[asterisk-users] Asterisk 1.4 Conference with G.722
digium at sanguinarius.co.uk
Thu Apr 26 06:25:50 MST 2007
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.
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