audioasteriskvoipfreepbx

asterisk - write plugin to "catch" voice stream


I have installed Asterisk and FreePBX on a server. I have set up some extensions for testing and I have configured voicemail for these. I have discovered that voice mail files gets stored in /var/spool/asterisk/voicemail/default/(EXTENSION)/tmp/(TMPFILE).wav and that they are later moved to /var/spool/asterisk/voicemail/default/(EXTENSION)/INBOX/msg####.wav

I want to stream the voice data to another server in real time so that the server can start processing of the data. That is I want to start streaming the data as soon as a call is received by Asterisk and I do not want to wait for the hang up before sending the sound.

I am looking for how to hook into Asterisk to get hold of the voice data. I have a few ideas so far:

1) Regularly check the /tmp directories for new files and start reading them as soon as they are available. But the file names are random and I also want to find out who is calling for example. Seem like a dead end. Files might also get stuck if I read them while Asterisk tries to move them?

2) Maybe I could write a plugin for Asterisk that can pick up the files in the right step of the process. If so, what do I need to write such a plugin and how can I configure it to get access to the voice data as soon as the voice mail module picks up the phone?


Solution

  • You can use EAGI program which able get sound from channel #3.

    http://www.voip-info.org/wiki/view/Asterisk+EAGI

    You also can do c/c++ app for asterisk which will do streamming(see source code in apps/ folder for examples)