similar to: Internationalisation/Internationalization

Hi Vic, I hit that same problem! My SIP phones would sound okay when I made changes to indications.conf but incoming calls in to my TE410P had their own thing going on! Have a look at the zaptel source files, there's one called zonedata.c. You'll see the au settings... replace what's there with this: { 1, "au", "Australia", { 400, 200, 400, 2000 }, { {

Hello list, I've been experiencing choppy sound as well. The version on Asterisk I was using originally was dated 10/24/03 (I think), the problem appeared after I updated from that version. My setup is a little different though. I'm having choppy sound only on some incoming calls -- from PSTN->PBX (between spans on a TE410) and PSTN->SIP. We use Cisco 7940 handsets and we also

We are not having any luck with the E100p card here in Australia, it will work with a crossover cable to another device but will not talk to our Telco Telstra who probably have a weird implementation of an E1. Any suggestions on a replacement? Regards Mark McKibbin DCS Internet 64 Queen St Warragul Victoria 3820 Australia www.dcsi.net.au mark@team.dcsi.net.au Ph. 1300 665575 Fx. 1300 556595

This is not really a problem just something I noticed in my testing.. When two or more Asterisk servers are connected by IAX2 trunks it does not make use of any "shortest path" type system.. (maybe this is still planned somwhere down the line, but may come in handy to those who have multi asterisk installations) Here is the setup.. UA1--- Asterisk1----[IAX2 Trunk]---Asterisk2---UA2

hi, i saw the cdr file called Master.csv and i want to know what these represent. examples "","","4","incoming","","Zap/1-1","Zap/4-1","Voicemail","u8888","2003-11-07 17:43:04","2003-11-07 17:43:04","2003-11-07 17:43:22","ANSWERED","DOCUMENTATION"

In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd like to do something possibly unique with the formatting of extensions in my dial plan, and am having trouble. We use VoicePulse connect, which gives us local DID for inbound and outbound calls (even though DTMF tones are not working in Voice Pulse Connect at the moment). To dial local numbers, you have to

asterisk*CLI> load cdr_unixodbc.so Loaded /usr/lib/asterisk/modules/cdr_unixodbc.so => (unixODBC CDR Backend) == Parsing '/etc/asterisk/cdr_unixodbc.conf': == Parsing '/etc/asterisk/cdr_unixodbc.conf': Found -- cdr_unixodbc: dsn is MySQL-asterisk -- cdr_unixodbc: username is root -- cdr_unixodbc: password is [secret] -- Connected to MySQL-asterisk it

Hello All, I have pingtel and asterisk working really well. I have a really annoying little problem - mainly with pingtel. When a call comes in pingtel displays the caller ID on the phone. If I miss it then I click on the number for redial - this doesn't include a 9 to dial an outside line. The second problem is with the dialer from outlook again it bypasses the outlook dialing rules so

I would just like to get a refresh of the situation for Australian users. It would seem that the TE400P is currently available and is likely to acheive approval for use in Australia within the next 2 months ? (when is the end of summer?). Once this is done, it will certainly suit larger installations, but it still leaves a number of 'gaps'. Anyone with analog phone lines will need

Does anybody have useful tips on creating good quality recordings for use with prompts in asterisk? I'm interested in hearing input on hardware (mics, dats, sound cards, etc) and software (recording software, dsp) as well as recording techniques. Jerimiah Tularosa Communications

I've seen this question asked before but haven't seen a definative answer. Does the e100p work in australia? Did any one who was asking the question before bite the bullet and get one? I can get a te410 if i really have to but would prefer to stay with the cheaper option. Any comments appreciated, Thanks ------------------------------------------------- This mail sent through IMP:

Hi list! I have a little question about groups and Asterisk ... is there anyone out there that can say if Asterisk can do any of this; We have a customer that want call handling we cant give him with a traditional PBX, and I'm running Asterisk @home so I thought I could give it a try ... The customer wants that incoming call should go to one group with some phones in it, if the group is

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I've added internationalisation to guestconv, and updated HACKING with a new section. Any thoughts/comments? Matt

I''m trying to use haml in my locale files but cant get it to parse. en.yml --------- page: content: " %p some paragraph %p another paragraph " view.html.haml ------------------- (works) =Haml::Engine.new(t(''page.content)).render -------------------(does not) =t(''page.content) is the first example the only way to do this? or am I going about

Hello! I am trying to migrate from Teapop to Dovecot. I'm using Dovecot 1.1.14. I need to make use of the X-UIDL header so clients don't download messages they've already received. I have the X-UIDL head in the message. I have pop3_reuse_xuidl(pop3): yes My problem is the X-UIDL in the message is being ignored and Dovecot is still generating it's own. The INBOX is an mbx

I've got the call cuing all setup and working, but im trying to get the Callswaiting,you are caller #, etc, and its not working. I have the following inthere as stated: queue-youarenext = "queue-youarenext" ("You are now first in line.") queue-thereare = "queue-thereare" ("There are") queue-callswaiting = "queue-callswaiting" ("calls

Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using

I'm looking for the patch that enables suppotr for the following lines in queue.conf: announce-holdtime = yes queue-youarenext = "queue-youarenext" queue-thereare = "queue-thereare" queue-callswaiting = "queue-callswaiting" queue-holdtime = "queue-holdtime" queue-minutes = "queue-minutes" queue-thankyou = "queue-thankyou" It

Hello list, There seem to be some problem with the ACD of asterisk where when we use this parameter in queues.conf . We could not get any announcement as expected. Iam useing the latest CVS-head Even weight also doesnot seem to work properly I tried like this where we have two queues one with 100 weight and another with 200 as weight when both enter into the queue when queue is empty when