R: R: [Asterisk-Users] How to force G729

[Rich Adamson]

> >Define that per user.
> 
> 
> Of course... The user part is not the problem. If I force a user in its extensions to 
use G729 only, he actually talks G729 to Asterisk, but asterisk still talks ULAW to the 
PSTN gateway, doing the transcoding. This is driving me crazy...
> 

"If" I understood your initial objective correctly (and I may not have),
the user's phones are negotiating the codec to be used for each rtp
session.

Asterisk parameters can be used to dictate rtp sessions between the
sip phone and asterisk, but that won't influence the next step in
which the sip phone negotiates a new rtp session directly with the 
gateway.

The gateway and the phone will negotiate a common codec based on
whatever logic those two devices have been programmed with by their
respective manufacturers; asterisk isn't involved.

So, it sounds like the issue is understanding the codec selection
logic that has been programmed into the gateway and the phone.