[Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?
Anthony Minessale
anthony.minessale at gmail.com
Wed Apr 15 09:46:50 PDT 2009
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 at 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 at 192.168.1.155:5060;lr SIP/2.0
>
> Accept-Language: en
>
> Call-ID: 80948a675733de14449f79df00
>
> CSeq: 1 INVITE
>
> From: "Peter Olsson" <sip:1002 at 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 at 192.168.94.53 <sip%3A2100 at 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 at 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 at invalid.unknown.domain
> >;avaya-cm-alert-type=internal
>
> Min-SE: 1200
>
> Session-Expires: 1200;refresher=uac
>
> P-Asserted-Identity: "Peter Olsson" <sip:1002 at sip.se:6001>
>
> History-Info: <sip:2100 at 192.168.94.53 <sip%3A2100 at 192.168.94.53>>;index=1,"2100"
> <sip:2100 at 192.168.94.53 <sip%3A2100 at 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 at sip.se:6001
> >;tag=80948a675733de13449f79df00
>
> To: "2100" <sip:2100 at 192.168.94.53 <sip%3A2100 at 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 at 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 at 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 at sip.se:6001
> >;tag=80948a675733de13449f79df00
>
> To: "2100" <sip:2100 at 192.168.94.53 <sip%3A2100 at 192.168.94.53>
> >;tag=Sv6KrDv9vQrer
>
> Call-ID: 80948a675733de14449f79df00
>
> CSeq: 1 INVITE
>
> Contact: <sip:mod_sofia at 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 at 192.168.1.155:5060;transport=udp SIP/2.0
>
> From: "Peter Olsson" <sip:1002 at sip.se:6001
> >;tag=80948a675733de13449f79df00
>
> To: "2100" <sip:2100 at 192.168.94.53 <sip%3A2100 at 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 at 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 at 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 at sip.se:6001[CS_DESTROY]
>
>
>
>
>
> *Från:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *För *Anthony Minessale
> *Skickat:* den 15 april 2009 17:27
> *Till:* freeswitch-users at 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 at 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
>
>
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>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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>
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> iax:guest at conference.freeswitch.org/888
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> !DSPAM:49e5fe5232932637379622!
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>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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