<div>I haven't tried the app_conference yet. I want to know if the conference is consisting of 3 users with G.722, does the app_conference perform transcoding? If it is not, then app_conference will solve the issue of having conference consists of only
G.722 user since no transcoding is needed. Is my understanding correct?</div>
<div> </div>
<div>Regards,</div>
<div>chong<br><br> </div>
<div><span class="gmail_quote">On 4/26/07, <b class="gmail_sendername">Thomas Kenyon</b> <<a href="mailto:digium@sanguinarius.co.uk">digium@sanguinarius.co.uk</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">TienSen Chong wrote:<br>> Hi all,<br>><br>> I am having problem with conference call (meetme feature) using
G.722<br>> phone. G.722 phone to phone is working fine. I suspect this is due to<br>> the fact that Asterisk 1.4 only support G.722 passthrough.<br>><br>This will be the case, Meetme transcodes the audio (to slin iirc), where
<br>it mixes it.<br><br>> Any ideas how this problem can be fixed.<br>><br>Have you tried using app_conference?<br>To be honest, I don't know how you would be able to have more than 2<br>people in a call without some transcoding going on.
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