I&#39;m pretty sure the zoom does not support sip originated calls to the FXO port.  It&#39;s FXO port is strictly used as failover or selectable via dialplan when the call originates from the FXS port (eg: dial 9 first to get FXO).  <div>
<br></div><div>Try:</div><div><br></div><div>cisco 3102</div><div>audiocodes</div><div>grandstream</div><div><br></div><div>for atas that support full FXS/FXO operation.<br><br><div class="gmail_quote">On Mon, Aug 9, 2010 at 1:49 PM, Ken Gillett <span dir="ltr">&lt;<a href="mailto:ken@ukgb.net">ken@ukgb.net</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">I am actually trying this out with a Zoom 5801 which with an FXO and FXS port and the ability to bridge in both directions can apparently do what I require, but I cannot get my head around what I am even trying to get it to do. And this is before I&#39;ve even thought about bringing FreeSwitch into the equation.</blockquote>
</div><br><br clear="all"><br>-- <br>-Rupa<br>
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