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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