As the title attempts to graciously explain the scenario but is grievously click-baity.
How, can I setup the hardware (server), with a soundcard, to output paging/intercom thru a soundcard when an enduser dials the paging extension to make an announcement.
The output of the soundcard will go to the buildings’ speaker, and will be near the amplifier.
Still new at all this. I learned many acronyms, still forget. I have seen flow-diagrams of SIP, but do not remember. Speak to me as a baby/noob.
TO clarify, I don’t know if paging is to be enabled, or if automatically there (if so I need to boot up the server to check it out in a bit); if the right way to reference the paging digits is by using extension number or something else.
I’m decent with Linux. I would say my understanding of Linux is greater than asterisks.
Will I have to have a softphone on the system to auto-answer and than output? I would rather it be internally handled by Asterisks.
Buy a Snom PA1 or CyberData 011233 SIP Paging Adapter.
If you like to deal with non-standard things, Asterisk does contain a channel driver for this, chan_oss. There are pretty much no examples out there for this beyond the default config samples on the Asterisk Wiki. You can search up old Trixbox examples.
I’m running FreePBX 15 / Asterisk 16. Where can I find the chan_oss.conf file?
Also, on the topic of buying a paging adapter. Will the Snom PA1 line-out eliminate the built in amplifier? I see it has a line-out, and indefinite we want to use the buildings’ Amplifier, with ~16 speakers wired to it.
All Asterisk configuration files go in /etc/asterisk, and I don’t believe that FreePBX overrides that.
Please note that no serious work has been done on these drivers for over a decade, and I would consider them as proof of concept, more than production, tools.
yes you will need a conf file and yes you can set your console extension to autoanswer if you want that behavior , all will be immediately obvious after 5 minutes with our friend google knowing about the ‘asterisk console channel’ there is an example included in the asterisk source code.
I know everyone is trying to get you where you want to go, but I tried this and it was harder to get working reliably that the Cisco 79xx phones.
I literally played with this on and off for almost two years before I decided it was a cool idea but entirely impractical for anything but the kludgiest setups. There are lots of reasons, including the mike-key which doesn’t exist, the VOX that you might have on your amp that’s going to drop the first few seconds of your announcements, and the entire db-level/impedance matching dance you’re going to have to do.
Please learn from my mistakes and go for a solution designed to work. The SIP enunciators from several different places (Anixter makes a good one) that let you connect to all sorts of devices would be a better solution and save you a lot of time and probably money. Burning out the primary amp on your PA system is never cheap…
And yet a a large bunch of Raspberians have been using their 3.5mm audio jack without a problem for many years. If you can aplay and arecord from your sound card, you can have a ‘console’ extention defined in a simple /etc/asterisk/console.conf
[default]
input_device = default ; When configuring an input device and output device,
output_device = default ; use the name that you see when you run the "console
; list available" CLI command. If you say "default", the
; system default input and output devices will be used.
autoanswer = yes
context = ext-local
extension = 999
callerid = "Console me" <(888) 420-1234>
language = en
;overridecontext = no
;mohinterpret = default
active = yes ; This option should only be set for one console.
; It means that it is the active console to be
; used from the Asterisk CLI.
Reload the dialplan, or if more comfortable with it, reboot the machine, pick up a phone and dial 999
Matching the output of your sound card which is often a low ohm earphone/electromagenetic speaker driver to the input of your PA system which is often a choice of 10kohm RCA jack unbalanced or 600/300 ohm balanced XLR s a different problem but well documented
I came across an interesting setup a few months ago. I needed to temporarily connect an existing PA system with our voip phones. Didn’t have time, budget, or need for long term solution to order dedicated hardware. Note, I would recommend doing it the “right way”, but for the sake of exploring all options, here’s what I came across. Unfortunately I don’t have details as to what is what, just a picture with some dusty wires. Attached to the PA system there was a block that had some phone wires screwed down on it. I took a voip phone, set it to auto-answer and to use the headset. I then stripped one end off of a rj12 (I think) and replaced the red and green wires on the block shown on the left. Plugged the rj12 jack into the headset port on the phone, and from there, whenever the phone rang and auto-answered or when the intercom function was used, it sent the audio through the headset wire, went through the block, and then fed into the PA system. Sorry I don’t know exactly what everything is, but just wanted to share this weird Frankenstein setup I ran into last year.
