VoIP Mailing List Archives

[Mike at Trest.COM]

Hmmm,

IMHO this is a fundamental SIP architecture issue.

To meet my understanding of distribution, this would required a proxy
function before the call "answer()" on the Asterisk. If , in an
ideal world, this proxy function were to be in the path before
answer(), the proxy would need added intelligence to examine the
INVITE and deduce from it's content the need for IVR or VoiceMail.
This is an Active Call Director functionality. IMHO, this well
within the capabilities of several SIP proxy packages available today.

Once answered, the Asterisk will remain in the call path. If that
is ok in your need for distribution, you can push the call onward via
any Asterisk dialplan extension to be serviced by a whole farm of
other Asterisks for specific chores.

..mike..


At 07:03 AM 4/25/2008, you wrote:

[stotaro at totarotechn...]

On Fri, Apr 25, 2008 at 7:03 AM, gmail <gres.rasmy at gmail.com> wrote:

[breeveslist at eusnetw...]

You have the basic idea right, the dial plans are limited down to
specific functions to be provided and then told how to connect to
other features. For example you might have a box that only provide
PSTN connectivity so all calls come in and the dial plan routes the
calls to another box or boxes. I prefer to use Dundi in larger setups
to avoid multiple IAX trunks having to be configured.

On Mon, Apr 28, 2008 at 7:19 PM, gmail <gres.rasmy at gmail.com> wrote:

[stotaro at totarotechn...]

I would avoid IAX and use SIP if at all possible.

Thanks,
Steve Totaro

On Wed, Apr 30, 2008 at 7:23 AM, Bruce Reeves
<breeveslist at eusnetworks.com> wrote:

[tobias.wolf at evision.de]

Hi,
Steve Totaro schrieb:

[greymanvoip at gmail.com]