RE FS/Sofia not sending bye when FreeSWITCH nds the call?
My current revision is r13015. I will do an update as soon as possible and see if that solves the issue. Thanks! //Peter ________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Anthony Minessale [anthony.minessale@gmail.com] Skickat: den 15 april 2009 18:46 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call? 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 <peter.olsson@visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> 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:2100@192.168.1.155:5060;lr SIP/2.0 Accept-Language: en Call-ID: 80948a675733de14449f79df00 CSeq: 1 INVITE From: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00 Record-Route: <sip:192.168.94.53:5060;lr>,<sip:192.168.94.53:6001;lr;transport=tls> To: "2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>> 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:1002@192.168.94.53:6001;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:internal@invalid.unknown.domain>;avaya-cm-alert-type=internal Min-SE: 1200 Session-Expires: 1200;refresher=uac P-Asserted-Identity: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>> History-Info: <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>>;index=1,"2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>>;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:1002@sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00 To: "2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>> 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/1002@sip.se:6001<http://1002@sip.se: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/1002@sip.se:6001<http://1002@sip.se: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:1002@sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00 To: "2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>>;tag=Sv6KrDv9vQrer Call-ID: 80948a675733de14449f79df00 CSeq: 1 INVITE Contact: <sip:mod_sofia@192.168.1.155:5060;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:mod_sofia@192.168.1.155:5060;transport=udp SIP/2.0 From: "Peter Olsson" <sip:1002@sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00 To: "2100" <sip:2100@192.168.94.53<mailto:sip%3A2100@192.168.94.53>>;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/1002@sip.se:6001<http://1002@sip.se: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/1002@sip.se:6001<http://1002@sip.se:6001>) Ended 2009-04-15 18:18:02 [NOTICE] switch_core_session.c:1023 switch_core_session_thread() Close Channel sofia/internal/1002@sip.se:6001<http://1002@sip.se:6001> [CS_DESTROY] Från: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Anthony Minessale Skickat: den 15 april 2009 17:27 Till: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> 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 <peter.olsson@visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> 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 Freeswitch-users@lists.freeswitch.org<mailto:Freeswitch-users@lists.freeswitch.org> 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:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org<mailto:Freeswitch-users@lists.freeswitch.org> 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:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 !DSPAM:49e611a532932126013823!
I recall this and transport=tls in this case... maybe MikeJ can chime in on this one..I thought we already fixed this. On Apr 15, 2009, at 12:50 PM, Peter Olsson wrote:
Record-Route: <sip:192.168.94.53:6001;lr;transport=tls>
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
I've built using latest trunk now, but I won't be able to test again until tomorrow - I'll get back to you after that. Just to make the scenario a bit more clear; The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be something that causes the problem? I also tried to dial into the dialplan, answer the call, and then try to deflect the call using REFER. This didn't create any SIP messages either (and nothing happened with the call), so it seems there might be a bigger issue than just BYE. Peter On 09-04-15 20.04, "Brian West" <brian@freeswitch.org> wrote: I recall this and transport=tls in this case... maybe MikeJ can chime in on this one..I thought we already fixed this. On Apr 15, 2009, at 12:50 PM, Peter Olsson wrote: Record-Route: <sip:192.168.94.53:6001;lr;transport=tls> Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com <http://www.cluecon.com/><http://www.cluecon.com/> !DSPAM:49e623a532931080721630!
What port are you hitting? Make sure you turn sip tracing on external and internal just in case you're using either or both. /b On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote:
I've built using latest trunk now, but I won't be able to test again until tomorrow - I'll get back to you after that.
Just to make the scenario a bit more clear; The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be something that causes the problem? I also tried to dial into the dialplan, answer the call, and then try to deflect the call using REFER. This didn't create any SIP messages either (and nothing happened with the call), so it seems there might be a bigger issue than just BYE.
Peter
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
Allright, I tried this again now, with revision 13042 - it's the same result as before.. Should I file a jira case for this? If you want any more information, or more traces, please get back to me, and I'll try to help out as much as possible. Peter Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Brian West Skickat: den 15 april 2009 23:21 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call? What port are you hitting? Make sure you turn sip tracing on external and internal just in case you're using either or both. /b On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote: I've built using latest trunk now, but I won't be able to test again until tomorrow - I'll get back to you after that. Just to make the scenario a bit more clear; The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be something that causes the problem? I also tried to dial into the dialplan, answer the call, and then try to deflect the call using REFER. This didn't create any SIP messages either (and nothing happened with the call), so it seems there might be a bigger issue than just BYE. Peter Brian West brian@freeswitch.org<mailto:brian@freeswitch.org> -- Meet us at ClueCon! http://www.cluecon.com<http://www.cluecon.com/> !DSPAM:49e651b332933023977319!
yes open a jira http://jira.freeswitch.org *attach* the following (do not paste it inline into the comments and give all trace files a .txt extension) repeat the trace you did earlier with more debugging enabled. type these 3 cli commands before you call sofia profile internal siptrace on sofia loglevel all 9 console loglevel debug On Thu, Apr 16, 2009 at 2:13 AM, Peter Olsson < peter.olsson@visionutveckling.se> wrote:
Allright, I tried this again now, with revision 13042 – it’s the same result as before.. Should I file a jira case for this?
If you want any more information, or more traces, please get back to me, and I’ll try to help out as much as possible.
Peter
*Från:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *För *Brian West *Skickat:* den 15 april 2009 23:21 *Till:* freeswitch-users@lists.freeswitch.org *Ämne:* Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call?
What port are you hitting? Make sure you turn sip tracing on external and internal just in case you're using either or both.
/b
On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote:
I've built using latest trunk now, but I won't be able to test again until tomorrow - I'll get back to you after that.
Just to make the scenario a bit more clear; The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be something that causes the problem? I also tried to dial into the dialplan, answer the call, and then try to deflect the call using REFER. This didn't create any SIP messages either (and nothing happened with the call), so it seems there might be a bigger issue than just BYE.
Peter
Brian West
brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
!DSPAM:49e651b332933023977319!
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org 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:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I've added this as jira case http://jira.freeswitch.org/browse/MODSOFIA-4 I wasn't sure if it should be under mod_sofia or sofia-sip. The report has a full debug log attached. Regards, Peter Olsson Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony Minessale Skickat: den 16 april 2009 14:23 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call? yes open a jira http://jira.freeswitch.org *attach* the following (do not paste it inline into the comments and give all trace files a .txt extension) repeat the trace you did earlier with more debugging enabled. type these 3 cli commands before you call sofia profile internal siptrace on sofia loglevel all 9 console loglevel debug On Thu, Apr 16, 2009 at 2:13 AM, Peter Olsson <peter.olsson@visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: Allright, I tried this again now, with revision 13042 - it's the same result as before.. Should I file a jira case for this? If you want any more information, or more traces, please get back to me, and I'll try to help out as much as possible. Peter Från: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Brian West Skickat: den 15 april 2009 23:21 Till: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: Re: [Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call? What port are you hitting? Make sure you turn sip tracing on external and internal just in case you're using either or both. /b On Apr 15, 2009, at 4:12 PM, Peter Olsson wrote: I've built using latest trunk now, but I won't be able to test again until tomorrow - I'll get back to you after that. Just to make the scenario a bit more clear; The Avaya CM has an internal SIP-trunk over tls, to an Avaya SES Server (SIP Enablement Services), this one talks UDP to FreeSWITCH. Could this be something that causes the problem? I also tried to dial into the dialplan, answer the call, and then try to deflect the call using REFER. This didn't create any SIP messages either (and nothing happened with the call), so it seems there might be a bigger issue than just BYE. Peter Brian West brian@freeswitch.org<mailto:brian@freeswitch.org> -- Meet us at ClueCon! http://www.cluecon.com<http://www.cluecon.com/> _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org<mailto:Freeswitch-users@lists.freeswitch.org> 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:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888<http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 !DSPAM:49e725b432939831339029!
participants (3)
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Anthony Minessale
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Brian West
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Peter Olsson