Also check out this great write up:<div><br></div><div><a href="http://wiki.freeswitch.org/wiki/Multi_home_tutorial">http://wiki.freeswitch.org/wiki/Multi_home_tutorial</a></div><div><br></div><div>This is pfSense specific.<br>
<br><div class="gmail_quote">On Sat, Dec 5, 2009 at 10:22 AM, ram <span dir="ltr">&lt;<a href="mailto:talk2ram@gmail.com">talk2ram@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;">
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<div class="gmail_quote"><div><div></div><div class="h5">On Sat, Dec 5, 2009 at 3:16 PM, mailinglist <span dir="ltr">&lt;<a href="mailto:mailinglist@fribert.dk" target="_blank">mailinglist@fribert.dk</a>&gt;</span> wrote:<br>

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<div>Has anybody done this?</div>
<div> </div>
<div>I&#39;m completely at a loss, having tinkered very little with Asterisk, and giving up on that, I wonder if there&#39;s any help to be found on FreeSwitch?</div>
<div>Anybody that can give pointers to a good step-by-step instruction?</div>
<div> </div>
<div>I want to have it handle my two sip-phones (siemens dect ip and spa 901), and handle a sip account at my provider.</div>
<div>Of course transferring calls between the two, as well as group calls would be a nice benefit.</div>
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</div></div><div>in short answer Fusionpbx.com</div>
<div> </div><font color="#888888">
<div>Ram</div></font><div class="im">
<div> </div>
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<blockquote style="border-left:#ccc 1px solid;margin:0px 0px 0px 0.8ex;padding-left:1ex" class="gmail_quote">_______________________________________________<br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>

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