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Help for Paging via ALSA


(N3 T Group) #1

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?


#2

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


#3

Asterisk also comes with a console driver

/usr/lib/asterisk/modules/chan_console.so


(N3 T Group) #4

I loaded both chan_alsa and chan_console correctly, and defined the 900 extension in chan_alsa.conf as:
autoanswer=yes
; 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…


#5

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


(N3 T Group) #6

Can u help me with some steps to do this? If u like email me: zio.rick@gmail.com


#7

Have you got it working from the asterisk cli?

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


(N3 T Group) #8

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


#9

rasterisk -x “console dial 13235551212@from-internal”


(N3 T Group) #10

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

#11

What does your console.conf file look like?

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

LOCAL/(theasteriskconsolerxtension)@from-internal

And then it should be accessable by FreePBX


(N3 T Group) #12

The console.conf has this only two lines:
[general]
autoanswer=yes


#13

There’s you prob.this is what it needs


(N3 T Group) #14

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


#15

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 etc.as appropriate,


(N3 T Group) #16

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).


#17

alsa will need to use the same format as asterisk slin

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


#18

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.,


(N3 T Group) #19

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).


#20

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
http://ipgate.ae/index.php/solutions/2015-01-20-17-05-54/2-way-radio-over-ip-solution