provided by Star2Billing S.L.

This is just more than perfact for its target.

The thing I got amaged that we are able to run even for users with ordinary prepay and postpay.

The most out of it that we could bill for users of "SER". OOHH, damn. Just no words to say


Please mind it:- We are still running his older version [a2billing] successfully.

And the new version [asterisk2billing] is still under testing on test server, not yet on our Production server.

As of 31st May 2006:-

We are already running Chamleon_v1.1 for last three weeks on production server. call accounting, paypal - works great [bugs are same as before - what is posted on forum].

hello

i need help in using a2billing. can someone give me the steps
to create a single card and also how to upload the rates.
when i make a call it gives me teh balance on teh account
but i got the message: phone is unavailable.

Hi, it seems you are using asterisk2billing in a production enviroment. Could you please tell me how many users you have, and how does a2b do under some heavy load?
I'd really need to know some details as what server is it running on, how many simultaneous users you have on heavy hours, etc.

Thanks,

Juan Alvarez

it seems [guessing], you have quite all setup accordingly. just set your trunk rightly to send calls to your PSTN provider then you are done.

To creat rate cards:-

1st. Create TariffGroup
2nd. Create new RateCard
3rd. List TariffGroup; and EDIT your TariffGroup
4th. Choose TARIFFPLAN:- [2nd]new rate card & Add. Then confirm Data

now:-
5th. Add rate to [2nd] rate card

Add rate link:- This is the minimum parameters to start with

5.a. DIALPREFIX
5.b. DESTINATION
5.c. BUYING RATE
5.d. SELLING RATE
5.e. TRUNK

thats all, just submit the data, you are ready to go. >But logically, you can not put rate one by one.

You have to import rate card [this is minimum to select from that import page]:- Hoping, you have other setup rightly ...

6.a. Choose the ratecard to import :
6.b. Choose the trunk to use :
6.c. Browse your CSV file and press Import Ratecard
6.d. It will confirm you with first line of rate.csv file. But here is not finish to import to ratecard. You have to again browse the same rate.csv file [waht you selected on previous page] and press "Continue to import the Ratecard" on that page. thats all.

::-- I've inserted all DEMO rates under voippanel_ratecard on areski's demo site. To check, click on "Ratecard --> Define RateCard --> choose voippanel_ratecard from drop down menu "

- If you need rate.csv file to import with asterisk2billing, leave me with your email [you may drop on private message, if you dont want publicly]. I will send it.

Hello,

1st : -about users:-

2nd : We have our own billing system for Prepaid , postpaid and wholesale. but last 8 months we have set a2billing in parallel for prepaid, postpaid & calling card. But The target idea was to use a2billing for calling card.

3rd: Its running just perfact [for us] with multiple agi-conf [several different parameters]. didnt crash even for single time. Easier for us to compare the billing as we've parallel.

4th. Servers: There are lot of things to consider [users, RTP, codec & protocol conversion], a2b itself doesnt cause so much CPU overhead but depend on asterisk. main what is your target and how many you want to handle.

5th. It is better to consider to share the loading. Dont let to handle the same machine to handle sip users, RTP, SQL, http. use distributed.

6th. we are trying to set a2b in wholesale manner, we will put links also for live test.

Hope this helps.

Note: we didnt put yet his latest version"asterisk2billing" in a production enviroment but the earlier

Please I need help. Is this possible to change the default timeout for ringing or nobody picked value?

Thanks in advance.

Hello,

by default in a2billing.conf:

the time out [FOR THE CALLED PARTY TO PICK UP THE PHONE] BETWEEN 30 seconds:-

The dial command looks:-
dialcommand_param="|30|HL(%timeout%:61000:30000)"

AND

As for example to change it to 90 seconds timeout:-

dialcommand_param="|90|HL(%timeout%:61000:30000)"

thank you, it works.

Ref. [Previous Conversation from below link]
http://forum.asterisk2billing.org/viewt ... =1346#1346



Hello there,

These are just g729 sound files. If you use IVR of g729 sound files in between - its not anymore PAssthrough. SO, You have to have codec to understand those g729 sound files. [eg. Can you play mp3 songs without MP3 player or MP3 codec??].

Ref. http://forum.asterisk2billing.org/viewt ... =1378#1378


This topic is not actually for this forum, but anyway posting here - as there are many inquiry about it.



1st. we are assuming - Asterisk is properly setup & not running! So 1st command [cmd]

[root@voippanel root]# asterisk -vvvvvc

Note: check from output [and the end output should show - Asterisk Ready]- what are the errors & fix it. If there is no errors[fine], May be there could be some warnings[that not setup properly]. But you could ignore - if you dont need those parameters.

2nd. [After the above command on Asterisk console mode]
finndial*CLI>stop now

3rd.
[root@voippanel root]# asterisk
Note: Now your asterisk is running

4th. [If you would like to see - whats going on Asterisk]
[root@voippanel root]# asterisk -rvvv

Enjoy! Welcome on board of Asterisk & Areski [A&A].

NOTES: if you would like to load any new * modules - stop * and follow from 1st step. Otherwise just below cmd to reload everything on * console.

voippanel*CLI>reload

OR - after third step[means - * already running] to reload sip.conf - if any changed happened in sip.conf

[root@voippanel root]#asterisk -rx "sip reload"

NOTE: If you change parameters & would like to reload conf file of H323. It doesnt work just by cmd reload on * console.

Steps are:-
if * already running

voippanel*CLI>stop now
And then follow from 1st step.

NOTE: stop now command will shutdown your * completely, so you have start * again for normal operation.

If you want that users, will not get the prompt to enter the PIN number.

We assume:-

You have done other setup accordingly & its working normally, Now you just want to disable the Prompt for some users:-

1st step: you've to edit in a2billing.conf

[root@voippanel root]# vi /etc/asterisk/a2billing.conf
Note: use your file editor either [ vi OR pico OR what you have]

2nd step: set the parameter as below [highlighted in RED] in a2billing.conf

2a.

; if YES it will catch the DNID and try to dial it out directly without asking for the phonenumber to call
; value : YES, NO
use_dnid=YES


2b.

; enable the callerid authentication
; if this option is active the CC system will check the CID of caller
cid_enable=YES


3rd step:

3a.
Goto Asterisk2Billing Admin web interface.

3b.
Click on "EDIT" of the cardnumber that you gave to your user. [And now you want to enable CID for that user. So that he/she doesnt have to enter the PIN/CARD number over again].

3c.
Under different properties in that card, go to bottom of that card - where is written "Add a new CALLERID" and

Enter CALLERID in the input field. Then press on Add button. You are Done for this part.

NOTE:
1. If you are using Asterisk2Billing for your SIP & IAX users - the CALLERID should be the "CARD NUMBER".

2. If you are using external system for your users. You can set "Asterisk2Billing" in between call path for Billing. In that case, you have to enter the CALLERID from that system, what users are holding.

Do you mean that a2billing can do billing for SER ? How to config?
Please help. thanks.

Hello,

SER -> ASTERISK -> Asterisk2Billing

Ref. http://forum.asterisk2billing.org/viewtopic.php?t=492


Hello,

Here is Little Patch:-


This is just $ sign on the script. but you can do so:

go to line -97,1 + 97,2 on file:
/var/www/html/A2BCustomer_ui/userinfo.php


- <br><b><?php echo BALANCE_REMAINING?> :</b>$ <?php echo $customer_info[1]; ?>

+ <br><b><?php echo BALANCE_REMAINING?> :</b><?php echo $value.' '.BASE_CURRENCY ?> <?php echo $customer_info[1]; ?>


So, the customer interface will show the currency, what you have written for

base_currency = EUR

under [webui] section in a2billing.conf