TienSen Chong wrote: > Hi all, > > I am having problem with conference call (meetme feature) using G.722 > phone. G.722 phone to phone is working fine. I suspect this is due to > the fact that Asterisk 1.4 only support G.722 passthrough. > This will be the case, Meetme transcodes the audio (to slin iirc), where it mixes it.
> Any ideas how this problem can be fixed. > Have you tried using app_conference? To be honest, I don't know how you would be able to have more than 2 people in a call without some transcoding going on. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
