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<div class="gmail_quote">On Thu, Dec 31, 2009 at 12:10 PM, Kevin P. Fleming <span dir="ltr"><<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>></span> wrote:<br>
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<div class="h5">hadi motamedi wrote:<br><br>> Can you please let me know if we can have different codec schemes for<br>> audio codec in & audio codec out ? I mean , in one application , we<br>> can have our audio codec input set to G.711 a-law and our audio codec<br>
> output set to G.711 u-law . I am facing with an application that calls<br>> for such a settings .<br><br></div></div>Asterisk does not support asymmetric codec configurations.<br><br>--<br>Kevin P. Fleming<br>Digium, Inc. | Director of Software Technologies<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>skype: kpfleming | jabber: <a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a><br>Check us out at <a href="http://www.digium.com/" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org/" target="_blank">www.asterisk.org</a><br>
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<div>Sorry . I didn't get the point clearly . In the SIP Invite message , it says "my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C". The remote endpoint responds with a 200 OK, saying "my audio stream is at IP y.y.y.y port y, and I choose codec B". Can you please do me favor and let me know if my understanding is right or not ?</div>
<div>Thank you</div>
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