[asterisk-users] Live Audio Streaming- From Aux interface-Online resource

[Jonathan Addleman]

nik600 wrote:
> I was trying to record a call usng Mixmonitor and then convert it
> using ffmpeg but the recording file is continuosly growing and ffmpeg
> ends the conversion before of the call completion.

Here's my quick and easy eagi script:

#!/bin/sh
cat /dev/fd/3 | sox -t raw -r 8000 -w -s -c 1 - -t raw -r 44100 - vol 2|
ffmpeg -f s16le -ar 44100 -ac 1 -i - -ab 32k -f mp3 - | ezstream -c
/var/lib/asterisk/ices/stream.mp3.xml

It just dumps the audio through sox, to increase the volume a bit, and
convert the sample rate, then ffmpeg to encode the mp3, and then
ezstream to send it to an icecast server. I could probably skip the sox
step, and get ffmpeg to do those adjustments on its own, but for now, I
know sox's command line better, so I used that. :)

The dialplan is as simple as
exten => meetme,n,MeetMe(confname,1qd)

put all the members of the conversation in there,

exten => mp3stream,n,EAGI(mp3stream.sh)

and then put this in as well to start recording.


-- 
Jon-o Addleman - http://www.redowl.ca