[asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?

[Adam Moffett]

Thanks nice script.

But why au files in addition to gsm?

  ----- Original Message ----- 
  From: dave cantera 
  To: Asterisk Users Mailing List - Non-Commercial Discussion ; vincent.delporte at bigfoot.com 
  Sent: Tuesday, January 01, 2008 11:27 AM
  Subject: Re: [asterisk-users] [1.4 + FreeBSD 6.2] Playing WAV PCM file?


  vincent,
  here is a script that I used to convert a single wav file or the entire directory... no file specified on launch, converts all files in the current directory...
  creates a logfile, although trivial... 
  daveC

  #!/bin/sh
  #
  #    convert-all.sh
  #
  #    convert all *.wav files to .gsm .au formats
  #

  if [ "null${1}" == "null" ]
  then
      FILE_LIST=`ls *.wav`
  else
      FILE_LIST=`ls ${1}*.wav`
  fi

  LOG="./log_convert.log"
  echo "======================================================= " >>${LOG}
  echo "        started at `date`         " >>${LOG}

  echo " Removing all current .gsm files..."
  rm -f *.gsm

  for FNAME in ${FILE_LIST}
  do
      echo "----   -------   ----- "
      echo "----   " >>${LOG}
      echo " Processing ${FNAME}... "
      echo " Processing ${FNAME}... " >>${LOG}
      BASEFNAME=`echo ${FNAME} | awk '{print substr($0,1,length($0)-4)}'`

      echo "         making ${BASEFNAME}.gsm... "
      echo "         making ${BASEFNAME}.gsm... " >>${LOG}
      #sox -q -V -c 1  ${FNAME} -r 8000 -c 1 -w ${BASEFNAME}.gsm resample -ql  2>>${LOG}
      sox -q -V ${FNAME} -r 8000 -c 1 ${BASEFNAME}.gsm resample -ql  2>>${LOG}
      echo "----   " >>${LOG}
      echo "         making ${BASEFNAME}.au... "
      echo "         making ${BASEFNAME}.au... " >>${LOG}
      sox -q -V ${FNAME} -t au -r 8000 -c 1 -w ${BASEFNAME}.au resample -ql 2>>${LOG} 
  done






  Vincent wrote: 
Hello

Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0
and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd
like to play PCM WAV files instead of eg. GSM. Per...

www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk

... I recorded a sample of my voice using XP's Sound Recorder, then
ran the following :

sox test_wav.wav -r 8000 -c 1 -s -w test_wav_out.wav resample -ql

But it seems like I'm missing the codec or something:

===========
  -- Executing [s at default:2] Playback("SIP/2000-0871d000",
"/usr/local/lib/asterisk/test_wav_out.wav") in new stack

WARNING[37390]: file.c:563 ast_openstream_full: File
/usr/local/lib/asterisk/test_wav_out.wav does not exist in any format

WARNING[37390]: file.c:866 ast_streamfile: Unable to open
/usr/local/lib/asterisk/test_wav_out.wav (format 0x4 (ulaw)): No such
file or directory
===========

Here's what "core show file formats" says:
===========
Format     Name       Extensions
gsm        wav49      WAV|wav49
slin       wav        wav
adpcm      vox        vox
slin       sln        sln|raw
g722       g722       g722
ulaw       au         au
alaw       alaw       alaw|al
ulaw       pcm        pcm|ulaw|ul|mu
ilbc       iLBC       ilbc
h264       h264       h264
h263       h263       h263
gsm        gsm        gsm
g729       g729       g729
g726       g726-16    g726-16
g726       g726-24    g726-24
g726       g726-32    g726-32
g726       g726-40    g726-40
g723       g723sf     g723|g723sf
18 file formats registered.
===========

Am I missing something in the configuration files, or maybe I'm
missing some module?

Thank you.


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