[asterisk-users] Problems with Codecs in Asterisk

[Rosli Sukri]

thanks radamson for the proper explanation,

actually this question was also posted on the ast-dev list. I believe the
issue here is that:
is asterisk smart enuff to choose the proper codec over 2 sip channels and
not defaulting the the ordering or preference list

         know how I could make them compatible ?
>
> I believe the issue is this...
>
> When sip1 initiates a call, a codec is selected based on the sip phone
> preference and asterisk codec "ordering". That selection has nothing to
> do with "where" the call is going to be directed (eg, sip2 or sip3).
> That negotiation happens early, otherwise you would not be able to hear
> busy & congested tones, audio messages, etc.
>
> "After" that negotiation happens, then asterisk begins processing the
> call by doing the same thing with the destination sip phone. In other
> words, asterisk negotiates an appropriate codec with sip2 (or sip3) that
> is based on that phone's codec preference and what asterisk's codec
> ordering for that sip phone definition.
>
> "After" both of the above steps are completed, asterisk then tries to
> bridge the two calls, and if you don't have the g729 codec installed, it
> can't bridge ulaw to g729. There is no more codec negotiation going on
> after step 1 and 2 above.
>
> The above can easily be verified by simply doing a "sip debug" and
> placing a call.
>
>
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