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: [email protected] 
[mailto:[email protected]] För Anthony Minessale
Skickat: den 16 april 2009 14:23
Till: [email protected]
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 
<[email protected]<mailto:[email protected]>> 
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: 
[email protected]<mailto:[email protected]>
 
[mailto:[email protected]<mailto:[email protected]>]
 För Brian West
Skickat: den 15 april 2009 23:21

Till: 
[email protected]<mailto:[email protected]>
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

[email protected]<mailto:[email protected]>



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