[Asterisk-Users] All Page ??

[Robert Murray]

I did something like this I had to work on the perl to get it working
myself.
I found if I put a wait and a beep in the dial plain for the calling user
then
they would get beep 3 or 4 seconds later this give the calls time to set up.

Worked great for me.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Jeremy
Gault
Sent: Monday, August 22, 2005 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] All Page ??


I have (sort of) made this work in our environment.

There was an AGI script (Google for polycom allcall.agi) written in Perl
that would implement a hack to do this.  Basically, you can set the
Alert-Info SIP header and cause the Polycom phones to auto-answer.  The
general idea behind allcall.agi (and the associated dialplan that's
documented with it) is that you dial a paging extension, allcall.agi
creates callfiles for each extension (with Alert-Info set for
auto-answer), puts them in /var/spool/asterusk/outgoing and causes
Asterisk to call each extension.  When those extensions answer, it puts
them into a MeetMe conference (in listen-only mode.)  Then the caller
goes into the same conference in talk-only mode.  There's a hard time
limit after which the call is terminated.

I had some issues with the Perl script, so I wrote my own version in
PHP.  I also made some modifications to the conference flags (making the
person paging a marked user, and having all the pagees be disconnected
when the marked user hangs up, which takes away the time limit on pages
and even the need for it.)  The PHP script itself worked well.

However, I ran into another issue: I had an extra call file that would
basically dial Local/pagebeep at internal-special (I setup an
internal-special context, with a pagebeep extension that would wait a
second, play a beep, and hang up.)  The idea was to tell people to wait
until after the beep (which would give the phones all time to sync up)
before speaking.

That worked great when paging from an analog extension, but when doing
it from a SIP extension, Asterisk complained about codecs and formats
not matching.  I tried it with MeetMe2 and it worked, but MeetMe2 does
not support marked users and such (which I want to have.)

I could dispose of the beep and just tell people to wait a few seconds,
and that would work, and would probably work for you.

          Jeremy

--
Jeremy Gault, KD4NED    <jgault at winworld.cc>
Network Administrator, WinWorld Corporation
Member: Bradley County ACS/RACES/SkyWarn
voice: +1.423.473.8084  fax: +1.423.472.9465
fwd: 461771             msn msgr: jgault at winworld.cc

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