<br>thanks radamson for the proper explanation, <br><br>actually this question was also posted on the ast-dev list. I believe the issue here is that:<br>is asterisk smart enuff to choose the proper codec over 2 sip channels and not defaulting the the ordering or preference list
<br><br><div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> know how I could make them compatible ?<br><br>I believe the issue is this...
<br><br>When sip1 initiates a call, a codec is selected based on the sip phone<br>preference and asterisk codec "ordering". That selection has nothing to<br>do with "where" the call is going to be directed (eg, sip2 or sip3).
<br>That negotiation happens early, otherwise you would not be able to hear<br>busy & congested tones, audio messages, etc.<br><br>"After" that negotiation happens, then asterisk begins processing the<br>call by doing the same thing with the destination sip phone. In other
<br>words, asterisk negotiates an appropriate codec with sip2 (or sip3) that<br>is based on that phone's codec preference and what asterisk's codec<br>ordering for that sip phone definition.<br><br>"After" both of the above steps are completed, asterisk then tries to
<br>bridge the two calls, and if you don't have the g729 codec installed, it<br>can't bridge ulaw to g729. There is no more codec negotiation going on<br>after step 1 and 2 above.<br><br>The above can easily be verified by simply doing a "sip debug" and
<br>placing a call.<br><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:
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