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

[Eric "ManxPower" Wieling]

Ray Jackson wrote:
> Hi Andrew,
> 
> All that happens is Asterisk sets up the call using G723 as the codec 
> and calls then fail due to transcoding problems (since it cannot 
> encode/decode g723).  The softphone always prefers g723 over gsm, so if 
> you allow both in sip.conf Asterisk always select g723 due to the 
> preference of the client.  What I need is for Asterisk to understand 
> what the 'other end' is capable of before selecting a codec. i.e. if the 
> call terminates to the Voicemail then select gsm and do NOT allow g723.  
> Codec negotiation is usually something that occurs in the call setup 
> between 2 endpoints, but Asterisk seems to do this in isolation for each 
> client without understanding where the call is going first to make that 
> decision.  What I really need is something that forces the 'allow' 
> codecs in the dial plan per call.  Does anybody know if this is this 
> possible?

You can force the codec for an OUTOGING call.  You cannot force the 
codec for an INCOMING call because by the time the call hits the 
dialplan all that sort of stuff is already negotiated.