<div dir="ltr"><div>I am trying to make sure my DID and SIP account details are working properly and engaging the extensions.conf and dial plan.<br><br></div>I have a successful SIP session registered:<br><br>Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)<br>
Asterisk*CLI&gt; sip show registry<br>Host                                    dnsmgr Username       Refresh State                Reg.Time<br><a href="http://sip3.voipvoip.com:5060">sip3.voipvoip.com:5060</a>                  N      1112530146         105 Registered           Mon, 08 Apr 2013 06:02:09<br>
1 SIP registrations.<br>Asterisk*CLI&gt;<br><br><div>Here is the dial plan:<br>[incoming]<br>exten =&gt; 17036361355,1,Playback(beep)<br>exten =&gt; 17036361355,2,SayDigits(${EXTEN})<br>exten =&gt; 17036361355,3,Goto(testdtmf|s|1<br>
;Ring on Elle  mobile phone.<br>;exten =&gt; s,1,Answer()<br>;exten =&gt; s,n,Dial(SIP/17037171234,150,r,t,)<br><br><br>[general]<br>register =&gt;1112530146:albany!@#<a href="http://123@sip3.voipvoip.com/1112530146">123@sip3.voipvoip.com/1112530146</a><br>
registertimeout=20<br>context=incoming<br>allowoverlap=no<br>bindport=5060<br>bindaddr=192.168.1.10<br>srvlookup=no<br>;context=incoming<br><br>; The SIP provider<br>[<a href="http://voipvoip.com">voipvoip.com</a>]<br>canreinvite=no<br>
username=1112530146<br>fromuser=1112530146<br>secret=albany!@#123<br>context=incoming<br>type=friend<br>fromdomain=<a href="mailto:sip3@voipvoip.com">sip3@voipvoip.com</a><br>host=69.90.209.57<br>dtmfmode=rfc2833<br>disallow=all<br>
allow=alaw<br>allow=ulaw<br>nat=force_rport<br>insecure=port,invite<br><br></div><div>Thoughts please?    I think something to do w/ &quot;incoming&quot; is incorrect.<br><br><br></div></div>