[asterisk-users] G723 pass-through and codec negotiation

[Andrew Joakimsen]

What happens if in your sip.conf you set

disallow=all
allow=g723,gsm

And then allow both codec in the phone?

On 11/19/06, Ray Jackson <ray at jacksonz.net> wrote:
>
> All,
>
> Our users have a softphone client that supports the G723 Codec which we
> want to use for bandwidth reasons, however we do not wish (or have the
> funds) to license the codec on our Asterisk servers.  We have G723
> pass-through working between the clients just fine, however calls fail
> when terminating with Asterisk itself (i.e. Voicemail) or out to the
> PSTN due to transcoding issues.
>
> If it possible to build the config into our Asterisk servers so that
> calls between the softphones defaults to G723 pass-through, whilst all
> other calls (PSTN, Voicemail etc.) default to GSM as their preferred
> codec?  Is there a way of getting Asterisk to be smart with Codec
> negotitation and figure out which codec the other end of the call is
> capable of before negotitating back to the Softphone with the selected
> codec?  I assume you would have to do something in the dial plan?  I saw
> the SIP_CODEC variable, but couldn't make it work.
>
> Any advice would be very welcome!
>
> Cheers,
> Ray
>
>
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