asterisk users - Aug 2012 - Japanese voicefiles

Hi Guys,

I've a few questions around languages.... I'm on 1.4.18 (old yes I know,
but upgradings not an option just yet).

I've downloaded the gsm Japanese files from
ftp://ftp.voip-info.jp/asterisk/sounds/ and put them in place

I've found that when I switch to jp, and play some of my own voicefiles in
Background and Playback, that it chooses the /var/lib/asterisk/sounds/jp folder
files and plays them, but, voicemail doesn't seem to do this, instead it
picks the English files (although the debug output says its using 'jp').

I've seen references to a patch for this, but any idea where the patch is ?

Secondly,

I'm trying to open .gsm files in Audacity (in particular these japanese
ones, so I can confirm they are Japanese), but I just can't get the audio
format right (audacity 2)

Open RAW: Encoding ?  Byte Order ? Channels: mono, Sample rate 8000hz.  I've
set my Pref Quality defaults to 8000hz and 16-bit, but I think that's only
for recording.  Anyone know the correct setting?   I've been able to play
them in Quicktime so I think they're ok, I just want to see them in
Audacity.



Thanks,

Adrian

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On 23/8/12 5:26 pm, Adrian Marsh wrote:
> I've a few questions around languages.... I'm on 1.4.18 (old yes I
know, but upgradings not an option just yet).
> I've downloaded the gsm Japanese files from
ftp://ftp.voip-info.jp/asterisk/sounds/ and put them in place
> I've found that when I switch to jp, and play some of my own voicefiles
in Background and Playback, that it chooses the /var/lib/asterisk/sounds/jp
folder files and plays them, but, voicemail doesn't seem to do this, instead
it picks the English files (although the debug output says its using
'jp').
We use a similar method to play British English sounds - they're in 
/var/lib/asterisk/sounds/britishfemale - and voicemail seems to pick 
them up correctly. Have you made sure to specify language = jp in the 
relevant places? You need to do it in whichever module is originating 
the call to Voicemail - so if it's a SIP client, you'd probably do it in
sip.conf, but if it's an incoming call, it's probably easier to do it in
extensions.conf.

FWIW, this is also using an old version - 1.4.21, so unless something's 
changed between .18 and .21, it should work with your setup.


Kind regards,

Chris
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