[Asterisk-Users] One SIP peer use 2 diff codecs?

[Roy Sigurd Karlsbakk]

>> How about this variable? :-)
>> ${SIP_CODEC}: Used to set the SIP codec for a call
>
> That only works for calls going OUT from Asterisk.  It does nothing 
> for incoming calls.  By the time the dialplan is called the codec is 
> already set.

perhaps tha should be changed, then, to allow more control over codec 
handling?

roy