<p>Try using loopback when you send the call to voicemail, also see the local extensions dialplan located in conf/dialplan/default.xml</p>
<div class="gmail_quote">On Mar 19, 2012 4:43 PM, &quot;Robert Longfield&quot; &lt;<a href="mailto:robert.longfield@klinsight.com">robert.longfield@klinsight.com</a>&gt; wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div dir="ltr">
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<div style="FONT-FAMILY:&#39;Verdana&#39;;COLOR:#000000;FONT-SIZE:10pt">
<div><span><font face="Arial"><font style="FONT-SIZE:9.8pt">I set up a group call for our support 
team in which all their phones ring when someone needs to speak with them. If 
they are busy the call should be transferred to a general extension which if not 
answered then goes to that extensions VM.</font></font></span></div>
<div><span><font face="Arial"></font></span> </div>
<div><span><font face="Arial">My dialplan looks like:</font></span></div>
<div><span><font face="Arial"></font></span> </div>
<div>&lt;extension name=&quot;group_dial_support&quot;&gt;</div>
<div>      &lt;condition field=&quot;destination_number&quot; 
expression=&quot;^2001$&quot;&gt;</div>
<div>        &lt;action application=&quot;set&quot; 
data=&quot;continue_on_fail=true&quot;/&gt;</div>
<div>        &lt;action application=&quot;bridge&quot; 
data=&quot;group/support@${domain_name}&quot;/&gt;</div>
<div>        &lt;!-- &lt;action 
application=&quot;voicemail&quot; data=&quot;default ${domain} 1000&quot;/&gt; --&gt;</div>
<div>        &lt;action 
application=&quot;transfer&quot; data=&quot;1000 XML default&quot;/&gt;</div>
<div>        &lt;action 
application=&quot;hangup&quot;/&gt;</div>
<div>     &lt;/condition&gt;</div>
<div>&lt;/extension&gt;</div>
<div> </div>
<div> </div>
<div>What is happening is a caller selects the support option from the IVR, ever 
phone in the support group rings, which is what should happen. If no one picks 
up the call Freeswitch hangs up instead of transferring the call to extension 
1000. You can see that I also tried to send the call directly to voicemail but 
that didn’t work either.</div>
<div> </div>
<div>The message I see when Freeswitch hangs up is:</div>
<div> </div>
<div>    Channel sofia/internal/<a href="http://sip:1002@72.38.184.18:39042" target="_blank">sip:1002@72.38.184.18:39042</a> 
hanging up, cause: USER_BUSY</div>
<div> </div>
<div>The full output from cli can be seen here: <a href="http://pastebin.freeswitch.org/18696" target="_blank"><font face="Times New Roman"><font style="FONT-SIZE:12pt">http://pastebin.freeswitch.org/18696</font></font></a><font face="Times New Roman"><font style="FONT-SIZE:12pt"> </font></font></div>

<div> </div>
<div>I would like to get the call to transfer properly.</div>
<div> </div>
<div>Thanks</div>
<div style="FONT-FAMILY:&#39;Verdana&#39;;COLOR:#000000;FONT-SIZE:10pt">-Robert<br></div></div></div></div>
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<br></blockquote></div>