login | register
Sun 07 of Sep, 2008 [10:30 UTC]

voip-info.org

A2Billing

Created by: areski,Last modification on Thu 22 of Nov, 2007 [15:16 UTC] by jq53333
http://www.asterisk2billing.org/

LAST VERSION : V1.3.0 Stable (ADMIN UI V1.3 - CUST UI V1.3 - AGI V1.3 - DB V1.3)



Asterisk2Billing V1.3


A2Billing has just released the latest version of A2Billing, a beta, with lots of bug fixes and many new features, which gives any Telecom company a very good reason to consider the A2Billing Platform over the traditional offerings for TDM and VoIP termination as well as wholesale billing.

A2Billing has completely re-written the call-back modules, added new methods of online payment integration with Moneybookers and Authorize.net in addition to Paypal. A2Billing have also improved the rating engine, giving the operator the ability to create Free Minutes packages to certain destinations. Additional reporting functions and alarms have also been added in the interests of revenue protection including automatic emails for High or Low ASR (Answer Seize Ratio), ALOC (Average Length of Call) and CIC (Consecutive Incomplete Calls) Alarms.

A2Billing is is now a fully featured telecom platform providing converged services, with self contained billing, reporting and statistics for IP and TDM based voice networks and can be configured to supply a wide range of services, rate calls, prepare and send out invoices, as well as accept payments via a number of payment service providers.

The A2Billing solution comprises of the following components: -

   * Linux The base operating system
   * Asterisk The telephony engine
   * Apache The web server
   * MySQL/Postgresql The back end database
   * A2Billing The Billing engine handling Authentication, Authorisation and Accounting. 

The platform can be fitted with a TDM cards to interconnect with the PSTN, and can support in excess of 120 concurrent calls given the right hardware. If more capacity is required, then more Asterisk servers can be added as necessary.

The A2Billing Platform has been deployed in a number of commercial environments by both traditional TDM based telecoms companies wishing to move into the VoIP market, and calling card and call-shop businesses. Additionally, there has been a lot of interest from IT and networking companies who are beginning to deploy VoIP PBXs in addition to their traditional business, and wish to enjoy an ongoing income by terminating their customer’s calls using A2Billing as their Wholesale Billing Platform

  1. Asterisk2Billing is distributed under GNU GPL.

KEY FEATURES :


   * Authenticate with the use of a Cardnumber
   * 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 - Interrupt the call if the card balance 
     reaches zero - Warn the caller about the call interupt X seconds before the call gets interrupted - 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
   * Reporting
         o Monthly & daily reporting
         o monthly traffic reports (pie graph)
         o Daily load
         o compare call-load with previous days
         o criterias definition for reporting
         o export report to PDF & CSV
         o Generate Invoices to PDF format
   * Powerfull rate-engine
         o LCR & LCD management
         o Billing Increment
         o Progressive Rate
         o Scheduled Rates (days of the weeks)
         o Expiration rates
         o importation ratecard from csv file
   * Simultaneous access for same card
   * SIP/IAX Friends Management
   * Generate conf file for SIP/IAX Friends
   * Reload Asterisk through UI & mananager
   * IVR Customization - many options such as use DNID, Directcall, saybalance, ...
   * Free Call on SIP/IAX Friends on AGI (press 9)
   * USE DNID to pass through calls
   * Internal help/info
   * Multi-language
   * post-pay & prepay
   * callerID authentication
   * Setup musiconhold according to the destination ;)
   * failover trunk configuration
   * Recurring service over the card
   * Complex expiration setup for the card
   * Voucher support
   * Currencies support management - use www.oanda.com for currencies list
   * ACL support for admin users (added 30 Oct 2005)
   * Advanced filter over the cards (added 30 Oct 2005)
   * Advanced filter over the Rates (added 5 Nov 2005)
   * Batch update for Rates and Cards (added 6 Nov 2005)
   * Signup modules (added 7 Nov 2005)



KEY FEATURES FROM 1.1 VERSION :


   * Ecommerce product with API addons - Integration with OsCommerce
   * Speeddial-support for UIs (Customer & Admin)
   * Add DB backup/restore tool
   * Currencies support management - yahoo financial (cront for auto update) - Add new model for update currencies from Yahoo , now currencies are in Database in cc_currencies table. Remove rates.inc and any information about.
   * Signup autocreates SIP/IAX
   * New features for PEAK & OFF-PEAK - Add new model for ratecard , removing week day and adding starttime and endtime instead.
   * Add Voip provider
   * Add the RATECARD SIMULATOR
   * Add support for Jiax web phone
   * notenoughcredit_assign_newcardnumber_cid - IF the CARD doesn't have enough credit, request to enter a new cardnumber. Assign the CallerID to the new cardnumber
   * Predictive Dialer Features - Manage Campaign, Phonelist, import phonelist.
     Customer Interface (Agent) have the ability to call a predefined amount of Phonenumber.
   * Support call at Zero-Cost & Negatif cost(plus param = maxtime_tocall_negatif_free_route)
   * CallerID authentication improvement - (new param : notenoughcredit_cardnumber ; cid_auto_assign_card_to_cid ; cid_auto_create_card ; cid_auto_assign_card_to_cid)
   * popup select card to avoid long load (issue for user that have create lot of cards)
   * PAYPAL SUPPORT - IPN - Customer can buy credit through paypal
   * DID SELLING SUPPORT - features to sell to your customer preconfigured DID.
     Customer would have the opportunity to redirect those to his phonenumber
     and even deploy a Follow-Me
   * DID monthly billing

KEY FEATURES FROM LAST 1.2 VERSION :


   * Very big change on the whole code, almost a full rewriting on the web interface, the Form layer is based over a new OO class, so the code is more structured and organized and All functionalities are centralized.
   * CallBack : Web callback from customer interface, ANI callback, DID callback
   * Multi Language support for the customer interface (Spanish, English, French, Chinese, Italian, Romanian, Turkish, Urdu)
   * SOAP-Webservice : Create Card, Remove card, Update... See WSDL : A2Billing_UI/api/SOAP/soap-card-server.php?wsdl
   * Ratecard Simulator on Customer interface
   * CallerID Update on Customer interface
   * Based on Adodb, dbms layer
   * Features to export to XML
   * Update of the Web Dialer WebPhone Jiax
   * Add the VAT on the customer invoice
   * Better support for ARA
   * lot of new little features and bug solved :D


REQUIREMENTS :


   * Apache
   * PHP
   * PHP-PGSQL or PHP-MYSQL
   * PHP-PCNTL
   * PHP-GETTEXT 
   * PEAR SOAP class
   * PostgreSQL or MySQL
   * Use PHPAGI 2.14 included (http://phpagi.sourceforge.net written by Matthew Asham)





WEBSITE : http://www.asterisk2billing.org/


Auto install script for A2Billing can be found here:

WEBSITE :http://a2billing2asterisk.googlepages.com


-Above script will install A2Billing with simple three commands on your latest Trixbox machine or vanilla Asterisk (not fully test with vanilla asterisk).


-Make sure you read the terms before you proceed as it will over-write your sip.conf, iax.conf, manager.conf, and other configuration files.



Notes :


SCREENSHOT :

Image


Image

Image



Comments

Comments Filter
222

333cdr configuration

by astsanthosh, Tuesday 19 of June, 2007 [14:28:48 UTC]
Hi,

I am using a2billing beat 1.3 version with asterisk1.2 on centOS.
How can i configure the a2billing such that it will take the cdrs generated by the asterisk server ?
And where we have to configure ?

Any help regarding this is much appreciated

Regards,
Santhosh
222

333A2Billing upgrade and setup

by TalkPBX, Thursday 21 of December, 2006 [16:48:00 UTC]
Take a look here for directions on how to upgrade A2Billing: http://www.sureteq.com/asterisk/a2b.htm and take a look here for directions on how to do an initial setup with A2Billing: http://www.talkpbx.com/a2binstructions.htm
222

333Re: a2billing trunks problem

by ngocgl, Monday 29 of May, 2006 [10:04:17 UTC]
I have the same problem with Jose Babu, please tell me some where I can find a more detail installation guide
222

333Re: re: a2billing setup

by telcomm1, Sunday 07 of May, 2006 [20:46:13 UTC]
As far as making it work with VoiPJet, the answer is yes, however, getting VoiPJet working in A2Billing was a bit tricky because of how VoiPJet needs to be called. I had to change switchdialcommand=NO in the A2Billing.conf file and then it created problems with other providers. You then have to set the trunk to ####@voipjet in order to get it to append the account number in the Dial String. It was a pain but it worked. I'd recommend www.telcommone.net instead, since you don't have to alter your switchdialcommand parameter, you simply have to create a SIP entry for telcommone, then you can put "telcommone" as the "Provider IP" and just put your 6 digit PIN in the "Add Prefix" section. Works nicely. Going to create Wiki page on it soon.
222

333Upgrade a2billing from Asterisk@home 2.8 to the new version

by eportel, Friday 14 of April, 2006 [00:05:16 UTC]
Hello,
Can someone help me out, I need a help to Upgrade a2billing from Asterisk@home 2.8 to the new version of a2billing.
can you tell me the step to take for upgrade.
Thanks
Henry
222

333Upgrade a2billing from Asterisk@home 2.8 to the new version

by eportel, Thursday 13 of April, 2006 [23:57:09 UTC]
Hello,
Can someone help me out, I need a help to Upgrade a2billing from Asterisk@home 2.8 to the new version of a2billing.
can you tell me the step to take for upgrade.
Thanks
Henry
222

333a2billing trunks problem

by josebabu, Thursday 16 of March, 2006 [20:43:58 UTC]
I am having the same problem Installed a2billing , created cards etc. but when I try to dial a number it says the number you have dialed is currently unavaible..."

Please help !
222

333a2billing trunks problem

by asifshabbir, Sunday 05 of February, 2006 [11:54:54 UTC]
can anyone tell me how can i make the outgoing calls with a2billing.
i have created calling card and asterisk tells me my current balance,but when i dial to the outgoing number it says " the number you have dialed is currently unavaible..." pls guide me how can i do this
222

333If you need help contact me.

by shaonss, Monday 23 of January, 2006 [14:56:49 UTC]
We tested this application and works freat. if you need any help contact me through www.apphone.com
222

333

by josem, Friday 25 of November, 2005 [17:50:33 UTC]
Hi all. Sorry, please, I need help: I have a2billing installed but not work, only execute this:
-- Executing Answer("SIP/20-7f96", "") in new stack
   — Executing Wait("SIP/20-7f96", "2") in new stack
   — Executing DeadAGI("SIP/20-7f96", "a2billing.php") in new stack
   — Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
   — AGI Script a2billing.php completed, returning 0
   — Executing Wait("SIP/20-7f96", "2") in new stack
   — Executing Hangup("SIP/20-7f96", "") in new stack
 == Spawn extension (from-internal, 1, 5) exited non-zero on 'SIP/20-7f96'
   — Executing Macro("SIP/20-7f96", "hangupcall") in new stack
   — Executing ResetCDR("SIP/20-7f96", "w") in new stack
   — Executing NoCDR("SIP/20-7f96", "") in new stack
   — Executing Wait("SIP/20-7f96", "5") in new stack
   — Executing Hangup("SIP/20-7f96", "") in new stack
 == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/20-7f96' in macro 'hangupcall'
 == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/20-7f96'

Any ideas?? that I need to do or that I am not doing well?

Thanks in advance.