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<div><span class="gmail_quote">On 6/6/06, <b class="gmail_sendername">M.Hockings</b> <<a href="mailto:veeshooter@hockings.net">veeshooter@hockings.net</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">William Piper wrote:<br>> For Problem #2:<br>> I'm not sure what you are asking. Perhaps post your dialplan for this
<br>> problem & we will take a look.<br>><br>> bp<br>><br>> On 6/4/06, *M.Hockings* <<a href="mailto:veeshooter@hockings.net">veeshooter@hockings.net</a><br>> <mailto:<a href="mailto:veeshooter@hockings.net">
veeshooter@hockings.net</a>>> wrote:<br>><br>> Problem 2) Incoming sip calls from my voip provider get rejected unless<br>> I allow anyone to connect with sip. I have an incoming route set up with<br>
> the right DID that matches the DID that asterisk picks out but it still<br>> rejects the call. Any suggestions about how to get this to work without<br>> allowing any sip connection?<br>><br>>
<br>> Mike<br><br>Hi William, at the bottom of this is my extensions.conf which seems to<br>be the largest part of the equation for problem #2. I have not applied<br>any changes to try and resolve my problem #1 yet.
<br><br>I think the question here is the operation of the following statement in<br>the [from-sip-external] section:<br><br>exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?from-trunk,${DID},1)<br><br>
If I interpret it correctly it should go to from-trunk,1 if the freePBX<br>"allow anonymous sip connections" is true and go to<br>incoming-sip-did-value,1 if it is false ? That is should I be looking<br>for something like this in the config files to understand how this would
<br>be handled?<br><br>exten=>4169671111,1,....<br><br>As an aside, is there some beginners guide to understanding dial plans?<br>My original dial plan (based on things read on <a href="http://voip-info.org">voip-info.org
</a>) was very<br>simple and worked as far as it was configured. I have recently gone to<br>freePBX to try and make the dial plan changes easier and faster however<br>it adds a lot of gorp like this that I don't understand.
<br><br>Thanks for any guidance on this,<br><br>Mike<br></blockquote></div>
<div>I have no idea about FreePBX. I thought you were trying to create something from new. I believe that Asterisk @ home has a list of thier own, you may want to check there. </div>
<div><br>From my personal experience, Asteirsk @ Home is really good for the AMP, but to make it work, I deleted the extensions.conf and created my own then only work directly in the extensions.conf file, not AMP. Just use AMP for reports & such.
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<div>I wish I could help you but I can't spend half the day trying to figure out how FreePBX works then figure out your problem.</div>
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<div>Regards,</div>
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<div>bp<br> </div>