Thats really strange .. if you have made canreinvite=no then it should not even attampt native bridging and should transcode codecs ..something's fishy here .. Also try to put canreinvite=no in testulaw exntension too .<br>
<br><div><span class="gmail_quote">On 16/11/06, <b class="gmail_sendername">Victor Toofic</b> <<a href="mailto:toofics@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">toofics@gmail.com
</a>> wrote:</span><blockquote class="gmail_quote" style="margin-top: 0; margin-right: 0; margin-bottom: 0; margin-left: 0; margin-left: 0.80ex; border-left-color: #cccccc; border-left-width: 1px; border-left-style: solid; padding-left: 1ex">
El jue, nov 16 de 2006 a las 18:28 +0530, Vicky comentaba:<br>> g729 is not a free codec . YOu have to buy it from digium at rateof $10 per<br>> channel license . If you are just using asterisk and havent bought g729
<br>> license then asterisk will just do bridging of g729 and wont edit/transcode<br>> stream .<br>><br>> On 16/11/06, Victor Toofic <<a href="mailto:toofics@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
toofics@gmail.com</a>> wrote:<br>> > <br>> >I have the following scenario:<br>> ><br>> > g729 gsm<br>> > UAS <-----------> * <-----------> UAC<br>> >
<br>> >I am using sipp to generate the calls between the UAC and the UAS and <br>> >sending some rtp from the UAC, I want * to do transcoding but as I see<br>> >it is not. As long as I know 'Attempting native bridge' means only
<br>> >passing-through the rtp ¿Am I wrong?<br><br>I get the same message even if I'm not using g729: <br><br> --Attempting native bridge of SIP/testgsm-081784b0 and SIP/testulaw-0817da80<br><br> ulaw gsm
<br> UAS <-----------> * <-----------> UAC<br><br>_______________________________________________ <br>--Bandwidth and Colocation provided by <a href="http://Easynews.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
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