similar to: Asterisk i18n

I am trying to manage the country names with i18n. I have a countries table with the countries extension codes (us., co., ve., .bo. etc...) and the yml files with their complete names: en: countries: ar: Argentina co: Colombia . I am trying to make a form with a select option for country and came up with this: <%= f.select("country_id", Country.all.collect {|p| [ t

Hi all, I have a locale en.yml file like this: https://gist.github.com/1707858 If you launch a rails console and ask for: irb(main):003:0> I18n.t("country.ES") => "Spain" Cool. Expected response, but what if you do: irb(main):004:0> I18n.t("country.NO") => "translation missing: en.country.NO" Despite it exists (check it on the paste). I

Hi I''ve tried this on a virgin Rails 3.2.2 app. If you put the translations for country names in the locale file using the 2 letter ISO code, it looks something like this: en: countries: DE: "Germany" NO: "Norway" US: "USA" This works fine - but for Norway :-) I18n.t :''countries.DE'' # => "Germany" I18n.t

Yes it is. I'm remote at the moment so I can't send you the code but google for mobile remote receiver and you'll find what you are looking for. Lots of people do it so they don't have calls to cell phones picked up by voicemail. Cheers dean -----Original Message----- From: Pete Kay <petedao at gmail.com> Sent: Monday, March 31, 2008 2:27 PM To: Asterisk Users Mailing

Hi, I am havind deep trouble with a bunch of our newly arrived Optiplex 330 as it can't run Centos 5.2 property. The installation works fine, but when it boots up, it can't be connected to the network. I am getting error saying " link is not ready" when doing system-config-network. I check lspci and it can detect the network controller no problem. The light next to the cable

Hi, Does OCFS require NAS hardware to run or does normal PC hard disk work? Thanks, Pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://oss.oracle.com/pipermail/ocfs2-users/attachments/20081219/1bc9110b/attachment.html

Hi, I am running a VOIP application on Centos 5.3. It is a 16 core box with 12 G of mem and all what it does is passing packets. What happens is that at around 2K channels running g711 ( 64k) codec, all eth0 is used up and no more traffic can go through. I have checked google and it talked about interrupt scheduler. does anyone know how to configure the kernel to allow it to use all CPSs for

Dear all, I am having a very strange problem with VoicemailMain. When using this application to record unavail, greet, and busy, I an see the corresponding file gets created in the <..>/default/<SIP #> directory. When pressing "1" to confirm the recorded message, the *.wav file gets deleted from the file system. How can this happen? I can't figure out why. Is there

Hi, I am having problem with my Asterisk installation and find out it has to do with ztdummy. if the ztdummy module is loaded, the asterisk playback() command will not play files. DTMF is still properly received. If the ztdummy module is unloaded, sound playback works again. Here is my version zaptel-1.4.9.2 linux-source-2.6.18 asterisk-1.4.18 Can anyone tell me how to fix it? Or should I

Hi, I am looking to buy 2 used 1 or 4 ports E1 Cards. If you have one, would you please contact me? Thanks, Pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081217/48afd68d/attachment.htm

Hi, I have to install Broadcom driver because the Dell Optiplex 330 running Centos 5.2 is not able to connect to the network. I am trying to install a Broadcom driver, but I get the followng error: [root at localhost tg3-3.85l]# [root at localhost tg3-3.85l]# make make -C /lib/modules/2.6.18-92.el5/build SUBDIRS=/usr/src/Server/Linux/Driver/tg3-3.85l modules make[1]: Entering directory

On 2017-04-18 08:17 PM, Pete Mundy wrote: >> On 19/04/2017, at 7:58 am, D'Arcy Cain <darcy at VybeNetworks.com >> <mailto:darcy at VybeNetworks.com>> wrote: >> >> <snip> >> Everything looks the same as another one that works except for two >> things. The one that works doesn't have the "Probation passed" lines. >> I am

I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is

Hi, I am sorry my questinos are too fundamental. I am new to Asterisk, and hope to catch up as fast as I can. Problem 1: I have my SIP client ( in one PC .102) and SIP server ( in another PC .101) within the same land. They can make SIP connection, but when the SIP client makes call to play an audio file, I can only hear a "beat" sounds, and then nothing else. In the console, I can

On 2017-04-18 02:42 AM, Pete Mundy wrote: > Try this: > > asterisk -r > core set verbose 10 > [get user to trigger fault] > [examine console output, and post to list if still unclear] > > If you don't solve it yourself, then we'll be able to help further once > we've seen the output. I can't see much more than at my previous debug level but here it is

Hi all, I am hoping someone can help me out on this. I want to be able to interrupt MOH every X seconds after the DIAL command is executed. The interrupt greeting is something like "please wait while we transfer your call". How can I do that? Within the DIAL options, I can't see any announce frequency or options that can help. Could anyone please tell me how that function can

Ok, I have spent all night trying to figure this out, and hopefully somebody has a similar experience. I have a very basic asterisk config. Sample configs, with the only addition being by SIP phone, and my incoming voip. Last week I got everything setup, calls were working, etc.,. I cam across a tutorial for voicemail, followed it, and it worked. When I call my phone and dont answer, it goes

Hi List, Do you know if there are any plans to improve i18n for Asterisk? The current i18n way of doing it with asterisk is very limited and most of the time does not work. For example, take voicemail: "message" "received" "at" "seven" "30" "am" might sound good in English. But: "message" "recu" "a"

On 03/02/15 10:31, Always Learning wrote: > Remember machines should be working for the convenience of Humanity - > not for the convenience of anonymous nutters who know absolutely nothing > about the user's work situation ! 'anonymous nutters'? I guess those people on the cited mail lists are using fake names. Given you're carrying on about being English, maybe you

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was