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

[John Baker]

This hack is a tiny bit better:

http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html

John Baker


John Todd wrote:
> 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
>>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
>