I haven't tried the app_conference yet. I want to know if the conference is consisting of 3 users with G.722, does the app_conference perform transcoding? If it is not, then app_conference will solve the issue of having conference consists of only G.722 user since no transcoding is needed. Is my understanding correct?
Regards, chong On 4/26/07, Thomas Kenyon <[EMAIL PROTECTED]> wrote:
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
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