<div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">I would be very surprised if that were true. Your phones speak many<br>
codecs, but they negotiate with asterisk on registration which one they<br>
will be using. They don't switch codecs based on the remote channel<br>
(which they don't even know about). Today, if your phones are negotiating<br>
729 on registration, you are definitely transcoding calls to/from the<br>
PSTN.<br></blockquote><div><br></div><div>Assuming we're talking about SIP (and any other voip protocol I know of for that matter), that is incorrect, codec negotiation is done during the call setup based on preferences stored in both Asterisk and the phone (that is what the SDP is for among other things). However at that point Asterisk does not know that the dial plan is going to call the voicemail application to play a file in g729 format (how can possibly now that), and therefore when the file is being played the phone already expects the audio for the call in the format negotiated during call setup which may or may not be g729. Not sure if a re-invite could be issued to change the codec type in the middle of the call, but I suppose it should be possible to implement.</div>
<div><br></div></div>-- <br>Moises Silva<br>Software Developer<br>Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada<br>t. 1 905 474 1990 x 128 | e. <a href="mailto:moy@sangoma.com">moy@sangoma.com</a><br>