Hi all,

im my current test-environment, I want asterisk to play a wav-file as a
response to a g.726-32 call

extensions.conf
exten => 2000,1,Answer()
exten => 2000,n,Playback(some_sentence)
exten => 2000,n,Hangup()

When I play the wave-File with mplayer, I am getting the following
file-information:

Audio file file format detected.
==========================================================================
Forced audio codec: mad
Opening audio decoder: [pcm] Uncompressed PCM audio decoder
AUDIO: 8000 Hz, 1 ch, s16le, 128.0 kbit/100.00% (ratio: 16000->16000)
Selected audio codec: [pcm] afm: pcm (Uncompressed PCM)
==========================================================================
AO: [alsa] 48000Hz 1ch s16le (2 bytes per sample)

Now, when activating the G.711 alaw in the calling client, everything is
working fine, I can create the call to the * and I get the sentence
played correctly.

But when I create a SIP-call to the *PBX, where only G.726-32 is allowed
as RTP-codec on the caller, the call is also received, but I only get
random noise.

Well, as I had no clue, what to do, I tried to convert the wav-file to
g726-32 explicitely with the asterisk-"convert" command:
file convert /usr/share/asterisk/sounds/some_sentence.wav /usr/share/
asterisk/sounds/some_sentence.g726-32

As a result, I got the following error:
Failed to convert /usr/share/asterisk/sounds/some_sentence.wav to
/usr/share/asterisk/sounds/some_sentence.g726-32!
[Oct 19 16:13:54] WARNING[8846]: format_g726.c:166 g726_write: Invalid
data length 69, should be multiple of 40
[Oct 19 16:13:54] WARNING[8846]: file.c:209 ast_writestream: Translated
frame write failed
I do not know, if this error is related to the problem described above.

Any help is welcome,

br
Michael

PS: Asterisk 1.4.17~dfsg-2ubuntu1.1 built by buildd @ vernadsky on a
i686 running Linux on 2009-09-25 19:24:25 UTC on Ubuntu Hardy

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