[asterisk-users] Asterisk 1.8 Paging without DAHDI or MOH Shoutcast with DAHDI

[Faisal Hanif]

Call file are not suitable for you as asterisk process these files in serial
mode (single threaded) and in case of large number of files processing of
last file can be that much delayed that some portion of message may be
already played or the 1st phone may be hanged.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bob Pierce
Sent: Monday, June 27, 2011 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.8 Paging without DAHDI or MOH Shoutcast
with DAHDI

We just finished an upgrade of our Asterisk system to an HA environment on a
pair of servers using Linux-HA. As part of the upgrade, we also moved to
Asterisk version 1.8.4.3

Most things are working quite nicely on the new system. However, I'm having
trouble getting a paging feature to work. In Asterisk 1.4, we simply used
the Page() application like this:
3400,n,Page(SIP/3011&SIP/3021&SIP/3110&SIP/3120&SIP/3121&SIP/3122&SIP/3124&S
IP/3125&SIP/3126&SIP/3127&SIP/3221&SIP/3222&SIP/3223&SIP/3250&SIP/3261&SIP/3
262&SIP/3310&SIP/3311&SIP/3324&SIP/3329&SIP/3331&SIP/3332&SIP/3350&SIP/3455&
SIP/3457)

However, the Page() application seems to rely on the Meetme() application
which also relies on the DAHDI channel driver for mixing of the audio
streams. I have tried using the DAHDI channel driver on this system, but
that seems to make the Music On Hold application use the DAHDI timing module
instead of the pthread module. With the DAHDI timing module, Music On Hold
does not playback Shoutcast streams which is also a requirement for this
system.

As an alternate solution, we have tried implementing a workaround which
simply uses a set of .call files to dial each phone. Those phones then
auto-answer the call and are placed into a conference bridge on mute using
the ConfBridge application. At this point, the initiating caller speaks the
announcement and the phones automatically hangup after about 10 second. This
worked perfectly in our small scale tests. However, when we ramped this up
to the 25 phones that are required and tested it this morning, somehow this
caused the Asterisk service to restart. I suspect that processing the 25
call files and placing them into the conference all at the same time somehow
made the system crash and it immediately started up again.

Here's the relevant dialplan:
exten => 3400,1,Answer
exten => 3400,n,playback(beep)
exten => 3400,n,system(cp /etc/asterisk/testPage/*.call
/var/spool/asterisk/outgoing_staging/)
exten => 3400,n,system(mv /var/spool/asterisk/outgoing_staging/*.call
/var/spool/asterisk/outgoing/)
exten => 3400,n,ConfBridge(testPage,1)
exten => 3400,n,hangup

[testPage]
exten => s,1,Answer
exten => s,n,playback(beep)
exten => s,n,Set(TIMEOUT(absolute)=10)
exten => s,n,ConfBridge(testPage,m)
exten => s,n,hangup
exten => _XXXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten =>
_XXXX,n,Dial(SIP/${EXTEN}) exten => _XXXX,n,Hangup()

and here's a sample call file:
channel: Local/3011 at testPage
callerid: Page
context: testPage
extension: s
priority: 1
archive: no
waittime: 120

Does anyone have insight into how we could accomplish this paging feature or
of anything that we may have missed?

I suspect we could get this all to work with the original Page() application
if there was a way to force MusicOnHold to use the pthread timing module
instead of the Dahdi timing module. Is that configurable somewhere?

Thanks for your help,
Bob

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