060;received=YYY.YYY.YYY.12;branch=z9hG4bK1fe6bb8d;rport=5060 > Record-Route: <sip:XXX.XXX.XXX.24;lr;ftag=as0fb4ac11;did=f0d.85cca1c> > From: "device" <sip:[email protected]>;tag=as0fb4ac11 > To: <sip:[email protected]>;tag=as2661bdde > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[email protected]> > Content-Type: application/sdp > Content-Length: 262 > > v=0 > o=root 18239 18239 IN IP4 XXX.XXX.XXX.8 > s=session > c=IN IP4 XXX.XXX.XXX.8 > t=0 0 > m=audio 16734 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 09:08:55.476673 IP (tos 0x0, ttl 45, id 34348, offset 0, flags > [none], proto: UDP (17), length: 372) YYY.YYY.YYY.12.sip > > XXX.XXX.XXX.24.sip: SIP, length: 344 > CANCEL sip:[email protected] SIP/2.0 > Via: SIP/2.0/UDP YYY.YYY.YYY.12:5060;branch=z9hG4bK1fe6bb8d;rport > From: "device" <sip:[email protected]>;tag=as0fb4ac11 > To: <sip:[email protected]> > Call-ID: [email protected] > CSeq: 102 CANCEL > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > > 09:08:55.477405 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], > proto: UDP (17), length: 393) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip: > SIP, length: 365 > SIP/2.0 405 Method Not Allowed > Via: SIP/2.0/UDP YYY.YYY.YYY.12:5060;branch=z9hG4bK1fe6bb8d;rport=5060 > From: "device" <sip:[email protected]>;tag=as0fb4ac11 > To: > <sip:[email protected]>;tag=9508a3e09327a949e746abbd3d262852.51a3 > Call-ID: [email protected] > CSeq: 102 CANCEL > Server: VistaVox SIP Service > Content-Length: 0 > > > 09:09:04.124970 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], > proto: UDP (17), length: 860) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip: > SIP, length: 832 > SIP/2.0 200 OK > Via: SIP/2.0/UDP > YYY.YYY.YYY.12:5060;received=YYY.YYY.YYY.12;branch=z9hG4bK1fe6bb8d;rport=5060 > Record-Route: <sip:XXX.XXX.XXX.24;lr;ftag=as0fb4ac11;did=f0d.85cca1c> > From: "device" <sip:[email protected]>;tag=as0fb4ac11 > To: <sip:[email protected]>;tag=as2661bdde > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[email protected]> > Content-Type: application/sdp > Content-Length: 262 > > v=0 > o=root 18239 18240 IN IP4 XXX.XXX.XXX.8 > s=session > c=IN IP4 XXX.XXX.XXX.8 > t=0 0 > m=audio 16734 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 09:09:05.123714 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], > proto: UDP (17), length: 860) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip: > SIP, length: 832 > SIP/2.0 200 OK > Via: SIP/2.0/UDP > YYY.YYY.YYY.12:5060;received=YYY.YYY.YYY.12;branch=z9hG4bK1fe6bb8d;rport=5060 > Record-Route: <sip:XXX.XXX.XXX.24;lr;ftag=as0fb4ac11;did=f0d.85cca1c> > From: "device" <sip:[email protected]>;tag=as0fb4ac11 > To: <sip:[email protected]>;tag=as2661bdde > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[email protected]> > Content-Type: application/sdp > Content-Length: 262 > > v=0 > o=root 18239 18240 IN IP4 XXX.XXX.XXX.8 > s=session > c=IN IP4 XXX.XXX.XXX.8 > t=0 0 > m=audio 16734 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 09:09:06.123020 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], > proto: UDP (17), length: 860) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip: > SIP, length: 832 > SIP/2.0 200 OK > Via: SIP/2.0/UDP > YYY.YYY.YYY.12:5060;received=YYY.YYY.YYY.12;branch=z9hG4bK1fe6bb8d;rport=5060 > Record-Route: <sip:XXX.XXX.XXX.24;lr;ftag=as0fb4ac11;did=f0d.85cca1c> > From: "device" <sip:[email protected]>;tag=as0fb4ac11 > To: <sip:[email protected]>;tag=as2661bdde > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[email protected]> > Content-Type: application/sdp > Content-Length: 262 > > v=0 > o=root 18239 18240 IN IP4 XXX.XXX.XXX.8 > s=session > c=IN IP4 XXX.XXX.XXX.8 > t=0 0 > m=audio 16734 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 09:09:08.123528 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], > proto: UDP (17), length: 860) XXX.XXX.XXX.24.sip > YYY.YYY.YYY.12.sip: > SIP, length: 832 > SIP/2.0 200 OK > Via: SIP/2.0/UDP > YYY.YYY.YYY.12:5060;received=YYY.YYY.YYY.12;branch=z9hG4bK1fe6bb8d;rport=5060 > Record-Route: <sip:XXX.XXX.XXX.24;lr;ftag=as0fb4ac11;did=f0d.85cca1c> > From: "device" <sip:[email protected]>;tag=as0fb4ac11 > To: <sip:[email protected]>;tag=as2661bdde > Call-ID: [email protected] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[email protected]> > Content-Type: application/sdp > Content-Length: 262 > > v=0 > o=root 18239 18240 IN IP4 XXX.XXX.XXX.8 > s=session > c=IN IP4 XXX.XXX.XXX.8 > t=0 0 > m=audio 16734 RTP/AVP 0 8 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > > On Fri, Jun 19, 2009 at 2:18 AM, Bogdan-Andrei > Iancu<[email protected]> wrote: > >> Hi James, >> >> Could you please check if the "dialog" module sees the call as ended? Use >> "opensipsctl fifo dlg_list" >> (http://www.opensips.org/html/docs/modules/1.5.x/dialog.html#id272726) and >> paste the output here. >> >> Also, do you have a full SIP trace of the call (ngrep) ? >> >> Regards, >> Bogdan >> >> >> >> James Wiegand wrote: >> >>> Hi all, >>> >>> I am using OpenSIPS 1.5.1 and the lb module. Following the example I >>> see this chunk of code execute when the caller hangs up as the dial >>> progresses (but before the other side answers): >>> >>> # from now on we have only the initial requests >>> if (!is_method("INVITE")) { >>> send_reply("405","Method Not Allowed"); >>> exit; >>> } >>> >>> This leaves a session hanging in the load balancer: >>> >>> Destination:: sip:XXX.XXX.XXX.XXX id=3 >>> Resource:: pstn max=1 load=1 >>> >>> I'm seeing CANCEL come in from the caller and it looks like >>> !t_check_trans() is not picking this up? How do I catch this case? >>> >>> Thanks for the help, >>> >>> -jim >>> >>> >>> >>> >> > > > > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
