[Asterisk-Users] SIP Intercom & Paging (was Overhead Paging)

[DUSTIN WILDES]

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 at yahoo.com]
Sent: Friday, November 14, 2003 11:54 AM
To: asterisk-users at 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 at 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....