<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Hello,<div><br></div><div>If you would like to make out bound call (from Asterisk to SIP provider), it is fine.</div><div><br></div><div>But if you want have inbound call (from SIP provider to Asterisk). I think you are supposed to have something like this</div><div><br></div><div>sip.conf</div><div><div>register => 5552530146:<your_password>@sip3.voipvoip.com/5552530146</div><div><br></div><div>[5552530146]</div><div>.......</div><div>context=incoming</div><div><br></div><div>extensions.conf</div><div><br></div><div>[incoming]</div><div>;first creating extensions for your local users</div><div><br></div><div>exten => 5552530146,1,Goto(5552530146_incomming,s,1)</div><div><br></div><div>[5552530146_incomming]</div><div>;more logic</div></div><div><br></div><div><br></div><div>wish it would help.</div><div><br></div><div><br></div><div><br></div><div><br></div><div><div><div>On Jul 5, 2012, at 11:44 PM, Thomas Perron wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">I am new. Here is the code that I am playing with on CentOS 6.x<br><br>When I dial the number that corresponds w/ my SIP account I get a recording: "reached a non-working number........"<br><br>I built Asterisk a few times last year and am now back working on a similar project. In my view, there is something wrong in sip.conf<br>
I don't remember using a file that long to get a basic call set up. The format was provided to me by <a href="http://voipvoip.com/">voipvoip.com</a> (the SIP provider).<br><br>Does anyone have any comments please? I just want a very simple config to get my machine to recognize a call to the SIP provider.<br>
<br>Here is results of sip show registry: <br><br>Host dnsmgr Username Refresh State Reg.Time <br><a href="http://sip3.voipvoip.com:5060/">sip3.voipvoip.com:5060</a> N 5552530146 285 Registered Thu, 05 Jul 2012 21:39:56<br>
1 SIP registrations.<br><br>Here is sip and extensions.conf<br><br>sip.conf<br><br>[general]<br>register => <a href="mailto:5552530146%3Afunnytiger123@sip3.voipvoip.com">5552530146:funnytiger123@sip3.voipvoip.com</a><br>
;<br><br>[<a href="http://sip3.voipvoip.com/">sip3.voipvoip.com</a>]<br><br>[outgoing]<br>username=5552530146<br>type=peer<br>qualify=yes<br>secret=funnytiger123<br>nat=auto<br>insecure=very<br>host=69.90.209.57<br>fromuser=5552530146<br>
fromdomain=69.90.209.57<br>dtmfmode=rfc2833<br>allow=g729<br>allow=ilbc<br>allow=ulaw<br>allow=alaw<br>disallow=all<br>srvlookup=no<br><br>[incoming]<br>username=5552530146<br>type=user<br>secret=funnytiger123<br>nat=auto<br>
insecure=very<br>host=69.90.209.57<br>fromdomain=69.90.209.57<br>dtmfmode=rfc2833<br>context=incoming<br>allow=g729<br>allow=ulaw<br>allow=alaw<br>allow=ilbc<br>disallow=all<br>srvlookup=no<br><br><br><br>extensions.conf<br>
<br>[general]<br><br>;<br>;<br>[incoming]<br>;first creating extensions for your local users<br>exten=> s,1,Dial(SIP/17037175555)<br>exten=> s,2,Hangup()<br><br><br><br><br><br><br><br>
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