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AreskiCC CallingCard Application

Created by: jht2,Last modification on Thu 27 of Oct, 2005 [15:02 UTC] by areski


IMPORTANT : The new version of AreskiCC have changed his name -> "A2Billing"

Please for a followup of the project go to the appropriate pages :







http://areski.net/areskicc-doc-v2/

Help to install AreskiCC

AreskiCC CallingCard Application The idiots guide
AreskiCC CallingCard Application The idiots guideV2

Information from official website:


LAST UPDATE : V2.2 - 9 May 2005


Description :

  • AreskiCC is an AGI script and PHP application which greatly handle the complete CallingCard System.
  • AreskiCC is distributed under GNU GPL.

FEATURES - AGI :

  • Authenticate with the use of a Cardnumber the Cardnumber can also be defined as accountcode into sip.conf, iax.conf, etc..
  • Take care of multiple calls using the same Cardnumber
  • Caller gets informed about his credit
  • Announce the remaining credit
  • Caller is requested to enter a destination number
  • Announce the maximal call time for the given destination number
    • It calculates the remaining duration of the actual call (based on tariffrate tables), informs the caller about this and sets a timeout
  • Interupt the call if the card balance gets zero
    • Warn the caller about the call interupt 60 & 30 seconds before the call gets interupted
  • It connects the Caller to the destination through the configured trunk
    • note : different trunks can be configured and associated by prefix
  • After disconnecting the call AGI updates the credit and stores the concerning Call-Detail-Records with CallingPartyNumber, CalledPartyNumber, CallSetupTime, Duration, Charge and the remaining credit


NEWEST FEATURES IN V2:

  • A new, rebuilt rate engine
  • LCR & LCD management (OOOOHHH YESSSSS)
  • Billing Increment
  • Progressive Rate
  • Scheduled Rates (days of the week)
  • Expiration rates
  • Buy rates configuration
  • Import ratecard from CSV file
  • Simultaneous access for same card
  • SIP/IAX Friends Management
  • Generate conf file for SIP/IAX Friends
  • Reload Asterisk from UI
  • AGI flexibility
    • Many options such as use DNID, Directcall, saybalance, etc.
  • SIP/IAX Friends on AGI (Press 9)
  • Use DNID
  • New graphic design :) Is it cute, no?
  • Internal help/info
  • Customer interface to see the balance & calls made


FEATURES - WEB INTERFACE:

  • CARD/CUSTOMERS
    • List customers/cards
    • Refill customer/card
    • Create customer/card
    • Generate customers/cards
    • List/Create SIP-FRIENDS
    • List/Create IAX-FRIENDS
    • Generate Asterisk configuration file
    • Reload Asterisk
  • BILLING
    • View money situation
    • View Payment
    • Add new Payment
  • RATECARD
    • List/Create TariffGroup (LCR/LCD)
    • List/Create Ratecard
    • Define Ratecard
    • Add Rates
    • Import Ratecard (csv files)
  • TRUNK
    • List Trunk
    • Add Trunk
  • CALL REPORT - BALANCE
  • LIST/CREATE - USER ACCESS (ADMIN)

REQUIREMENTS :

  • Apache
  • PHP & php-pgsql
  • postgresql
  • use phpagi included (http://phpagi.sourceforge.net)
  • php.ini : register_global = On


ABSOLUTELY, YOU MUST USE DeadAGI :

   if the called party hangup the CDR sessiontime is null, to solve this issue,
   ; CallingCard application
   exten => _X.,1,Answer
   exten => _X.,2,Wait,2
   exten => _X.,3,DeadAGI,areskicc.php
   exten => _X.,4,Wait,2
   exten => _X.,5,Hangup

WEBSITE : http://areski.net/areskicc-doc-v2/


More details about the release CHANGELOG.txt


Note :

  • by default the username & password for the UI is root/myroot or admin/mypassword


(:lol:)


SCREENSHOT :

Image


Image





Comments

Comments Filter
222

333Re:

by asifshabbir, Sunday 05 of February, 2006 [12:06:07 UTC]
i am also facing the same problem , did u get any solution for this . pls guide me as well
222

333Re:

by asifshabbir, Sunday 05 of February, 2006 [12:02:40 UTC]
i am also facing the same problem , did u get any solution for this . pls guide me as well
222

333

by overseacalling, Sunday 20 of November, 2005 [07:45:04 UTC]
Hello everyone! ... I have read little about a2billing here and Areski have installed for me the A2Billing application but the problem I am having now is that, I cant call out of the trunk. when I dial a numbers from my dialer, it give me the balance and ask me to enter the destination then after I enter the destination #, the voice just tell me (the numbers you have dail is currently unavailable). my system was working before install this a2billing but it was just straight from the asterisk without billing. I could called any pstn phone numbers, but after this software installed now I am stuck. I have setup the trunk and everything but it seem I cant go no where.... If anyone here can help me or have issue withe me please give me a hint .... thanks


222

333Re: only see the logout button. Please help

by albravob, Tuesday 25 of October, 2005 [23:34:56 UTC]
Did you get any response to your question? I'm having the same exact problem as yours....
222

333Re: only see the logout button. Please help

by albravob, Tuesday 25 of October, 2005 [23:34:34 UTC]
Did you get any response to your question? I'm having the same exact problem as yours....
222

333re: only see the logout button. Please help

by rikunj, Sunday 16 of October, 2005 [07:57:12 UTC]
Sorted.

Recompiled php with --with-config-file-path=PATH to /etc/

Rikunj

222

333only see the logout button. Please help

by rikunj, Sunday 16 of October, 2005 [07:22:33 UTC]
Installed Areski, following exact procedures from manuals.

with register globals on, tcp_ip connections allowed in postgresql.conf, and trust the wolrd in pg_hba.conf.

I log in but only see the logout button.
It seems it does not authenticate, and allows to login even with incorrect login user and password.

Able to login to postgres through phpPgAdmin 3.5.5.

Where am I going wrong, Please help.

222

333How Can I make call through Trunk in h323 channel

by goksie, Saturday 01 of October, 2005 [17:55:24 UTC]
Thank to Areski himself. its a great job... howver, I was able to make sip to sip call, sip -iax and iax calls. howver, I am not able to pass my call to my h323 gateway. I am sure the h323 gateway is working because I had used it before installing areski.

Can somebody help out so that I can add trunk running gnugk and got the call through to my cisco gateway.

Thank you

goksie


222

333How Can I make call through Trunk in h323 channel?

by goksie, Saturday 01 of October, 2005 [17:45:19 UTC]
Thank to Areski himself. its a great job... howver, I was able to make sip to sip call, sip -iax and iax calls. howver, I am not able to pass my call to my h323 gateway. I am sure the h323 gateway is working because I had used it before installing areski.

Can somebody help out so that I can add trunk running gnugk and got the call through to my cisco gateway.

Thank you

goksie


222

333Donation To Areski's Project ( *****)

by charliechaklam, Friday 16 of September, 2005 [15:30:39 UTC]
I Would like to Remind and encourage people to donate to this project. Every sensable person would understand that programing takes hours and hours of hard work, if you put the time into installing this project on your machine and comming back for help to make it work, you can probebly understand how much hard work areski put into this.

On the top of this post section i see some one saying areski not honest man, well for people like that , who only want things for free and think that they deserve them and try to pressure other people to give them free. Why dont they simply stop using this project.

Every one that puts time to aquire and install this project knows the value of this solution.

So lets all make a simple 20 to 25 dollar donation, and if thats too much how about a little less.

Lets togather encourage Areski to continue produce his brand of excellance.

( I would like to know what people think of this post). (:confused:)

charlie