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

[Manuel Wenger]

> "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.


I think you're getting my point, at least I think so (I'm getting more and more confused myself about this...)

The problem is that the phone negotiates a codec with asterisk when placing the call (remember I have all reinvite's set to "no", so the gateway and the phone won't talk directly to each other!). This negotiation actually works correctly, because I force the phone's codec using "disallow=all; allow=g729" in the SIP phone's peer configuration. 

The negotiation which doesn't work the way I want is the asterisk-to-gateway part. The gateway can talk either ULAW or G729, whatever I tell it, if I "force" it using the disallow/allow method in sip.conf. The problem is that I need asterisk to talk to the gateway sometimes with ULAW, sometimes with G729, depending on the SIP phone who placed the call in the first place.

What I need is some sort of command which says "OK, now Dial(... @gateway), but force G729" (which works *if* I tell asterisk that the gateway supports G729 *only* in sip.conf, but we want it to support both codecs, right?). Apparently I can only force the codec on incoming channels, not on outgoing channels. Is this really an asterisk limitation?

-Manuel


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