similar to: Monitor Current outgoing calls

Not sure if a2billing can be shared here, but ill give a shot If the credit < min_credit the IVR play: sorry you have 0 credit and hangup, I want it to FW me to the IVR to add voucher, please let me know: here is log: [18/10/2010 07:01:12]:[file:a2billing.php - line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1] [18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE : standard]

Hey there, I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its web page I see the a2billing directories:Index of /a2billingNameLast modifiedSizeDescriptionParent Directory -admin/15-Sep-2010 19:19-agent/15-Sep-2010 19:21-common/15-Sep-2010 19:18-customer/15-Sep-2010 19:20-Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at Att, Flavio Roberto

Regards! During the use of areski a2billing software I'm getting same problem all the time. Actually, after 15 minutes of speaking to someone over calling card, connection brakes. Installation was as smooth as it could be so I don't think I made same kind of a mess in that domain. This is the only problem in the aplication. In the logs everything seems to be fine. I'am sending You

Hello to all, Im trying to use DeadAGI to implement billing with Asterisk2Billing. Before the billing, I had something like: exten => _2XXXXXXXX,1,Dial(SIP/${EXTEN}@voiprovider) Now, with Asterisk2Billing would be something like this? exten => _2XXXXXXXX,1,Answer exten => _2XXXXXXXX,2,Wait,2 exten => _2XXXXXXXX,3,DeadAGI,a2billing.php exten => _2XXXXXXXX,4,Wait,2 exten =>

Hi All; Anyone has tried to use A2Billing with Asterisk 1.6 and 1.8 to confirm that is working fine and it is same as 1.4? Appreciate ur kindly help. Regards Bilal

Hi, I'm trying to write a AGI in PHP to get the numbers dialed (with read()), save it into a variable to insert it into a SQL server database. But I cannot see results into the variable, it always return NULL. Here is a piece of the AGI. fwrite(STDOUT,"exec Read my_var|/sound_to_play|5|||15 \n"); fflush(STDOUT); $conn=odbc_connect('MSSQL', 'USER', 'PASS');

Hi guys, I've registered some g729 licenses, during register process everything worked fine. astk2*CLI> show g729 0/0 encoders/decoders of 20 licensed channels are currently in use But I'm not able to use this codec. I'm trying to use a linksys PAP2 to talk using g729 but I got this answer from asterisk: Got SIP response 488 "Not Acceptable Here" back from

On 1/11/07, Mike D'Ambrogia <miked@jamagination.com> wrote: > > Ralph > > Kind of new to asterisk, and really new to AGI but it looks like you were > trying to have the AGI script tell asterisk to read and lay the results into > my_var and then regain control in the AGI script, is that correct? > > If so I don't think that will work since the dialplan

Hi All, new to the list. Wondering if anyone has / knows of, a good rate importer tool that can be used to standardize and normalize the ratesheets / rate decks etc. obtained from various carriers so they can be analysed and imported into a DB or be saved as a CSV or something? Thanks so much in advance aeg -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm trying to start with Asterisk, but I could not put 2 softphones to talk. The asterisk server rejects the connections always when I dial. May 17 07:49:22 NOTICE[1924]: Rejected connect attempt from 192.168.0.106 What is necessary to put it to work? There is no need to configure external lines. extensions.conf [internal1] exten => 311000,1,Dial(SIP/teste1) [internal2] exten

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

Hi List; I understand that I have to compile zaptel but what about asterisk? Is it enough to extract it? Well, how I will run asterisk (without compilation and installation)? Any advise? Regards Bilal ____________________________________________________________________________________ Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited

I want to essentially transplant my existing Asterisk server to a new machine, and take the old sever out of service. Assuming I install Asterisk on the new machine, does anyone know what files I would have to copy over? What comes to mind are the *.conf files in /etc/asterisk, as well as the voicemail audio files. Anything else? -------------- next part -------------- An HTML attachment was

Hi, i have the following data.frame : > Data[1:3,] dt amt geoTree merTree ref 1 0.71002484 3.334570 A2b B2b 0 2 0.49074936 2.544464 A2b B1a 0 3 0.06223433 3.617133 A1b B2a 0 i want to convert it to a list, like this: list(Data[1,],Data[2,],Data[3,]) [[1]] dt amt geoTree merTree ref 1 0.07333459 0.969585 A2a

Hello There, I *think* i've setuped the AreskiCC2 Calling Card system right , but i've yet to make any calls out of it , i added a rate card , trunk and defined some rates , generated some users , added 10 dollars in them , okay , now i call any number , it asks me to enter my pin , i do , it tells me i have ten $ , right after that it says sorry you dont have enough funds for this call

Hello R-users,    I have a data frame whose names of columns I don't know a priori, but the user of my code will know them. The user is supposed to save the name of the column that will need some reordering of the levels of the factor later on. The name of the column will be saved in an object called: variab the data frame is called df. If I try to the do following:

OK Dears; Is the exten => _X.,2,Wait,2 no more working with Asterisk 1.8? What is the equivalent? I installed Asterisk 1.8 and Star2Billing 1.9 but I am getting this error if someone can advise me: Executing [9615806234 at a2billing:1] Answer("SIP/gwsshihabuddinkw-00000014", "") in new stack [Apr 5 02:59:05] WARNING[2941]: pbx.c:4055 pbx_extension_helper: No application

Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the

Regards! During the use of areski a2billing software I'm getting same problem all the time. Actually, after 15 minutes of speaking to someone over calling card, connection brakes. Installation was as smooth as it could be so I don't think I made same kind of a mess in that domain. This is the only problem in the aplication. In the logs everything seems to be fine. I'am sending You

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======