Chris,<br><br>Welcome to the FS community. Glad to have you. It might be easier for you to hop on the IRC channel and talk to some of the guys there. Go to #freeswitch on <a href="http://irc.freenode.net">irc.freenode.net</a>. We usually have a few hundred people in there and one of them is bound to be able to discuss your questions.<br>
<br>-MC (IRC: mercutioviz)<br><br><div class="gmail_quote">On Wed, Feb 8, 2012 at 11:01 PM, chris <span dir="ltr">&lt;<a href="mailto:tknchris@gmail.com">tknchris@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello,<br><br>I am new to FS but I&#39;m impressed already with my first experiences with it. I was just wondering about somethins that I&#39;ve always wanted on asterisk that was possible but always been a pain. Is it possible to do source based routing and route certain SIP extensions/devices to certain outbound trunks? If so how can I do it easily?<div>

<br></div><div>Thanks for any advice / info,</div><span class="HOEnZb"><font color="#888888"><div>chris</div><div><br></div><div><br></div>
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