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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