[Asterisk-Users] Cisco 7940 Phones as paging system?

[John Todd]

At 10:31 AM -0400 5/8/04, Billy Huddleston wrote:
>That won't work.. That'll DIAL multiple phones/extensions, but will only
>bridge 1 of them when it auto-answers..
>
>What we need is a way to have something like meetme call multiple extensions
>and bridge them to a meetme confrence (all of them muted but the admin of
>course, as it's a one way page) and then we would have a true paging
>system..

OK, I typically would badger people into looking in Google for this, 
but I'll be darned if I can't find this post on Google myself (search 
for "Office-wide paging with Asterisk" or "AGI(callall)" so I'll 
re-post here.  This is a terrible hack.  Someone _please_ make this 
cleaner.

I'm looking at how to add this to the Wiki, but I don't see anything 
that's obviously marked as "start new thread" or similar links.  If 
anyone is feeling ambitious, please add the stuff below.

JT



>Date: Sun, 18 Jan 2004 17:22:11 -0700
>To: asterisk-users-lists.digium.com
>From: John Todd <jtodd at loligo.com>
>Subject: Office-wide paging with Asterisk and Cisco 7960 7940 phones
>
>
>I spoke the other day about my preliminary tests with office-wide 
>paging with Cisco phones using the new SIP 6.1 image which supports 
>auto-answer.  I've got a small and crude recipe for those of you who 
>want to experiment and hopefully create some better and more 
>complete examples than the one I've thrown together below.
>
>Create a new line on each of the Cisco phones, and put the 
>configuration into sip.conf as you normally would.  I figure you 
>have enough clue to create a new line in sip.conf and on your Cisco 
>phones at this point.  Go into settings -> Call Preferences -> Auto 
>Answer (intercom)  and then make the "new" line you've just created 
>as auto-answer.  (I wish there was a way to do this via the 
>configuration file!  Having to set this while sitting in front of 
>the phone is silly and wasteful.)
>
>Now that you have created a valid Asterisk-capable SIP line that 
>auto-answers, here's how you get the paging features to work:
>
>Here's what I have in extensions.conf:
>
>[conference]
>exten => 5555,1,AbsoluteTimeout(21)
>exten => 5555,2,AGI(callall)
>exten => 5555,3,MeetMe(5555,dq)
>exten => 5555,4,Hangup
>
>exten => t,1,Hangup
>exten => T,1,Hangup
>exten => h,1,Hangup
>;
>[add-to-conference]
>exten => start,1,AbsoluteTimeout(20)
>exten => start,2,MeetMe(5555,dmq)
>exten => h,1,Hangup
>exten => t,1,Hangup
>exten => T,1,Hangup
>
>
>Here are the contents of /var/lib/asterisk/agi-bin/callall
>
>#!/bin/sh
>cp /var/lib/asterisk/agi-bin/*conf /var/spool/asterisk/outgoing
>
>
>Make sure to make the script executable.
>
>And then for every extension I have as an auto-answer, I have a file 
>like this in /var/lib/asterisk/agi-bin :
>
>Channel: SIP/2006
>Context: add-to-conference
>Extension: start
>Priority: 1
>CallerID: Office Pager <5555>
>
>
>So, I have three lines that are configured for automatic answering - 
>SIP/2006, SIP/2007, SIP/2008.  I have three files named 2006-conf, 
>2007-conf, 2008-conf in /var/lib/asterisk/agi-bin that get copied 
>into the outgoing call spool directory every time I call extension 
>5555.   These three lines are the auto-answer lines on each of the 
>three phone devices I'm experimenting with.
>
>Now, dial 5555 from any phone and you should have one-way paging. 
>Voila!  People who use the pager may have to get used to waiting 1-2 
>seconds before speaking to allow all the phones to catch up with the 
>audio stream.  All of the phones hang up after 20 seconds, 
>regardless of if the person originating the page has stopped 
>talking.  Change the AbsoluteTimeout values to increase this 
>interval.
>
>If you want a really confusing loud mess, then change the "dmq" 
>options to "dq" and you'll get an N-way conversation going with 
>everyone who has a phone.  Bad.
>
>If you want a really interesting office surveillance tool, change 
>the "dmq" to "dt" and you'll suddenly be listening to all of the 
>extensions in the office, like some kind of mega-snoop tool. Useful 
>for after-hours listening throughout the entire office.
>
>
>Someone should improve my scripts with the following changes:
>  1) AGI should automatically show the caller ID of the person 
>originating the call instead of a generic pager address
>  2) The AGI should take arguments of what extensions to call and 
>then dynamically create the list of files that get copied out to the 
>/var/spool/asterisk/outgoing directory
>
>JT
>