I'm pretty sure the zoom does not support sip originated calls to the FXO port. It'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"><<a href="mailto:ken@ukgb.net">ken@ukgb.net</a>></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've even thought about bringing FreeSwitch into the equation.</blockquote>
</div><br><br clear="all"><br>-- <br>-Rupa<br>
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