[asterisk-users] Asterisk does not translate from wav to alaw
BryantZ at zktech.com
Sat Aug 28 10:35:59 CDT 2010
On Sat, Aug 28, 2010 at 3:22 AM, Jonas Kellens <jonas.kellens at telenet.be>
I have a file to be played in wav-format.
I thought Asterisk would automatically take the wav-file and translate it
to the codec used, but I see this :
[Aug 28 11:16:29] WARNING: file.c:664 ast_openstream_full: File
/var/lib/asterisk/sounds/vprompts/zip-code.wav does not exist in any
[Aug 28 11:16:29] WARNING: file.c:991 ast_streamfile: Unable to open
/var/lib/asterisk/sounds/vprompts/zip-code.wav (format 0x8 (alaw)): No such
file or directory
[Aug 28 11:16:29] WARNING: pbx.c:5752 pbx_builtin_background:
ast_streamfile failed on SIP/test1-0000000f for
Am I missing a module to translate from wav to alaw/gsm/g726/... ??
My guess is that your .wav file is NOT 8khz. The system doesn't accept
anything but wav files at 8khz. Use
sox to downsample to 8khz (and 1 chan), and the problems should go away.
While you are at it, you could use sox to convert
to the target format in a single operation.
The scripts that Digium uses to take Allison's voice prompts (at 48khz) to
the different formats, convert things to slin (raw) as a central
format, but in my experience, the fewer steps the better. But I doubt that
anyone could detect the difference in the end result...
Here's what I do with CD-qual sounds to turn them into the common Asterisk
Assume $i is the name of the .wav file you want to convert:
x=`basename $i .wav`
sox -v 0.7 $i -r 16000 -c 1 -t sw $x.sln16
sox -v 0.7 $i -r 8000 -c 1 -t sw $x.raw
sox -r 8000 -c 1 -t sw $x.raw -t gsm $x.gsm ## OR ### sox -v 0.7
$i -r 8000 -t gsm $x.gsm
sox -r 8000 -c 1 -t sw $x.raw -t ul $x.ulaw ## OR ### sox -v
0.7 $i -r 8000 -t ul $x.ulaw
sox -r 8000 -c 1 -t sw $x.raw -t al $x.alaw ## OR ### sox -v
0.7 $i -r 8000 -t wav $x.wav
sudo asterisk -rx "file convert $y/$i $y/$x.g722"
I'm ignoring the siren & g729 formats; use asterisk for those in like
format, depending on your asterisk version and codecs.
Allison normalizes the volume of sounds she distributes; use the -v 0.7 to
bring the volume down a bit to
the standard, and your sounds won't stick out against rest of Allison's
existing recordings in Asterisk.
Digium uses a filter program to 'heighten' the sounds a little; That's the
main reason, I think, that they
use the .raw format as an in-between. I've been skipping this step, as it
doesn't seem critical, in which
case the direct conversion is probably preferable.
I suggest, that if you are converting sounds for Asterisk's sake, that you
convert to all the possible
formats. Disk space is cheap, and you'll squeeze a little extra performance
out of Asterisk by allowing
it to pick the 'best' format. Dahdi type interfaces would prefer the
ulaw/alaw formats; High-def phones
like Snom (and appropriate Polycoms, etc) could use the g722. Ulaw and gsm
transcodings are cheap,
but no transcoding is cheaper still.
Thanks for sharing I appericate your insight as this is something I run up
against as well.
What about g729 we use this coded a lot what is the best method to
transcode it it?
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