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&SIP/3125&SIP/3126&SIP/3127&SIP/3221&SIP/3222&SIP/3223&SIP/3250&SIP/3261&SIP/3262&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@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

Hi

Since switching from 1.6.x to 1.8.4 I have noticed the following

1. When you do a ‘core show channel ‘ the resulting
information only shows data for “Frames In” , “Frames out” is always
0.

2. The rtptimeout option in the sip.conf no longer seems to work. I
have this set to 60 seconds but have had channels which have not
timeout when the rtp stops. If I subsequently do a “channel request
hangup” then the CLI reports that there was a rtptimeout but will be
hugely over the set amount. For instance on requesting the hangup on
one such channel today it reported rtp timeout for 18000 seconds.

Anybody got any ideas how to get 1.8.4 rtptimeout correctly?

Regards

Jon

Jon Farmer
Tel 07795 118140

Hey – Is there an asterisk interface to gstreamer?
I have not found anything…

Perhaps using a web cam and v4l to get some video going on a PC with
asterisk running.
Call come into extensions.conf and take the video and send it to
xvimagesink and audio to alsa
with outgoing video coming from v4l.

Is anything like that out there?

Jerry

That’s not the password.

I switched it to that in the config file for realism.

I always give some honey out to those who have a sugar tooth.

Any ideas on the fix?

Hi all
How can I change the festival application voice in asterisk from mailvoice
to femailvoice.

Best Regards,

Mahesh Katta
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