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Auto-Answer Conference Cisco /9xx

Taken directly from a post by John Todd:

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

Improved extension.conf (evolved with new Asterisk features):

extensions.conf without the need for a timeout:

[conference]
exten => 5555,1,AGI(callall)
exten => 5555,2,MeetMe(5555,dtqp) ; press # to exit the conference
exten => 5555,3,MeetMeAdmin(5555,K) ; kick all users out
exten => 5555,4,Hangup
exten => h,1,Hangup
;
[add-to-conference]
exten => start,1,MeetMe(5555,dmqp)
exten => h,1,Hangup


 

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