[Asterisk-Users] AGI or somthing else??

[WipeOut .]

The situation I was trying to get to was that the Asterisk SIP
authentication would be based soley on IP address and that my
other app would provide the user validation..

That way I could map an extension to a user and it would not
matter which PC they logged into their calls would be routed
to the right place.. 

In thinking about it, it would not work anyway seeing as the
conf files are all read into the system on startup so a change in
the sip.conf would not take effect until the system was reloaded..

Which brings me back to the original problem of allowing muliple
users on the same workstation without having to teach them how to
reconfigure the IP phone to recieve calls on their own
extension??

----- Original Message -----
From: Steven Critchfield <critch at basesys.com>
Date: Sun, 23 Mar 2003 10:09:36 -0600 
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] AGI or somthing else??

> On Sun, 2003-03-23 at 08:03, WipeOut . wrote:
> > Hi,
> > 
> > I know that AGI can be used to determine variables from external
> > sources, but AGI can only be used from the extensions.conf file.
> > 
> > Is there anyway to retrieve external information for the sip.conf
> > file??
> > 
> > Here is why..
> > 
> > I will have users on remote networks with dynamic IP addys..
> > 
> > I have an application that can log the IP address of the person
> > that sucessfully logs into it..
> > 
> > So I want to tie the app system together with the PBX in that
> > by authenticating to the app a user will be granted access to
> > the PBX using a SIP phone based on the IP address, so users don't
> > have to remember another password.
> 
> 2 options here.
> Easiest: The sip software/device should only need to be configured once
> no matter what IP address it is on and will login itself. So no need to
> remember new password.
> 
> Harder but not really that hard: Lock down your asterisk ports with
> linux firewall capabilities and when your app logs the user in, it
> applies a rule allowing that user into the aterisk machine. Of course if
> you plan to route calls to the user, they need to have at least a
> username configured to the sip device so it can identify the user. 
> -- 
> Steven Critchfield <critch at basesys.com>
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

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