asterisk users - Nov 2003 - SIP Intercom & Paging (was Overhead Paging)

I wasn't thinking of using the conference system as the basis.  I was
thinking more along the lines of:

1)  Setup a second extension on the Cisco phone named "INTERCOM"
enabled for auto-answer
2)  Create a call group on asterisk to dial that "INTERCOM" extension
on every phone that will participate
3)  Add a feature code that would dial the intercom extension and connect to all
phones in the group

This model could also be used for the paging feature since the
"INTERCOM" extension has already been setup.





-----Original Message-----
From: Chris Albertson [mailto:chrisalbertson90278@yahoo.com]
Sent: Friday, November 14, 2003 11:54 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Overhead Paging



I'd hate to see conference bridging use for paging.  A lot of 
wasted CPU and bandwidth.  Could you "multicast" the UDP packets?

We assume you don't need to page across multiple Asterisk servers
but if you did the software wuld need to be smart enough to
"know" which groups of extensions could be in a multicast and
whci need to be bridged.  Basically check to see if the SIP phone
are on the same subnet.


--- DUSTIN WILDES <dwildes@pabbankshares.com>
wrote:
> I feel this needs to be a separate application in Asterisk, like
> app_sipintercom
> The application would connect to all available auto-answer SIP
> phones, play a short frequency tone for the intercom alert, only
> allow one-way streaming to the phones, then disconnect all phones
> whenever the originator hangs up.
>  
> Same is true for a paging application, app_sippage
> The application should work the same as intercom, but allow two-way
> audio streaming.
>  
> I was starting the design of these two applications unless anyone
> else has a better idea or has already begun work?
> Feedback welcome....