Sound output to sound card

Yes, you’re right, I’ve already tried ‘default’, ‘hw:0,0’, ‘bcm2835:0,0’ in /etc/asteris/alsa.conf. I don’t know what else could be written to that file .

The asterisk user can use the hardware? check the permission and owner also check if the card is not in use for other app.

This now feels more like a Raspbian or Alsa issue.

It’s a long story how everything started working (my collegue finished this deal) :smile:
So, we go on.

Now the system “IP-phone - RPi” can be used as a loudspeaker.

But our scheme will look as on the picture:
http://s2.postimg.org/lspjmee3t/alarm.jpg

  • there is a main office PBX (also running Asterisk)
  • all IP-phones are connected (registered) to it
  • several RPis with Asterisks on board are connected to main PBX by SIP trunks. Each of them has the same extensions (p.e. 111), after dialing which the sound is outputed from the speaker, connected to RPi’s sound card.

Now the problem.
I understand how to dial to a single RPi, through main PBX (… Dial(SIP/rpi1_trunk/111) or Dial(SIP/rpi2_trunk/111) and so on) and it’s OK.
But we need an ability to dial to all RPis simultaniously (for example to speak the same words to all zones, where RPis are located in the building, like “Fire! Leave the office” :smile: ).

What’s the best tool to do this?
If I will use the construction ‘Dial(SIP/rpi1_trunk/111)&Dial(SIP/rpi2_trunk/111)&Dial(SIP/rpi3_trunk/111)&Dial(SIP/rpi4_trunk/111)’ , I’m sure that only one RPi will speak. The same is about queues. The first answered RPi will output the voice (am I right?), others will not.
I know that the phone queue with the strategy ‘ringall’ behaves in t his way - all phones are ringing, but when any of them is picked up, others stop ringing and of course they do not output the caller’s sound, only the picked phone is used in the conversation.

In fact, all our RPis answer automatically, but still… I’m not sure that &-construction or Queues can be the best solution here.

Should I use Conference ?
If it’s so, each RPi must automatically enter the conference, created on the main office PBX, am I right ?

See the page application.

Also, the application name should only appear once.