<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Arial; font-size: 10pt; color: #000000'>Lamine,<br>Welcome to the Asterisk community! Getting into Asterisk can have a big learning curve depending on your telephony/linux background. Fortunately, there are a few resources out there to help with this curve. If you haven't heard of it yet, I'd suggest looking into <http://www.asteriskdocs.org>. This will be a great place to start.<br><br>As for a little bit of more focused help, try thinking of extensions in the following manner. When you get an incoming call you have to tell it where to go. Extensions is the means to do that. Since you only have one incoming line, it makes things really simple. The line comes in to a given dialplan context and searches that context for any matching extensions. Restated: An incoming call walks into a room (the incoming context) and yells "HEY, I HAVE A CALL FOR NXXNXXXXX, ANYONE HERE BY THAT EXTENSION?". The call then looks around and tries to find any match for that and either points to the match or shrugs and walks back out of the room. This is simplified of course, but enough to help understand the basics of extensions.<br><br>Hope it helps.<br><br>-- <br>Ryan Brindley <br>Digium, Inc. | Software Developer <br>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA <br>main: +1 256-428-6000 fax: +1 256-864-0464 <br>Check us out at: http://digium.com & http://asterisk.org<br><br>----- Original Message -----<br>From: "Lamine Ndiaye" <lndiaye@gmail.com><br>To: asterisk-gui@lists.digium.com<br>Sent: Tuesday, February 24, 2009 7:10:03 PM GMT -06:00 US/Canada Central<br>Subject: [asterisk-gui] Extension config<br><br>Hello,<div>I'm starting to use asterisk trying to learn how it work. I'm able to make a call betwen different sip account with X-lite. I can setup it with the astrisk-gui.</div><div><br></div><div><span class="Apple-style-span" style="font-family: -webkit-monospace; font-size: 16px;">Now I have with my VoIP service provider VoiceNetwork that demonstrates how to integrate their service with Asterisk. <br>
The configuration seems to be working with the CLI because I am able to see the incoming call is immediately rejected with this error message. (rejected because extension not found). <br>My question is: It is possible to redirect my call to a sip asterisk. <br>
I read but I do not really understand the extensions concepts. </span></div><div><span class="Apple-style-span" style="font-family: -webkit-monospace; font-size: 16px;">Thank you</span><br>
</div><div>-- <br>Lamine<br>
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