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

[Ray Jackson]

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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