On 6/6/06, M.Hockings <[EMAIL PROTECTED]> wrote:
William Piper wrote:
> For Problem #2:
> I'm not sure what you are asking. Perhaps post your dialplan for this
> problem & we will take a look.
>
> bp
>
> On 6/4/06, *M.Hockings* <[EMAIL PROTECTED]
> <mailto: [EMAIL PROTECTED]>> wrote:
>
> Problem 2) Incoming sip calls from my voip provider get rejected unless
> I allow anyone to connect with sip. I have an incoming route set up with
> the right DID that matches the DID that asterisk picks out but it still
> rejects the call. Any suggestions about how to get this to work without
> allowing any sip connection?
>
>
> Mike
Hi William, at the bottom of this is my extensions.conf which seems to
be the largest part of the equation for problem #2. I have not applied
any changes to try and resolve my problem #1 yet.
I think the question here is the operation of the following statement in
the [from-sip-external] section:
exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?from-trunk,${DID},1)
If I interpret it correctly it should go to from-trunk,1 if the freePBX
"allow anonymous sip connections" is true and go to
incoming-sip-did-value,1 if it is false ? That is should I be looking
for something like this in the config files to understand how this would
be handled?
exten=>4169671111,1,....
As an aside, is there some beginners guide to understanding dial plans?
My original dial plan (based on things read on voip-info.org ) was very
simple and worked as far as it was configured. I have recently gone to
freePBX to try and make the dial plan changes easier and faster however
it adds a lot of gorp like this that I don't understand.
Thanks for any guidance on this,
Mike
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.
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.
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.
Regards,
bp
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