It's standard port 5060 UDP at both ends. I'll enable tracing on the external profile as well - but I'm quite sure it's not used at all in this case.
I'm on my way to the lab now, so I'll soon get back with my results after using latest SVN revision (my 'old' revision was just about 2 days old though). Peter ________________________________ Från: Brian West <br...@freeswitch.org> Skickat: den 15 april 2009 23:29 Till: freeswitch-users@lists.freeswitch.org <freeswitch-users@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. /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 br...@freeswitch.org<mailto:br...@freeswitch.org> -- Meet us at ClueCon! http://www.cluecon.com<http://www.cluecon.com/> !DSPAM:49e651b332933023977319! _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
