<div dir="ltr">&quot;<span class="Apple-style-span" style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(255,255,255)">If no extension is given, the &#39;s&#39; extension is used. The extension needs to</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(255,255,255)">

<span class="Apple-style-span" style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(255,255,255)">be defined in extensions.conf to be able to accept calls from this SIP provider.&quot;</span><div>

<font class="Apple-style-span" face="Verdana, Geneva, Arial, Helvetica, sans-serif"><span class="Apple-style-span" style="font-size:12px">That sounds like &quot;when we get an incoming call, which extension should it go to?&quot;</span></font></div>

<div><font class="Apple-style-span" face="Verdana, Geneva, Arial, Helvetica, sans-serif"><span class="Apple-style-span" style="font-size:12px"><br></span></font></div><div><font class="Apple-style-span" face="Verdana, Geneva, Arial, Helvetica, sans-serif"><span class="Apple-style-span" style="font-size:12px">Looks like the equivalent would be:</span></font></div>

<div><pre style="padding-top:1em;padding-right:1em;padding-bottom:1em;padding-left:1em;border-top-width:1px;border-right-width:1px;border-bottom-width:1px;border-left-width:1px;border-top-style:dashed;border-right-style:dashed;border-bottom-style:dashed;border-left-style:dashed;border-top-color:rgb(47,111,171);border-right-color:rgb(47,111,171);border-bottom-color:rgb(47,111,171);border-left-color:rgb(47,111,171);background-color:rgb(249,249,249);line-height:1.1em;font-size:10px">

&lt;!--/// extension for inbound calls: *optional* same as username, if blank ///--&gt;
&lt;param name=&quot;extension&quot; value=&quot;cluecon&quot;/&gt;</pre></div><div><font class="Apple-style-span" face="Verdana, Geneva, Arial, Helvetica, sans-serif"><span class="Apple-style-span" style="font-size:12px"><br clear="all">

</span></font><div dir="ltr"><span style="font-family:Verdana,Arial,Helvetica,sans-serif"><span style="font-size:small">-Avi</span></span></div><br>
<br><br><div class="gmail_quote">On Fri, Jan 13, 2012 at 9:16 AM, Nandy Dagondon <span dir="ltr">&lt;<a href="mailto:gcd@i.ph">gcd@i.ph</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

hi bob,<br><br>i found in <a href="http://www.voip-info.org/wiki/view/Asterisk+SIP+register" target="_blank">http://www.voip-info.org/wiki/view/Asterisk+SIP+register</a> this sip.conf directive:<br><pre><code>register =&gt; user[:secret[:authuser]]@host[:port][/extension]</code></pre>


if i&#39;m correct, there&#39;s NO equivalent of &quot;extension&quot; in FS. so i would try:<br> <br>&lt;include&gt;<br>
        &lt;gateway name=&quot;pchdba&quot;&gt;<br>
        &lt;param name=&quot;username&quot; value=&quot;0000*1&quot;/&gt;<br>       
&lt;param name=&quot;password&quot; value=&quot;secret&quot;/&gt;<br>       
&lt;param name=&quot;auth-username&quot; value=&quot;authuser&quot;/&gt;<br>       
&lt;param name=&quot;extension&quot; value=&quot;pchdba&quot;/&gt;<br>       
&lt;param name=&quot;expire-seconds&quot; value=&quot;60&quot;/&gt;<br>        &lt;param name=&quot;proxy&quot; value=&quot;sip-host:port&quot;/&gt;  &lt;!-- port values: 5060 (internal), 5080 (external) --&gt;<br>        &lt;param name=&quot;register&quot; value=&quot;true&quot;/&gt;<br>


       
&lt;/gateway&gt;<br>
&lt;/include&gt;<br>
<br>hope it helps.<div class="HOEnZb"><div class="h5"><br><br><div class="gmail_quote">On Thu, Jan 12, 2012 at 4:46 PM, Bob Smith <span dir="ltr">&lt;<a href="mailto:gb10hkzo-freeswitch@yahoo.co.uk" target="_blank">gb10hkzo-freeswitch@yahoo.co.uk</a>&gt;</span> wrote:<br>


<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Problem is Michael that I don&#39;t have an Asterisk box, that config line was from some instructions on their website where they document an Asterisk setup.<br>



<br>
I&#39;d rather not download Asterisk and try to get to grips with something that I&#39;ll never have a need to use again, since getting to grips with FreeSwitch is keeping me busy enough !<br>
<br>
<br>
&gt;I would do is get a siptrace from it working on your Asterisk box and compare that with the sip trace from the freeswitch box.<br>
<div><div><br>
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