Re: [asterisk-users] Japanese voicefiles
Ok, This is something to do with folder layouts. I have: /var/lib/asterisk/sounds - uk files /var/lib/asterisk/sounds/digits -uk/us digits /var/lib/asterisk/sounds/jp - Japanese files /var/lib/asterisk/sounds/jp/digits - Japanese digits I read the 1.4 notes on : http://www.voip-info.org/wiki/view/Asterisk+multi-language Which says that, in 1.4, by default it'll work as 1.2, which expects: /var/lib/asterisk/sounds - uk files /var/lib/asterisk/sounds/digits -uk/us digits /var/lib/asterisk/sounds/digits/jp - japanese digits But if you put languageprefix=yes in asterisk.conf (and restart I presume), then it would work the 1.4/1.6 way But I've now got languageprefix=yes set in my test setup, and I can only get Japanese digits if I put them in sounds/digits/jp. Odd, but at least I've a solution, sort of... Will just test now in my old 1.4.18 setup with a symbolic link I think -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Japanese voicefiles
Ok... I'm baffled..
I took a copy of my machine and put it in a virtual machine, then upgraded the
VM to 1.4.44 to experiment, and unknowingly let it install the default US GSM
sounds again.
My code runs, but, it still plays the US digits when the debug says the below.
You can see its set to JP, and that its picking normal voiceprompts from JP
I tried deleting the 2.* files from sounds to force it to error (and confirm
which file its playing), but it doesn't error. SAYDIGITS just skips the
numbers it can't find :|
[2012-08-24 11:33:31] VERBOSE[18633] logger.c: -- Executing
[9222@TokyoReception:1] SayDigits("SIP/XXX-005", "9222") in new stack
[2012-08-24 11:33:31] VERBOSE[18633] logger.c: --
Playing 'digits/9' (language 'jp')
[2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing
[9222@TokyoReception:2] NoOp("SIP/XXX-0005", "LoopCounter is 0") in new
stack
[2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing
[9222@TokyoReception:3] GotoIf("SIP/XXX-0005", "0?dialit") in new stack
[2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing
[9222@TokyoReception:4] GotoIf("SIP/XXX-0005", "0?dialit") in new stack
[2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing
[9222@TokyoReception:5] GotoIf("SIP/XXX-0005", "0?dialit") in new stack
[2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing
[9222@TokyoReception:6] GotoIf("SIP/XXX-0005", "0?dialit") in new stack
[2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing
[9222@TokyoReception:7] GotoIf("SIP/XXX-0005", "1?dialit") in new stack
[2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Goto
(TokyoReception,9222,13)
[2012-08-24 11:33:32] VERBOSE[18633] logger.c: -- Executing
[9222@TokyoReception:13] Playback("SIP/XXX-0005", "vm-dialout") in new stack
[2012-08-24 11:33:32] VERBOSE[18633] logger.c: --
Playing 'vm-dialout' (language 'jp')
[2012-08-24 11:33:34] VERBOSE[18633] logger.c: == Spawn extension
(TokyoReception, 9222, 13) exited non-zero on 'SIP/XXX-0005'
Thanks,
Adrian
-Original Message-
From: Adrian Marsh
Sent: 24 August 2012 09:42
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Japanese voicefiles
Hi Chris, Thanks for replying,
I've got it set in the context in extensions.conf:
[TokyoReception]
exten => s,1(TOKYORECEPTION),Answer
exten => s,n,Set(CHANNEL(language)=jp) ; set japanese by default
exten => s,n,SET(LOOP=0)
exten => s,n,SET(LANG=JP)
It could be something fixed between 1.4.18 and 1.4.21. Wish I could find the
bug ID now... Can you confirm you set the language the same way ?
If you've got files in ..sounds/britishfemale, then how are you setting the
sub-folder ? (I thought it would only choose en, fr, jp, etc based on country
codes).
If I put a custom vm-dialout.sln file in sounds/jp, then it does play that
file, so it seems to only affect the sounds/jp/digits folder (a sub-sub folder
with numbers).
Thanks,
Adrian
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Re: [asterisk-users] Japanese voicefiles
Hi Chris, Thanks for replying, I've got it set in the context in extensions.conf: [TokyoReception] exten => s,1(TOKYORECEPTION),Answer exten => s,n,Set(CHANNEL(language)=jp) ; set japanese by default exten => s,n,SET(LOOP=0) exten => s,n,SET(LANG=JP) It could be something fixed between 1.4.18 and 1.4.21. Wish I could find the bug ID now... Can you confirm you set the language the same way ? If you've got files in ..sounds/britishfemale, then how are you setting the sub-folder ? (I thought it would only choose en, fr, jp, etc based on country codes). If I put a custom vm-dialout.sln file in sounds/jp, then it does play that file, so it seems to only affect the sounds/jp/digits folder (a sub-sub folder with numbers). Thanks, Adrian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Japanese voicefiles
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 -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
