[Freeswitch-users] RE FS/Sofia not sending bye when FreeSWITCH nds the call?

Wed Apr 15 10:50:40 PDT 2009

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 at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Anthony Minessale [anthony.minessale at gmail.com]
Skickat: den 15 april 2009 18:46
Till: freeswitch-users at 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 at visionutveckling.se<mailto: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<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 at 192.168.94.53<mailto: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<http://sip:1002@sip.se:6001>>

   History-Info: <sip:2100 at 192.168.94.53<mailto:sip%3A2100 at 192.168.94.53>>;index=1,"2100" <sip:2100 at 192.168.94.53<mailto: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<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00

   To: "2100" <sip:2100 at 192.168.94.53<mailto: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<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 at 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 at sip.se:6001<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00

   To: "2100" <sip:2100 at 192.168.94.53<mailto: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<http://sip:1002@sip.se:6001>>;tag=80948a675733de13449f79df00

   To: "2100" <sip:2100 at 192.168.94.53<mailto: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<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 at 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 at sip.se:6001<http://1002@sip.se:6001> [CS_DESTROY]

Från: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto: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<mailto: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<mailto: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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