This sounds familiar: What revision of the code is this? Can you confirm you have this problem with SVN trunk (r13034 at the time of this writing).
On Wed, Apr 15, 2009 at 11:24 AM, Peter Olsson < [email protected]> wrote: > This is the full SIP-trace for the call. It’s not sending a BYE at all, > and I can’t see one in Wireshark either. As you can see in the end there is > a call to hangup_function(), but no SIP messages after that. When I manually > hangup the phone I can see it sends BYE to FreeSWITCH (which is quite > expected, since it thinks the call still exists), and FreeSWITCH just > answers ”481 Call Does Not Exist” – which of course is also expected, since > the call was dropped. > > > > recv 1255 bytes from udp/[192.168.94.53]:32769 at 16:17:57.853727: > > ------------------------------------------------------------------------ > > INVITE sip:[email protected]:5060;lr SIP/2.0 > > Accept-Language: en > > Call-ID: 80948a675733de14449f79df00 > > CSeq: 1 INVITE > > From: "Peter Olsson" <sip:[email protected]:6001 > >;tag=80948a675733de13449f79df00 > > Record-Route: <sip:192.168.94.53:5060;lr>,<sip:192.168.94.53:6001 > ;lr;transport=tls> > > To: "2100" <sip:[email protected] <sip%[email protected]>> > > Via: SIP/2.0/UDP > 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS > 192.168.94.53:6001 > ;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00 > > Content-Length: 165 > > Content-Type: application/sdp > > Contact: "Peter Olsson" <sip:[email protected]:6001;transport=tls> > > Max-Forwards: 67 > > User-Agent: Avaya CM/R015x.01.1.415.1 > > Allow: > INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH > > Supported: 100rel,timer,replaces,join,histinfo > > Alert-Info: <cid:[email protected] > >;avaya-cm-alert-type=internal > > Min-SE: 1200 > > Session-Expires: 1200;refresher=uac > > P-Asserted-Identity: "Peter Olsson" <sip:[email protected]:6001> > > History-Info: <sip:[email protected] > <sip%[email protected]>>;index=1,"2100" > <sip:[email protected] <sip%[email protected]>>;index=1.1 > > > > v=0 > > o=- 1 1 IN IP4 192.168.94.53 > > s=- > > c=IN IP4 192.168.94.59 > > b=AS:64 > > t=0 0 > > m=audio 2062 RTP/AVP 8 127 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:127 telephone-event/8000 > > ------------------------------------------------------------------------ > > send 541 bytes to udp/[192.168.94.53]:5060 at 16:17:57.854727: > > ------------------------------------------------------------------------ > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP > 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS > 192.168.94.53:6001 > ;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00 > > Record-Route: <sip:192.168.94.53:5060;lr> > > Record-Route: <sip:192.168.94.53:6001;lr;transport=tls> > > From: "Peter Olsson" <sip:[email protected]:6001 > >;tag=80948a675733de13449f79df00 > > To: "2100" <sip:[email protected] <sip%[email protected]>> > > Call-ID: 80948a675733de14449f79df00 > > CSeq: 1 INVITE > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > > Content-Length: 0 > > > > ------------------------------------------------------------------------ > > 2009-04-15 18:17:57 [NOTICE] switch_channel.c:597 switch_channel_set_name() > NewChannel > sofia/internal/[email protected]:6001[fa1c328e-bdfe-7d49-ab6f-dc9ec791c455] > > 2009-04-15 18:17:57 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() > Processing Peter Olsson->2100 in context public > > 2009-04-15 18:17:57 [NOTICE] mod_dptools.c:649 answer_function() Channel > [sofia/internal/[email protected]:6001] has been answered > > send 1322 bytes to udp/[192.168.94.53]:5060 at 16:17:57.871727: > > ------------------------------------------------------------------------ > > SIP/2.0 200 OK > > Via: SIP/2.0/UDP > 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.0,SIP/2.0/TLS > 192.168.94.53:6001 > ;psrrposn=2;received=192.168.94.53;branch=z9hG4bK80948a675733de15449f79df00 > > Record-Route: <sip:192.168.94.53:5060;lr> > > Record-Route: <sip:192.168.94.53:6001;lr;transport=tls> > > From: "Peter Olsson" <sip:[email protected]:6001 > >;tag=80948a675733de13449f79df00 > > To: "2100" <sip:[email protected] <sip%[email protected]> > >;tag=Sv6KrDv9vQrer > > Call-ID: 80948a675733de14449f79df00 > > CSeq: 1 INVITE > > Contact: <sip:[email protected]:5060;transport=udp> > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN > > Accept: application/sdp > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > Require: timer > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, presence, dialog, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > > Session-Expires: 1200;refresher=uac > > Min-SE: 1200 > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 265 > > > > v=0 > > o=FreeSWITCH 484797194364394181 220756314446402535 IN IP4 192.168.1.155 > > s=FreeSWITCH > > c=IN IP4 192.168.1.155 > > t=0 0 > > m=audio 23574 RTP/AVP 8 127 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:127 telephone-event/8000 > > a=fmtp:127 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > ------------------------------------------------------------------------ > > recv 521 bytes from udp/[192.168.94.53]:32769 at 16:17:57.880727: > > ------------------------------------------------------------------------ > > ACK sip:[email protected]:5060;transport=udp SIP/2.0 > > From: "Peter Olsson" <sip:[email protected]:6001 > >;tag=80948a675733de13449f79df00 > > To: "2100" <sip:[email protected] <sip%[email protected]> > >;tag=Sv6KrDv9vQrer > > Call-ID: 80948a675733de14449f79df00 > > CSeq: 1 ACK > > Max-Forwards: 69 > > Via: SIP/2.0/UDP > 192.168.94.53:5060;branch=z9hG4bK8383830303039393936541.1,SIP/2.0/TLS > 192.168.94.53:6001;psrrposn=1;branch=z9hG4bK80948a675733de16449f79df00 > > > > User-Agent: Avaya CM/R015x.01.1.415.1 > > Content-Length: 0 > > Record-Route: <sip:192.168.94.53:5060;lr> > > > > ------------------------------------------------------------------------ > > 2009-04-15 18:18:02 [NOTICE] mod_dptools.c:633 hangup_function() Hangup > sofia/internal/[email protected]:6001 [CS_EXECUTE] [NORMAL_CLEARING] > > 2009-04-15 18:18:02 [NOTICE] switch_core_session.c:1021 > switch_core_session_thread() Session 5 (sofia/internal/[email protected]:6001) > Ended > > 2009-04-15 18:18:02 [NOTICE] switch_core_session.c:1023 > switch_core_session_thread() Close Channel > sofia/internal/[email protected]:6001[CS_DESTROY] > > > > > > *Från:* [email protected] [mailto: > [email protected]] *För *Anthony Minessale > *Skickat:* den 15 april 2009 17:27 > *Till:* [email protected] > *Ämne:* Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH > ends the call? > > > > type: sofia profile internal siptrace on at the cli and try again > > see if you cen see FS sending BYE to the wrong address. > > This can be caused by a false positive on the NAT detection or when you > need NAT mode and you don't have it enabled. > > first edit the sofia profile in your config and comment out any line with > the word nat in them > > > On Wed, Apr 15, 2009 at 8:43 AM, Peter Olsson < > [email protected]> wrote: > > When I do a call from my Avaya SIP Server to FreeSWITCH. And then let > FreeSWITCH do a hangup of the call, FreeSWITCH doesn’t seem to send a ”BYE” > back to the Avaya PBX. I’ve narrowed it down to this simple example in the > dialplan; > > > > <action application="answer"/> > > <action application="sleep" data="5000"/> > > <action application="hangup"/> > > > > In this case no BYE is issued, and the phone still thinks the call is > alive. If you want to I could send the SIP headers as well for this > scenario.. > > > > Regards, > > > > Peter Olsson > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:[email protected] <msn%[email protected]> > GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[email protected] <sip%[email protected]> > iax:[email protected]/888 > googletalk:[email protected]<googletalk%3aconf%[email protected]> > pstn:213-799-1400 > !DSPAM:49e5fe5232932637379622! > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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