Davin O'Neill wrote: > I have Asterisk running on a Linux 2.4.x box with ztdummy. Once I > did a modprobe on ztdummy I was able to enter into a conference room > using my softphone clients. I'm using SJphone and Firefly. I have > noticed a significant delay (1 to 3 seconds) while talking within the > conference room. I have tried both clients, SIP and IAX protocols > and various codecs. I have also tried it from different host > machine. They are all on the same LAN, so that shouldn't be an > issue. I can call a client directly with SIP or IAX and have clear, > timely audio. I have also done echo tests (dialing 600) through > Asterisk and that works fine too. The delay only occurs within the > conference room. I'm wondering if I just need to purchase one of the > zaptel cards. I would appreciate any thoughts or suggestions. > > Thanks!
try adding "q" flag to meetme app ... _______________________________________________ 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
