[Asterisk-Users] Is it possible to push the media processing off to a gateway for processing?

[Todd Wallace]

Well, I like the features asterisk gives me such as voicemail and IVR,
Prompts, etc.  I would like to offer an IP Centrex like service, but don't
believe that I can handle very large amounts of users on a box.  The reason
I believe this is that the box would be doing all the media processing/DSP
work on the processor and would be bound by the speed and memory of the box
as to how many simultaneous sessions it could manage.  A gateway has DSP's
which are designed to handle this processing.  I know they are more
expensive, but I could handle large amounts of call volume this way and
still keep the features asterisk offers.  

Another question I also meant to ask was having the ability to read
extensions from a database instead of a .conf file.  I was curious if anyone
has asterisk pulling configs from a database like mysql.

Todd


-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of John Todd
Sent: Friday, January 23, 2004 9:05 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Is it possible to push the media processing
off to a gateway for processing?

>I was wondering if it is possible to have Asterisk push the media
processing
>off to something with DSP's such as a gateway?  That way, asterisk just has
>to handle the call setups and tear downs.
>
>Todd Wallace

You mean, like what SIP does by default?  This is an incomplete 
question.  Please be more specific.  If I have a "gateway", and I 
have SIP calls coming in from desktop SIP UA's (hardphones or 
softphones) then Asterisk can simply re-direct those calls to the 
gateway.

Of course, Asterisk _is_ a gateway, so unless you have specific 
reasons for doing so, it would make more sense to use Asterisk to 
tackle those jobs with generic, cheap processing horsepower rather 
than expensive, proprietary DSP's.

If you're just getting Asterisk to handle call setups and teardowns, 
why not just use a real SIP proxy for that?  Or do you not know 
enough about your question to understand why I would differentiate 
between the two?  (not being nasty here, just wondering if I need to 
explain more)

JT
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