Page sound card and paging group
i have a sound card set up so that i can page over our PA system using *51. i also have a couple paging groups set up that will page through the intercom on the phones (group 150 & 151). i am wondering how i can page to both the PA system and a paging group at the same time.
I've been following the instructions here: http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound... as best I can and don't seem to be able to get it working either. I started with step 5, as suggested by the author of the article. When I dial the custom extension for paging that is defined extensions_custom.conf, I just get an immediate dial tone. I plan on working on this some more over the weekend, so I'll let you know if I have any success, but I'd certainly like to hear from others.
I'm using a SoundBlaster Live! card that CentOS picked up and configured with no problem. I'm then running a 1/8" plug from the sound card to our amplifier, which is a QSC MX700.
Andrew
i followed those same instructions and have gotten paging through my sound card to work, but we have areas where there is no PA speaker, just phones. i want the paging to go through the sound card (*51) and the paging group (150) at the same time. i cant see how i can add the sound card that i access through *51 to a paging group.
We actually have a similar setup to what you're asking about. We have security phones on our external doors. We can call to the phone from any extension in the building. To do this, I used two pieces of equipment from Viking Electronics. For the phones, I used the Viking E-10A (http://www.vikingelectronics.com/products/view_product.php?pid=42...) and then wired them to a Viking C-2000A, which then connects to an FXO port on the trixbox machine. The reason we went with the C-2000A is b/c we could control phones at 4 different doors and also "buzz" people in using door strikes. The phone itself is pretty cool too. It has caller id and call waiting. Caller ID is really helpful b/c you can use time conditions within trixbox to route the correct phone call to the proper extension.
I don't work for Viking, just have been happy using them in the past.
it seems this thread is going sideways a little and getting away from my original question of how i can page the sound card (*51) and a paging group (150) at the same time, which i still havent gotten any answer to. im looking more into the asterisk config files side of things and think that perhaps i can create a custom extension for this, but so far i havent gotten that figured out. any help would be greatly appreciated.
friesen
In answer to you question it will be no simply because the device that you are paging is in effect "answering" the call and so no you can't have 2 diffrent devices to answer together. BUT what you could do is set up a spare phone set it to auto answer and connect the overhead paging input to the phone headphone jack output. That should work.
Simon
Go to Free pbx extensions add new extension
chose other custom device name page for example it and put a extension number 444 for example.
This device uses custom technology.
dial console/dsp
next step
go in /etc/asterisk and create file called oss.conf paste the following:
;
; Open Sound System Console Driver Configuration File
;
[general]
;
; Automatically answer incoming calls on the console? Choose yes if
; for example you want to use this as an intercom.
;
autoanswer=yes
;
; Default context (is overridden with @context syntax)
;
context=from-internal
;
; Set overridecontext to yes if you want the context specified above
; to override what someone places on the command line.
;
overridecontext=yes
;
; Default extension to call
;
extension=s
;
; Default language
;
language=en
;
; Silence supression can be enabled when sound is over a certain threshold.
; The value for the threshold should probably be between 500 and 2000 or so,
; but your mileage may vary. Use the echo test to evaluate the best setting.
;silencesuppression = yes
;silencethreshold = 1000
;
; On half-duplex cards, the driver attempts to switch back and forth between
; read and write modes. Unfortunately, this fails sometimes on older hardware.
; To prevent the driver from switching (ie. only play files on your speakers),
; then set the playbackonly option to yes. Default is no. Note this option has
; no effect on full-duplex cards.
playbackonly=yes
;
next step
edit modules.conf
[modules]
autoload=yes
;
; If you want, load the GTK console right away.
; Don't load the KDE console since
; it's not as sophisticated right now.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss. Don't load it.
;
noload => app_intercom.so
;
; DON'T load the chan_modem.so, as they are obsolete in * 1.2
noload => chan_modem.so
noload => chan_modem_aopen.so
noload => chan_modem_bestdata.so
noload => chan_modem_i4l.so
; Trunkisavail is a broken module supplied by Trixbox
noload => app_trunkisavail.so
; Ensure that format_* modules are loaded before res_musiconhold
;load => format_ogg_vorbis.so
load => format_wav.so
load => format_pcm.so
load => format_au.so
; This isn't part of 'asterisk' iteslf, it's part of asterisk-addons. If this isn't
; installed, asterisk will fail to start. But it does need to go here for native MOH
; to work using mp3's.
load => format_mp3.so
load => res_musiconhold.so
;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
load => chan_oss.so
;
; Module names listed in "global" section will have symbols globally
; exported to modules loaded after them.
;
[global]
next step
in the console
chmod 0777 oss.conf
chown asterisk:asterisk oss.conf
amportal restart
alsamixer
unmute everything
next step
go in Paging and intercom
you can select extension 444 along with the other extensions
Hope this work for you
Spase



Member Since:
2007-09-04