I solve it for h323 in follow way: 1. Exclude all codecs except g723.1 from h323.conf:
disallow=ULAW allow=g723.1 2. Add format_g723 module (http://www.agk.nnov.ru/format_g723.c.gz) into project 3. convert all wav and gsm sound into g723 format (use lbccodec from g723_1 demo package, don't ask me where you can download it) 4. set maxsilence=0 in voicemail.conf to suppress conversion into pcm format for silence detection. And it works fine for me. But where are some bugs in h323 module: * not supported g7231 without sound detection (simple to fix). * sometime data transfer (rtp traffic) begins before negotiation complete and first packet is going in g711 codec and channel going down (not yet reviewed). if will any question regards format_g723 module send mail to: f723 >< agk.nnov.ru Friday, January 16, 2004, 10:30:41 PM, Dan Tusa wrote: DT> Hi, DT> Want to do some experiments with the G.723 codecs - where can I download the DT> 723 source code for Asterisk? DT> I know there are some ongoing discussion regarding patents and license fees DT> for the g.723 but I have some hardware on which I only have the 723 and need DT> to test it privately. DT> Thanks! DT> Dan DT> _________________________________________________________________ DT> Use MSN Messenger to send music and pics to your friends DT> http://www.msn.co.uk/messenger DT> _______________________________________________ DT> Asterisk-Users mailing list DT> [EMAIL PROTECTED] DT> http://lists.digium.com/mailman/listinfo/asterisk-users DT> To UNSUBSCRIBE or update options visit: DT> http://lists.digium.com/mailman/listinfo/asterisk-users -- Andrei Koulik. System administrator, Sandy Info Ltd. (ISP), Nizhny Novgorod, Russia _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
