[Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?
Peter Olsson
peter.olsson at visionutveckling.se
Wed Apr 15 09:24:31 PDT 2009
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>
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>;index=1,"2100" <sip:2100 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>
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>;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>;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<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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Anthony Minessale II
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