asterisk users - Apr 2010 - All incoming calls landing in [customers] context

Hi all,

I'm trying to tighten things up a bit and I seem be be running into
something
that doesn't make sense to me.

I've got 2 contexts, one for customers, and one for guests, that I include 
into [customers] and [default], in extensions.conf, as below:

============================================================[default]
include = dial_GUEST

[customers]
include = parkedcalls
include = dial
============================================================
The contexts, dial, and dial_GUEST essentially handle all call routing, with 
the idea that guests (anonymous internet callers) can't get out to the pstn.

The problem is that ALL incoming calls are landing in [customers] even if the 
caller is an unregistered SIP client.  

As soon as a call comes in, I see it jump immediately to xxxx at customers:1 and
this happends with registered or unregistered clients.

What am I doing wrong?

-- 

Take care and have fun,
Mike Diehl.

Have a look at:
http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication

It's about IAX but guess will give you some good hints on how to solve your
problem.

Alyed


2010/4/13 Mike Diehl <mdiehl at diehlnet.com>
> Hi all,
>
> I'm trying to tighten things up a bit and I seem be be running into
> something
> that doesn't make sense to me.
>
> I've got 2 contexts, one for customers, and one for guests, that I
include
> into [customers] and [default], in extensions.conf, as below:
>
> ============================================================> [default]
> include = dial_GUEST
>
> [customers]
> include = parkedcalls
> include = dial
> ============================================================>
> The contexts, dial, and dial_GUEST essentially handle all call routing,
> with
> the idea that guests (anonymous internet callers) can't get out to the
> pstn.
>
> The problem is that ALL incoming calls are landing in [customers] even if
> the
> caller is an unregistered SIP client.
>
> As soon as a call comes in, I see it jump immediately to xxxx at
customers:1
> and
> this happends with registered or unregistered clients.
>
> What am I doing wrong?
>
> --
>
> Take care and have fun,
> Mike Diehl.
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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You need to post your sip.conf and any included files in it.

Zeeshan A Zakaria

--
Sent from my Android phone with K-9 Mail.

On 2010-04-13 2:04 AM, "Mike Diehl" <mdiehl at diehlnet.com>
wrote:

Hi all,

I'm trying to tighten things up a bit and I seem be be running into
something
that doesn't make sense to me.

I've got 2 contexts, one for customers, and one for guests, that I include
into [customers] and [default], in extensions.conf, as below:

============================================================[default]
include = dial_GUEST

[customers]
include = parkedcalls
include = dial
============================================================
The contexts, dial, and dial_GUEST essentially handle all call routing, with
the idea that guests (anonymous internet callers) can't get out to the pstn.

The problem is that ALL incoming calls are landing in [customers] even if
the
caller is an unregistered SIP client.

As soon as a call comes in, I see it jump immediately to xxxx at customers:1
and
this happends with registered or unregistered clients.

What am I doing wrong?

--

Take care and have fun,
Mike Diehl.

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
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