<br><br>
<div class="gmail_quote">On Sat, Dec 5, 2009 at 3:16 PM, mailinglist <span dir="ltr"><<a href="mailto:mailinglist@fribert.dk">mailinglist@fribert.dk</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div style="MARGIN: 4px 4px 1px; FONT: 10pt Segoe UI">
<div>Has anybody done this?</div>
<div> </div>
<div>I'm completely at a loss, having tinkered very little with Asterisk, and giving up on that, I wonder if there'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>
<div> </div></div><br></blockquote>
<div> </div>
<div> </div>
<div>in short answer Fusionpbx.com</div>
<div> </div>
<div>Ram</div>
<div> </div>
<div> </div>
<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">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org</a><br><br></blockquote></div><br>