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>;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>

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>

History-Info: <sip:2100@192.168.94.53>;index=1,"2100" <sip:2100@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>;tag=80948a675733de13449f79df00

To: "2100" <sip:2100@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 [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] 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>;tag=80948a675733de13449f79df00

To: "2100" <sip:2100@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>;tag=80948a675733de13449f79df00

To: "2100" <sip:2100@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 [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) 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 [CS_DESTROY]

Från: 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
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> 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;

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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