[asterisk-users] how to improve sound file quality?

[Tilghman Lesher]

On Wednesday 03 December 2008 04:04:10 Ronald Wiplinger (Lists) wrote:
> We have recorded wav files with 44k, 22k, 16k, 11k and 8k
>
> Asterisk does not accept these wav files. I used sox input.wav
> output.gsm to get them to work.
> However, the only the 8k file did convert and the quality is poor. How
> can I improve the quality?

Asterisk will only directly process files that are 8000 Hz, single channel.
If your 8k wav files will not work, my suspicion is that you're using
2-channel encoding or another incompatible setting.  By the way, the
only reason that we encode to gsm by default is that we package the sounds
with the tarball and gsm is the smallest format available that is also free
to use.  You're certainly welcome to create your sound files in .wav format
(in fact, if you run 'make menuselect', you can install ALL sound files in wav
format).  Also, if you want to further reduce the load on your Asterisk
server, you can install all formats for codecs that your users' phones use, as
that decreases the amount of transcoding needed to be done at runtime.

-- 
Tilghman