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

covici at ccs.covici.com covici at ccs.covici.com
Wed Mar 31 03:29:37 CDT 2010


Jonathan Addleman <jono at redowl.ca> wrote:

> 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.

What is the significance of /dev/fd/3 where does it come from?


-- 
Your life is like a penny.  You're going to lose it.  The question is:
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         John Covici
         covici at ccs.covici.com



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