Help for Paging via ALSA

Hi to all!!!
I have a problem: I need to configure an extension (ex. 901) in my FreePBX but the extension have to have Input/output on SoundCard (00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03)) on the same server machine. This is needed to put the input/ouput in a VHF Radio (no PTT needed but VOX controlled).
Can anyone help me?

A lot of info can be found on Google. This for example

Asterisk also comes with a console driver


I loaded both chan_alsa and chan_console correctly, and defined the 900 extension in chan_alsa.conf as:
; 900
exten => 900,1,Dial(CONSOLE/ALSA)
exten => 900,n,Hangup()

Now… In the IVR of FreePBX I need to add the 900 option… The “real” extension like my “100” work correctly, but I don’t see options to redirect to 900…

You will need a custom extension for 900 for FreePBX to route it where you set the ‘diak’ appropriately

Can u help me with some steps to do this? If u like email me: [email protected]

Have you got it working from the asterisk cli?

Until you do get that working you can’t really move forward.

Did u mean If I connect to Asterisk CLI (via -r option) ?

rasterisk -x “console dial [email protected]

Yes. From console I can dial all internals.

What I need is:

  • The user from external call the PBX
  • IVR answer (ok this is already done and works great)
  • If the user compose the “900” internal, will be redirect to alsa <- this not work

What does your console.conf file look like?

You will maybe need a FreePBX ‘custom extension’ that dials


And then it should be accessable by FreePBX

The console.conf has this only two lines:

There’s you prob.this is what it needs

But now FreePBX don’t know any extension that point to Console.

Did you actually read it? what context did you use? I would suggest

context = from-internal

What extension did you chose? I would suggest

extension = 900

Use the other parameters for caller ID appropriate,

Ok. Now console answer…
I have a big problem… Sound like a bird chirp… If i speak in the telephone mic, console output is no recognizable… Volumes are ok…
Codec settings? (I hope that asterisk use PCM codec by default).

alsa will need to use the same format as asterisk slin

8kHz,16 bit, mono, signed, raw PCM

I fail to under stand why anybody would use ALSA. You can buy a snom-pa1 for $120. Even a Cisco SPA112 would work.Turn the SAS on, and less then 600ohm output load…both will do it with far less hassle.,

I already have a SPA112 (2xFXS) but I need to use the console to send and receive audio to/from a VHF Radio (the PTT is controlled by a vox).

We did have something like that on our 2way radios, years ago. It connected to the mic and spk to RJ-11. any gateway would work. You press the star to go foo hook off line, Pound to hang up

Try this link.It’s something like that