Hi James, It might not be correct - by setting the expires to 600, you will not allow calls longer than 10 minutes! I think you should rather try to fix the problem with the CANCEL !
BTW, please get a full openips log (use debug=6) for the entire call and send it to me - just want to run some checks on what is going on there. Regards, Bogdan James Wiegand wrote: > Hmm, > > How about 600 seconds? The only problem is that I can't use OpenSIPS > if the sessions get hung - especially when I am getting 20 calls per > second. > > -jim > > On Fri, Jun 19, 2009 at 5:33 PM, Bogdan-Andrei > Iancu<[email protected]> wrote: > >> Hi James, >> >> So, continuing the previous email....what you do by playing with the dialog >> expire param is forcing (from proxy) side to terminate the ongoing calls >> after 30 secs. As said, your call was not CANCELed, but was established - >> and you force the termination of the call after 30 secs, that is why it >> works now :D....but it is not correct. >> >> Regards, >> Bogdan >> >> James Wiegand wrote: >> >>> Don't know if this is the right thing to try, but when I set the >>> dialog timeout the session clears after a few moments. Is 30 seconds >>> too short for use on general calling patterns? I am looking to pass >>> on the order of 700 simultaneous calls. >>> >>> ... >>> modparam("dialog", "default_timeout", 30) >>> ... >>> >>> -jim >>> >>> On Fri, Jun 19, 2009 at 9:28 AM, James >>> Wiegand<[email protected]> 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 >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>> -- >>>> -- >>>> Jim Wiegand >>>> ----------- >>>> Home: [email protected] >>>> AIM: originaljimdandy >>>> >>>> >>>> >>> >>> >>> >> > > > > _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
