Have a look at: <br><a href="http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication">http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication</a><br><br>It's about IAX but guess will give you some good hints on how to solve your problem.<br>
<br>Alyed<br><br><br><div class="gmail_quote">2010/4/13 Mike Diehl <span dir="ltr"><<a href="mailto:mdiehl@diehlnet.com">mdiehl@diehlnet.com</a>></span><br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi all,<br>
<br>
I'm trying to tighten things up a bit and I seem be be running into something<br>
that doesn't make sense to me.<br>
<br>
I've got 2 contexts, one for customers, and one for guests, that I include<br>
into [customers] and [default], in extensions.conf, as below:<br>
<br>
=============================================================<br>
[default]<br>
include = dial_GUEST<br>
<br>
[customers]<br>
include = parkedcalls<br>
include = dial<br>
=============================================================<br>
<br>
The contexts, dial, and dial_GUEST essentially handle all call routing, with<br>
the idea that guests (anonymous internet callers) can't get out to the pstn.<br>
<br>
The problem is that ALL incoming calls are landing in [customers] even if the<br>
caller is an unregistered SIP client.<br>
<br>
As soon as a call comes in, I see it jump immediately to xxxx@customers:1 and<br>
this happends with registered or unregistered clients.<br>
<br>
What am I doing wrong?<br>
<br>
--<br>
<br>
Take care and have fun,<br>
Mike Diehl.<br>
<br>
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</blockquote></div><br><br clear="all"><br>