On 2009-10-14 11:15 +0200, Tihomir Culjaga wrote freeswitch-users@lists.fre...: TC>On Wed, Oct 14, 2009 at 10:16 AM, Tihomir Culjaga <tculjaga@gmail.com>wrote: TC> TC>> TC>> TC>> 2009/10/14 Georgiewskiy Yuriy <bottleman@icf.org.ru> TC>> TC>>> On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote TC>>> freeswitch-users@lists.fre...: TC>>> TC>>> i need trace level 4 from mod_h323 and debug log of entire call, tcpdump TC>>> may be needed later, i have no way TC>>> to test it on this time, i do it later. TC>>> TC>>> TC> TC>Here are the logs: TC> TC>call flow is H323_endpoint => FS => SIP_user TC> TC>slow start => http://pastebin.freeswitch.org/10693 TC>slow start w eH245 => http://pastebin.freeswitch.org/10694 TC>fastStart w tunneling true => http://pastebin.freeswitch.org/10701 TC>fastStart w tunneling false => http://pastebin.freeswitch.org/10702 TC>fastStart w tunneling true eH245 => http://pastebin.freeswitch.org/10699 TC>fastStart w tunneling false eH245 => http://pastebin.freeswitch.org/10700 TC> TC> TC>In all cases, there is no audio on SIP User side... i see the SIP Phone TC>(terminator) sending RTP to FS but FS is not forwarding it back to the TC>originator. Also, i see originator (H323) sending RTP to FS but FS doesn't TC>forward it to the terminator (SIP). TC> TC> TC> TC>call flow is SIP_user => FS => H323_endpoint is failing .. coredumped.... TC>http://pastebin.freeswitch.org/10703 i fix some bugs now, ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/mod_h323.tar.bz2 this is updated version, try it, if you experience no audio try enable rtp proxy in you sip profile. C уважением With Best Regards Георгиевский Юрий. Georgiewskiy Yuriy +7 4872 711666 +7 4872 711666 факс +7 4872 711143 fax +7 4872 711143 Компания ООО "Ай Ти Сервис" IT Service Ltd http://nkoort.ru http://nkoort.ru JID: GHhost@jabber.tula-ix.net.ru JID: GHhost@jabber.tula-ix.net.ru YG129-RIPE YG129-RIPE