Meftah Tayeb wrote: > hello, > yes, host please to let users test it and report bug > YATE/Asterisk fully support but freeswitch no fully support it
I would disagree about YATE & Asterisk fully supporting H323. :) They both have some support for H323 for years now, but only for voice calls. None of the two platforms can do fax or video calls in H323 for example. YATE especially, cannot even pass DTMF between H323 and SIP unless it is in RTP (RFC 2833). I think it is a good opportunity for FS to make a difference if proper H323 support is built for it. Best regards, Vlasis Hatzistavrou. _______________________________________________ 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
