[Freeswitch-users] FS/Sofia not sending bye when FreeSWITCH ends the call?
peter.olsson at visionutveckling.se
Thu Apr 16 07:47:18 PDT 2009
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.
Från: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] För Anthony Minessale
Skickat: den 16 april 2009 14:23
Till: freeswitch-users at lists.freeswitch.org
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 <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> 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.
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 Brian West
Skickat: den 15 april 2009 23:21
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
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.
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.
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