<div>thanks for the information Peter, its really helpful. Also I have one more question - do you have any idea how many such simultaneous calls can an asterisk server handle (say running od 2.6Ghz, 1GB Ram, fedora machine)?
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<div>Thanks,</div>
<div>Nitin<br><br> </div>
<div><span class="gmail_quote">On 2/13/06, <b class="gmail_sendername">Peter Fern</b> <<a href="mailto:pete@keypoint.com.au">pete@keypoint.com.au</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">You can enable this on a per-peer basis with:<br><br>sip peers:<br>canreinvite=yes<br><br>iax peers:<br>notransfer=no
<br><br>Check the iax.conf.sample and sip.conf.sample files for usage.<br><br>Nitin Gupta wrote:<br><br>> Hi I was wondering if its possible to make Dial command bridge two<br>> channels and after bridging bypass asterisk, so that the voice doesn't
<br>> need to pass through my asterisk server.<br>><br>> For e.g., I have a user dialed in and he verifies himself and then<br>> dials an international extension, after the call connects I don't want<br>> the call to pass through asterisk server anymore. Is there any command
<br>> already there for any particular channel type?<br>><br>> Thanks,<br>> Nitin<br>><br>>------------------------------------------------------------------------<br>><br>>_______________________________________________
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