[Asterisk-Users] Re: fine-tuning asterisk questions

[William Piper]

On 6/6/06, M.Hockings <veeshooter at hockings.net> 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* <veeshooter at hockings.net
> > <mailto:veeshooter at hockings.net>> 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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