Podcast playback with call automation

Asterisk Asterisk SIP
1

Hi,

I would like to achieve the following.

  1. Download a short podcast from an URL.
  2. Optionally convert it to some more interchangeable format, like WAV.
  3. Call my cell phone and play the WAV file once.
  4. Wait a bunch of seconds for DTMF input (if you press number 1, it should replay the podcast).

I can do 1 with wget, 2 with ffmpeg but what should I do to achieve 3 and 4?

Can Asterisk do this? Can it ask like a SIP client and dial via a VOIP provider (like VOIPCheap)?

Thanks in advance

2
  1. Use a call file or AMI originate, with the Extension option.

That extension, optionally Waits, then uses Playback.

  1. It then uses WaitExten to direct it back to the Playback or to hang up, or it uses Read and GotoIf, to do the same.

Do you have the necessary copyright and performing rights licences to do this?

1 Like
3

Thank you. Could you please give me more detailed info or some good tutorials online that could help me write a call file and gives a shot about using/configuring extensions? It would be really helpful. To be honest, I’ve never configured Asterisk before.

I don’t care about copyright. It’s gonna be only for my own personal use.

4

The typical use for call files is hotel wake-up calls so it’s probably your easiest bet. There’s the asterisk docs here and there’s a chapter about call files in asterisk the definitive guide if you need more guidance.

You just need to write a script that fetches the podcast and creates a call file, passing the name of the file to play with Setvar. But it sounds like you need to get asterisk set up and making calls first. :slight_smile:

5

Great. Thanks.

I would like to make my calls through an external VoIP provider (VOIPCheap) with an SIP address, username and password. So I want Asterisk to log into that realm and perform these calls from there.

What about the DTMF functionality part? Can call files describe how Asterisk should react to DTMF codes sent back by the callee?

6

No, A call file just causes asterisk to orginate a call.

The dialplan you write for the call file to direct your call to will handle the actual playback of the audio and handling of DTMF.

The ControlPlayback application will let you play a file with fastforward and rewind functionality.

https://wiki.asterisk.org/wiki/display/AST/Application_ControlPlayback

7

The documentation is too much for me, the tutorials on voip-info.org are contradictional. I’m giving up without any success.

Please tell me what to write into modules.conf / sip.conf / manager.conf and how to load only the necessary modules.

8

Are you hiring people? Probably you will get more chances in a site like freelancer.

9

I find with modules.conf it’s easier to not load what I don’t need with noload commands but if you want to start with a basic set of modules I’d suggest looking in your configs/basic-pbx directory, there is a modules.conf there that loads just enough for the basic-pbx demo.

You won’t need app_queue, app_voicemail. app_directory , codecs other than ulaw/alaw (depending on your location)

You shouldn’t need manager.conf at all for what you want to achieve.