login | register
Wed 01 of Aug, 2007 [09:14 UTC]

voip-info.org

Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Shoutbox
  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.39s
  • Memory usage: 2.21MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 3.82

Asterisk multi-language

Setting up a Multi-Language Asterisk Installation


Many Asterisk Dialplan commands cause Asterisk to play sound files from the /var/lib/asterisk/sounds directory structure. Many of these sounds are recordings of phrases and sentences such as, "Please hold while I try that extension." Asterisk supports the automatic selection of different language editions of these sound files.

When Asterisk looks for a particular sound file, such as transfer.gsm, it will first look in a subdirectory corresponding with the currently selected language. For example, if the currently selected language "de", the Asterisk will first look for the file /var/lib/asterisk/sounds/de/transfer.gsm. If this file or the "de" directory does not exist, then Asterisk will look for the file /var/lib/asterisk/sounds/transfer.gsm, which is usually an English-language default recording.

If an Asterisk command specifies a sound file in a subdirectory, Asterisk looks in that subdirectory for the language subdirectory. For example, the SayDigits command may play the sound file "digits/6". Asterisk will, if the language code is "de", first look for /var/lib/asterisk/sounds/digits/de/6.gsm before falling back to /var/lib/asterisk/sounds/digits/6.gsm.

Setting the Language

The default language for a particular channel is set in that channel's configuration file. Look for a setting like:

   language=en

You may specify any text (up to 20 characters long) as the language code. When playing sounds from the /var/lib/asterisk/sounds directory, Asterisk will look for a subdirectory with the same name as the currently selected language.

You can override the channel's default language using the SetLanguage command in your Dialplan. You may use the ${LANGUAGE} channel variable to discover what the currently selected language code is.

(:exclaim:) The language code specified in the channel configuration files and using the SetLanguage command is quite separate from the country code specified in the indications.conf file used to select country-specific tones (dialtone, ring tone, busy tone etc) by the Playtones command.

Dialplan Commands

The Dialplan Commands that use the multi-language feature of Asterisk include:

Tip

This "language selection" feature does not have to be restricted just to making different language files. You could, for example, choose to have your voice prompts recorded by a male speaker and a female speaker. A channel configuration file might set:

   language=female

and you could override that with a call to SetLanguage(male). If you requested Asterisk to play the 'transfer' sound file, then Asterisk will look in /var/lib/asterisk/sounds/male/transfer.gsm directory, and if not found, revert to the default /var/lib/asterisk/sounds/transfer.gsm.

See also



Asterisk | Configuration
Created by oej, Last modification by hoowa on Mon 05 of Feb, 2007 [07:00 UTC]

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:

Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver