Pastebin the debug output of the FreeSWITCH log.<br clear="all"><br><div>Moises Silva<br>Senior Software Engineer<br>Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada<br>t. 1 905 474 1990 x 128 | e. <a href="mailto:moy@sangoma.com">moy@sangoma.com</a><br>

<br><br><div class="gmail_quote">On Wed, May 26, 2010 at 5:12 PM, Sean Holt <span dir="ltr">&lt;<a href="mailto:sean@obscuradigital.com">sean@obscuradigital.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">




<div>
<font face="Calibri, Verdana, Helvetica, Arial"><span style="font-size:11pt">Hello all,<br>
<br>
I’ve been hacking at this issue for the last 3 days and I can’t get beyond this error when attempting to call out through my sangoma card.<br>
<br>
  SIP/2.0 503 Service Unavailable<br>
   Via: SIP/2.0/UDP 192.168.100.6:12026;branch=z9hG4bK-d8754z-77564c04563bac12-1---d8754z-;rport=12026<br>
   From: &quot;Sean Holt&quot;&lt;sip:<a href="http://2028@172.16.15.10" target="_blank">2028@172.16.15.10</a>&gt;;tag=25dacb3d<br>
   To: &lt;sip:<a href="http://5102079553@172.16.15.10" target="_blank">5102079553@172.16.15.10</a>&gt;;tag=4St9j3N9yXr8g<br>
   Call-ID: NWViZTI0ODljMmY1YTM5ZDRiZmQzNTUwYzgwNWVkMDQ.<br>
   CSeq: 1 INVITE<br>
   User-Agent: Configured by TCAPI<br>
   Accept: application/sdp<br>
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE<br>
   Supported: precondition, path, replaces<br>
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br>
   Reason: Q.850;cause=34;text=&quot;NORMAL_CIRCUIT_CONGESTION&quot;<br>
   Content-Length: 0<br>
   Remote-Party-ID: &quot;5102079553&quot; &lt;sip:<a href="http://5102079553@172.16.15.10" target="_blank">5102079553@172.16.15.10</a>&gt;;party=calling;privacy=off;screen=no<br>
<br>
Here’s my configuration:<br>
I’m trying to call out from my internal voip phone through my sangoma card b600.  My phone line is plugged into the FXO port.  Module (mod_freetdm) is successfully loaded<br>
Dialplan:<br>
&lt;extension name=&quot;main_number_53&quot;&gt;<br>
      &lt;condition field=&quot;destination_number&quot; expression=&quot;^(\d{11})$|^(\d{10})$$&quot;&gt;<br>
       &lt;action application=&quot;bridge&quot; data=&quot;freetdm/2/2/$1&quot;/&gt;<br>
      &lt;/condition&gt;<br>
 &lt;/extension&gt; <br>
<br>
Freetdm.conf file<br>
[span wanpipe wp1]<br>
name =&gt; FreeTDM<br>
number =&gt; 1<br>
trunk_type =&gt; FXS<br>
fxs-channel =&gt; 1:5<br>
<br>
[span wanpipe wp2]<br>
name =&gt; FreeTDM<br>
number =&gt; 2<br>
trunk_type =&gt; FXO<br>
fxo-channel =&gt; 1:1-4<br>
<br>
Freetdm.conf.xml file<br>
&lt;configuration name=&quot;freetdm.conf&quot; description=&quot;FreeTDM Configuration&quot;&gt;<br>
  &lt;settings&gt;<br>
    &lt;param name=&quot;debug&quot; value=&quot;9&quot;/&gt;<br>
    &lt;!--&lt;param name=&quot;hold-music&quot; value=&quot;$${moh_uri}&quot;/&gt;--&gt;<br>
  &lt;/settings&gt;<br>
   &lt;analog_spans&gt;<br>
           &lt;span name=&quot;wp1&quot;&gt;<br>
                &lt;param name=&quot;context&quot; value=&quot;context_1&quot;/&gt;<br>
                &lt;param name=&quot;dialplan&quot; value=&quot;XML&quot;/&gt;<br>
           &lt;/span&gt;<br>
            &lt;span name=&quot;wp2&quot;&gt;<br>
                &lt;param name=&quot;context&quot; value=&quot;context_1&quot;/&gt;<br>
                &lt;param name=&quot;dialplan&quot; value=&quot;XML&quot;/&gt;<br>
            &lt;/span&gt;<br>
   &lt;/analog_spans&gt;<br>
&lt;/configuration&gt;<br>
<br>
Let me know if I need to provide additional info<br>
<br>
Thanks<br><font color="#888888">
Sean</font></span></font>
</div>


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<br></blockquote></div><br></div>