You are usinfg sip or iax ? Its possible to prevent in both cases for sip under peer definition you can put canreinvite=no and in iax2 you can put transfer=no for asterisk 1.4 or notranser=yes in asterisk 1.2 .. Search for this on <a href="http://voip-info.org">voip-info.org</a> wiki for more info .<br>
<br><div class="gmail_quote">On Jan 25, 2008 7:03 PM, <<a href="mailto:asterisk-users@rogg.is">asterisk-users@rogg.is</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
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<div>
<p>I have a call coming in from Asterisk-A going to Asterisk-B
where it's determined that the called party is in fact yet another number
in Asterisk-A so a new call is created from B to A and the two calls bridged
(by Asterisk) at Asterisk-B.</p>
<p> </p>
<p>Originating Caller ==> Asterisk-A ==> Asterisk-B
==> Asterisk-A</p>
<p> </p>
<p>Now, what happens is that in my case both A and B are on the
same network and therefore Asterisk-A apparently optimizes the round-trip to
Asterisk-B out and the original caller talks directly to the extension hosted
in Asterisk-A without the call path going the round-trip to Asterisk-B.</p>
<p> </p>
<p>Is it possible to prevent this optimization from happening?
Any way to control if it happens at all, or can it be selected on per-call
basis somehow?</p>
<p> </p>
<p>Can I find anywhere more details of call path optimization
and it's configuration, use, functionality and behaviour?</p>
<p> </p>
<p>tnx,</p>
<p>Baldvin</p>
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</div>
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