[asterisk-users] Billing for minutes

2016-04-13 Thread Tech Support
All;

I was wondering what people are doing to bill customers for minutes. I
know that A2Billing is a popular option, but I was wondering if there are
other good alternatives. They don't necessarily have to be free, but they
need to be cost effective. Any insight at all would be greatly appreciated.

Thanks;

John V. 

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[asterisk-users] Billing

2015-03-14 Thread Zakir Mahomedy
Hi

I have the following topology:


Sever A <> IAX2 <-> SERVER B <> SIP <> ITSP

Server A : Branch Server ( with billing cdr )  :  Billing module for auditing 
calls at branch level
Server B : Billing Server   ( with billing cdr )  : Official statements get 
generating from this server

I am trying to match my billsecs on server A TO that of server B
The problem seems to be that server A is counting the ringing as well, not just 
the ANSwered ( person talking )

If the phone rings for 25 secs and we speak for 30 secs, billsec is 55 secs and 
duration is 55 secs
CDR values for deposition are correct if if the call has not been answered or 
busy.
ie NO ANSWER or BUSY on Server A.

Asterisk 13 on branch server
any ideas on how to trouble shoot this problemthanks
Zakir
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Re: [asterisk-users] Billing software: Other than A2Billing because of the problem with the analogue channels

2014-08-22 Thread A J Stiles
On Thursday 21 Aug 2014, bilal ghayyad wrote:
> Hello;
> 
> I am facing a trouble with A2Billing when using analogue lines because the
> channels are not closing properly when dialing happen through A2Billing
> (it seems the dialing scenario including the hangup is not handled
> properly through A2Billing but I do not have control on this). But when I
> do dialing from asterisk and using analogue lines, I do not face a trouble
> because I can write the script in the extensions.conf in professional way
> to confirm that the channel is closed successfully.

If you think you have managed to detect the end of a call more successfully 
than A2Billing can, I would have thought the logical solution would be to 
patch A2Billing to work with your improved teardown detection.

> Is there alternative Billing solution than A2Billing which has another
> working mechanism?

Not really.  The problem is inherent to analogue lines; which by definition 
cannot carry full supervisory information, as there is no separate D-channel.

> How I can resolve such problem which is related to the
> analogue channels? 

Get some form of digital phone line  (such as a SIP trunk, or ISDN)  installed 
instead.  You should be able to switch from analogue to ISDN without 
terminating your existing telco contract; and the quarterly will be cheaper, 
as you are using less of the telco's equipment.


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[asterisk-users] Billing software: Other than A2Billing because of the problem with the analogue channels

2014-08-21 Thread bilal ghayyad
Hello;

I am facing a trouble with A2Billing when using analogue lines because the 
channels are not closing properly when dialing happen through A2Billing (it 
seems the dialing scenario including the hangup is not handled properly through 
A2Billing but I do not have control on this). But when I do dialing from 
asterisk and using analogue lines, I do not face a trouble because I can write 
the script in the extensions.conf in professional way to confirm that the 
channel is closed successfully.

Is there alternative Billing solution than A2Billing which has another working 
mechanism? How I can resolve such problem which is related to the analogue 
channels? 

Regards
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Re: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1

2014-03-17 Thread Steve Totaro
Gateway computers rejects calls like this.  I was informed that their
carrier rejects the calls because they cannot accurately bill.

It seems pretty silly with voip and number portability.

Thanks,
Steve T

On Mar 17, 2014 5:19 PM, "Eric Wieling"  wrote:
>
> Often it is P-Asserted-ID, but depends on the carrier.  You should be
asking your carrier how to do this.   Be careful, if the carrier doesn't
like your CID spoofing they might bill the call to a default number on the
account.
>
> I suspect it is the destination which is rejecting the call because toll
free numbers are not considered valid, not your carrier rejecting the call.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Positively Optimistic
> Sent: Monday, March 17, 2014 4:47 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Billing number vs. CallerID number | Asterisk
11.5.1
>
> In a multi-tenant environment, we are sending various CallerIDs outbound
from asterisk based on who the user is.  We have an insurance agency who
would like to present a toll free callerid.  This works..  unless they're
calling a toll free number.  In that case, occasionally, the call fails.
 However, should we send a correctly formatted npanxx of a local number,
the call completes.
>
> We have been advised that we can send the billing telephone number as a
separate header and the call will complete, all-the-while, presenting the
toll free number as the caller id.
>
> Does anyone know of the correct header required to provide this
functionality?
>
>
>
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Re: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1

2014-03-17 Thread Jim Lucas

On 03/17/2014 01:56 PM, Eric Wieling wrote:

Often it is P-Asserted-ID, but depends on the carrier.  You should be asking 
your carrier how to do this.   Be careful, if the carrier doesn't like your CID 
spoofing they might bill the call to a default number on the account.


Speaking as a carrier that allows this, we require the P-Asserted-Identity 
field.  This is the example of a header that we insert with our SBC:


P-Asserted-Identity: 

The phone number is the identifying marker to tell our Metaswitch the needed 
information to associate the call to the correct object for billing and call 
restriction purposes.


The IP is the internal IP of our Metaswitch.  It is the internal IP due to our 
MetaSwitch being behind our kamailio SBC.




I suspect it is the destination which is rejecting the call because toll free 
numbers are not considered valid, not your carrier rejecting the call.


As a carrier, I have never seen a case where a call (inbound or outbound) was 
rejected because the received caller ID string contained a toll free number. 
For me, as long as it passes the number validation step, we are good.  And a 
toll free number looks like any other NAMPA number.




-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Positively 
Optimistic
Sent: Monday, March 17, 2014 4:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1

In a multi-tenant environment, we are sending various CallerIDs outbound from 
asterisk based on who the user is.  We have an insurance agency who would like 
to present a toll free callerid.  This works..  unless they're calling a toll 
free number.  In that case, occasionally, the call fails.  However, should we 
send a correctly formatted npanxx of a local number, the call completes.

We have been advised that we can send the billing telephone number as a 
separate header and the call will complete, all-the-while, presenting the toll 
free number as the caller id.

Does anyone know of the correct header required to provide this functionality?






--
Jim Lucas

http://www.cmsws.com/
http://www.cmsws.com/examples/

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Re: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1

2014-03-17 Thread Eric Wieling
Often it is P-Asserted-ID, but depends on the carrier.  You should be asking 
your carrier how to do this.   Be careful, if the carrier doesn't like your CID 
spoofing they might bill the call to a default number on the account.

I suspect it is the destination which is rejecting the call because toll free 
numbers are not considered valid, not your carrier rejecting the call.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Positively 
Optimistic
Sent: Monday, March 17, 2014 4:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1

In a multi-tenant environment, we are sending various CallerIDs outbound from 
asterisk based on who the user is.  We have an insurance agency who would like 
to present a toll free callerid.  This works..  unless they're calling a toll 
free number.  In that case, occasionally, the call fails.  However, should we 
send a correctly formatted npanxx of a local number, the call completes.  

We have been advised that we can send the billing telephone number as a 
separate header and the call will complete, all-the-while, presenting the toll 
free number as the caller id.

Does anyone know of the correct header required to provide this functionality?  
 



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[asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1

2014-03-17 Thread Positively Optimistic
In a multi-tenant environment, we are sending various CallerIDs outbound
from asterisk based on who the user is.  We have an insurance agency who
would like to present a toll free callerid.  This works..  unless they're
calling a toll free number.  In that case, occasionally, the call fails.
 However, should we send a correctly formatted npanxx of a local number,
the call completes.

We have been advised that we can send the billing telephone number as a
separate header and the call will complete, all-the-while, presenting the
toll free number as the caller id.

Does anyone know of the correct header required to provide this
functionality?
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Re: [asterisk-users] billing based on local access number

2010-02-10 Thread C. Chad Wallace

At 4:02 AM on 10 Feb 2010, umesh maharjan wrote:

> 
> Hi all,
> 
> I am configuring asterisk as a prepaid calling card. I am getting
> different local rate from my ISDN provider e.g  0.002 for landline
> and 0.13 for mobile etc. In this case I thing I have to say my
> asterisk/a2billing to bill based on local access number. so How can I
> retrieve  called number (eg. 03-6832-1040 and 0120-272-060 is our
> ISDN PRI access number) to my asterisk server so i can trigger
> different rates. 

The number the caller called to get to you should be passed to Asterisk
as the inbound extension.  So, in your incoming context, you can
provide different extensions for the different incoming numbers.  Or
you can catch everything with the "_X." pattern and use the ${EXTEN}
variable to check the number in your dialplan.

One thing to note is that it doesn't always pass the whole number.  I
have two PRIs from different providers; one of them passes all 10
digits, but the other one only passes the last 4, and for some reason
with one of our numbers that ends in "9977" the PRI passes "2977".  You
can either ask your provider what they pass, or you can just make test
calls and log the value of the ${EXTEN} variable with Verbose() calls,
something like this:

[incoming]
exten => _X.,1,Verbose(Incoming call to ${EXTEN});
exten => _X.,n,Playback(welcome);



-- 

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The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0



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[asterisk-users] billing based on local access number

2010-02-09 Thread umesh maharjan

Hi all,

I am configuring asterisk as a prepaid calling card. I am getting different 
local rate from my ISDN provider e.g  0.002 for landline and 0.13 for mobile 
etc. In this case I thing I have to say my asterisk/a2billing to bill based on 
local access number. so How can I retrieve  called number (eg. 03-6832-1040 and 
0120-272-060 is our ISDN PRI access number) to my asterisk server so i can 
trigger different rates. 

Regards
Umesh
  
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Re: [asterisk-users] Billing applications

2009-10-10 Thread Mindaugas Kezys
You can try free version of MOR Softswitch with billing and routing:
http://www.kolmisoft.com/billing-and-routing/mor-voip-billing-demo/

We rewrote Asterisk CDR completely and yes, it supports transfers.

More info about MOR: http://www.voip-info.org/wiki/view/MOR

Free version supports up to 10 simultaneous calls which is enough for
majority of startups.

You can check our manual to see what functionality is supported:
http://wiki.kolmisoft.com/index.php/MOR_Manual


Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of voip crazy
Sent: 2009 m. spalio 9 d. 18:10
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Billing applications

Hello all,

I want to instal a Billing solution in the same asterisk's box. I have
browse for ast2bill asterisk billing, astercc, and more, bu ti do not
know which will be the best for me.
The only things i need, are,
  - Postpaid and prepaid applications.
  - True CDR. Better that asterisk one, With suport for transfers
  - I do not need support for reseller
  - Billing for Voip, PSTN trunks

I need a light app. I'm not searching a heavy app. with a lots of
modules and applicacions. I need a ligth application for a soho and
its needs.

Any one are using a billing application which fits this needs?
Any clue will be welcomed.

Thanks in advance.

VoipCrazy

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Re: [asterisk-users] Billing applications

2009-10-09 Thread Juan E. Rodríguez
A2billing (Star2Billing, I think, for commercial support) is a good
choice and it's very mature software.
Astercc is very fast and has a very good callshop solution.

Regards,
Juan

voip crazy wrote:
> Hello all,
>
> I want to instal a Billing solution in the same asterisk's box. I have
> browse for ast2bill asterisk billing, astercc, and more, bu ti do not
> know which will be the best for me.
> The only things i need, are,
>   - Postpaid and prepaid applications.
>   - True CDR. Better that asterisk one, With suport for transfers
>   - I do not need support for reseller
>   - Billing for Voip, PSTN trunks
>
> I need a light app. I'm not searching a heavy app. with a lots of
> modules and applicacions. I need a ligth application for a soho and
> its needs.
>
> Any one are using a billing application which fits this needs?
> Any clue will be welcomed.
>
> Thanks in advance.
>
> VoipCrazy
>
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[asterisk-users] Billing applications

2009-10-09 Thread voip crazy
Hello all,

I want to instal a Billing solution in the same asterisk's box. I have
browse for ast2bill asterisk billing, astercc, and more, bu ti do not
know which will be the best for me.
The only things i need, are,
  - Postpaid and prepaid applications.
  - True CDR. Better that asterisk one, With suport for transfers
  - I do not need support for reseller
  - Billing for Voip, PSTN trunks

I need a light app. I'm not searching a heavy app. with a lots of
modules and applicacions. I need a ligth application for a soho and
its needs.

Any one are using a billing application which fits this needs?
Any clue will be welcomed.

Thanks in advance.

VoipCrazy

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[asterisk-users] Billing applications

2009-10-09 Thread voip crazy
Hello all,

I want to instal a Billing solution in the same asterisk's box. I have
browse for ast2bill asterisk billing, astercc, and more, bu ti do not
know which will be the best for me.
The only things i need, are,
   - Postpaid and prepaid applications.
True CDR,

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[asterisk-users] Billing software for 1.6?

2009-03-24 Thread Oguzhan Kayhan
Hi, is there a Asterisk 1.6 billing software that i can use?
Prepaid supporting ones are more acceptable for me right now but,
post-paids are also welcome if they are available.
As i see most of the softwares are designed for 1.2 and 1.4




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Re: [asterisk-users] Billing and Soft Switch.

2009-02-11 Thread Alex Balashov
David @ULC wrote:

> Looking for a Free VOIP Billing and Soft Switch.
> 
> Any suggestions ?

I'm looking to put the milk back in the cow.

If you have the skinny on that, maybe we can swap suggestions.

-- 
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Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] Billing and Soft Switch.

2009-02-11 Thread Steve Howes
On 11 Feb 2009, at 14:22, David @ULC wrote:
> Looking for a Free VOIP Billing and Soft Switch.

And you are asking an Asterisk list... Asterisk? Billing is probably  
best doing a custom job..

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[asterisk-users] Billing and Soft Switch.

2009-02-11 Thread David @ULC
Looking for a Free VOIP Billing and Soft Switch.
Any suggestions ?
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Re: [asterisk-users] Billing and Soft Switch.

2009-02-11 Thread Philipp Kempgen
David @ULC schrieb:
> Looking for a Free VOIP Billing and Soft Switch.

"soft switch" includes back-to-back user agents (Asterisk) I guess?


   Philipp Kempgen

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Re: [asterisk-users] Billing/Call Control engine : AGI scripts/ AstMan API

2007-11-28 Thread Moises Silva
I have found out that executing AGI thru the AMI interface fill better
my needs of control. Take a look

http://bugs.digium.com/view.php?id=11282

Ignore the bug description and read the first note entry, that might
be a better way to get things done.

- Moy

On Nov 27, 2007 10:27 PM, Benjamin Jacob <[EMAIL PROTECTED]> wrote:
> Hello ppl,
>
> Have implemented a really nice Billing engine using AGI scripts. So far
> it works fine, tho haven't yet put it in the torture cell.
>
> The AGI scripts have been written in PHP, using MySQL for the billing
> and profile information.
> The major disadvantages I see using AGI scripts :
> 1. A new process(invocation of PHP scripts) on every new call.
> 2. MySQL connections on every instance of the PHP AGI script. (I am not
> too sure, if connections can be maintained across processes, am no PHP
> guru. I think, if I write in C/C++ can use shared memory for maintaining
> the connection).
>
> So, to overcome these issues, I was thinking of using AstMan APIs along
> with astmanproxy, with the setup being something like this :
>
> Asterisk <-> astmanproxy <-> Billing
> Engine(control/access)
>
> Has anyone ever tried this?
> The one seriously big work with this approach would be to have an FSM
> built into my billing engine, maintaining call states, etc. That seems
> to be quite a daunting task to be done in a short time.
>
> Any ideas anyone?or any similar experiences, in terms of performance,
> scalability, etc. w.r.t both AGI scripts and AstMan API?
>
> TiA
> - Benjamin Jacob.
>
>
>
>
>
>
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Re: [asterisk-users] Billing/Call Control engine : AGI scripts/ AstMan API

2007-11-27 Thread Steve Edwards
On Wed, 28 Nov 2007, Benjamin Jacob wrote:

> The AGI scripts have been written in PHP, using MySQL for the billing 
> and profile information.

> The major disadvantages I see using AGI scripts :

> 1. A new process(invocation of PHP scripts) on every new call.

I write all of my AGIs in C. While PHP is PDF (pretty darn fast), it 
cannot compare to C.

> 2. MySQL connections on every instance of the PHP AGI script. (I am not 
> too sure, if connections can be maintained across processes, am no PHP 
> guru. I think, if I write in C/C++ can use shared memory for maintaining 
> the connection).

When a process exits, all files, sockets, pipes, etc. are closed. You 
cannot maintain a connection across processes.

> Any ideas anyone?or any similar experiences, in terms of performance, 
> scalability, etc. w.r.t both AGI scripts and AstMan API?

I wrote a chat system that handles about 15,000 calls a day, peaking at 
about 100 simultaneous calls. About 90% of the calls execute 6 AGIs, the 
other 10% range from 10 to 50 AGIs. One of the AGIs is even muti-threaded 
-- 1 thread plays "please wait while..." as another thread authorizes 
their credit card. By the time the sound file has finished I know if the 
card is good or not.

AGIs get a bad rap for performance, but I think that is largely due to 
AGIs written in scripting languages. The 1.6gHz Celeron ("whit-whoo") I'm 
working on right now, will execute over 100 AGIs per second. (The 
"null-agi" I just cobbled up reads the AGI environment variables and then 
exits.)

If performance actually becomes a problem, you can re-code your 
application as an Asterisk application. Then you skip the cost of process 
creation and can maintain state and connections.

The same slug used above will execute over 1,000 NOOPs per second.

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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[asterisk-users] Billing/Call Control engine : AGI scripts/ AstMan API

2007-11-27 Thread Benjamin Jacob
Hello ppl,

Have implemented a really nice Billing engine using AGI scripts. So far 
it works fine, tho haven't yet put it in the torture cell.

The AGI scripts have been written in PHP, using MySQL for the billing 
and profile information.
The major disadvantages I see using AGI scripts :
1. A new process(invocation of PHP scripts) on every new call.
2. MySQL connections on every instance of the PHP AGI script. (I am not 
too sure, if connections can be maintained across processes, am no PHP 
guru. I think, if I write in C/C++ can use shared memory for maintaining 
the connection).

So, to overcome these issues, I was thinking of using AstMan APIs along 
with astmanproxy, with the setup being something like this :

Asterisk <-> astmanproxy <-> Billing 
Engine(control/access)

Has anyone ever tried this?
The one seriously big work with this approach would be to have an FSM 
built into my billing engine, maintaining call states, etc. That seems 
to be quite a daunting task to be done in a short time.

Any ideas anyone?or any similar experiences, in terms of performance, 
scalability, etc. w.r.t both AGI scripts and AstMan API?

TiA
- Benjamin Jacob.






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Re: [asterisk-users] Billing Telephone Number (BTN)

2007-02-27 Thread Steve Totaro

Forrest Beck wrote:

I have Asterisk setup with two PRI's one going to my telco and the
other going to a Norstar Meridian system.  The Norstar Meridian is
sending a BTN number that needs to be passed to the Telco.  Is there a
way to pass the BTN as a variable in the dial plan?  Like
CallerID(num)?  What is the variable for BTN if so?


Many Thanks.
Yes, CallerID(num) should work.  I had this issue when setting my 
outbound caller ID to a toll free number and trying to dial a few other 
toll free numbers.  The call could not be completed because they had no 
way to know how to bill the call.  Setting outbound callerID(num) to a 
regular toll number fixed it.


Thanks,
Steve Totaro
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[asterisk-users] Billing Telephone Number (BTN)

2007-02-27 Thread Forrest Beck

I have Asterisk setup with two PRI's one going to my telco and the
other going to a Norstar Meridian system.  The Norstar Meridian is
sending a BTN number that needs to be passed to the Telco.  Is there a
way to pass the BTN as a variable in the dial plan?  Like
CallerID(num)?  What is the variable for BTN if so?


Many Thanks.
--
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
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Re: [asterisk-users] Billing pulses

2007-02-09 Thread Cosmin Prund
The network terminator installed by the Telco in Romania works the same 
way: it has two "analog" outputs and two digital (S0) outputs. I've also 
got a full TDM400 card with 3 FXS and one FXO, but I gave them up gladly 
for a proper ISDN card (I'm using a Diva Eicon Server) - and I don't do 
billing. Sound quality is perfect, there's no echo and I can use all the 
functions of the ISDN card, like the ability to use multiple MSN's, send 
an proper "busy" signal at will, get two calls on the same number at the 
same time.


And now I've got two unused FXS ports in my Asterisk.

Stefano Corsi wrote:
I must clarify my original message. Maybe confusion is due to my poor 
english. So I'll make a list of statements:


- Each ISDN line in Italy can be splitted in two analog lines
- You can use those analog lines as normal analog lines
- I have already invested in analog hardware (my fault of course) for 
both FSX and FXO
- ISDN hardware installed by the telco can, in Italy, be programmed to 
send a "billing pulse".
- I guess this billing pulse is sent on each of the two analog lines 
in which a single ISDN line can be splitted (so there's no risk, I 
guess, for double billing).
- I'm considering if there's a small chance for me to avoid buying 
additional hardware (ISDN cards or gateways) and have an accurate 
billing using those analog lines resulting from splitting an ISDN line.
- To get an accurate billing, I'm wandering if it's possibile to use 
"billing pulse" provided by those analog lines.

- I have full specifications of the "billing pulse" provided:

frequency 
 
12 kHz ± 1%
level 
.. 
200 mVrms on 200
distortion... 
< 5%
pulse duration 
.125 ± 25 ms
pause duration 
> 180 ms
period 
...> 
300 ms


Do you think it's worth considering it?

Rgds
Stefano

> Bill them both.  We are talking about mere BRI's, right:-)  Good 
catch,
> David.  As others noted, billing pulse really applies to analogue 
lines

> only, and ISDN providers should always send status.
>
> Yuan Liu

Thanks, Yuan


But my confusion came from the original post stating the use of ISDN
circuits for this  implementation.  Id ISDN is in fact the circuit of
choice for this app, I agree why wouldn't he simply use the cause codes
for billing purposes.  We have a lot of experience in telecommunications
billing, and have always found cause codes to be more than sufficient
even for weird tiers, and bizarre rounding functions.



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Re: [asterisk-users] Billing pulses

2007-02-08 Thread David Boyd
Hi Stefano,

I have a question, how would you go about using the billing pulses to
generate an invoice/bill.  Also can you provide an ascii drawing of the
layout of the equipment as you intend to use it, they say a picture is
worth a thousand words:)


db




On Thu, 2007-02-08 at 15:13 +0100, Stefano Corsi wrote:
> I must clarify my original message. Maybe 
> confusion is due to my poor english. So I'll make a list of statements:
> 
> - Each ISDN line in Italy can be splitted in two analog lines
> - You can use those analog lines as normal analog lines
> - I have already invested in analog hardware (my 
> fault of course) for both FSX and FXO
> - ISDN hardware installed by the telco can, in 
> Italy, be programmed to send a "billing pulse".
> - I guess this billing pulse is sent on each of 
> the two analog lines in which a single ISDN line 
> can be splitted (so there's no risk, I guess, for double billing).
> - I'm considering if there's a small chance for 
> me to avoid buying additional hardware (ISDN 
> cards or gateways) and have an accurate billing 
> using those analog lines resulting from splitting an ISDN line.
> - To get an accurate billing, I'm wandering if 
> it's possibile to use "billing pulse" provided by those analog lines.
> - I have full specifications of the "billing pulse" provided:
> 
> frequency 
>  
> 12 kHz ± 1%
> level 
> .. 
> 200 mVrms on 200
> distortion... 
> < 5%
> pulse duration 
> .125 ± 25 ms
> pause duration 
> > 180 ms
> period 
> ...> 300 
> ms
> 
> Do you think it's worth considering it?
> 
> Rgds
> Stefano
> 
> > > Bill them both.  We are talking about mere BRI's, right:-)  Good catch,
> > > David.  As others noted, billing pulse really applies to analogue lines
> > > only, and ISDN providers should always send status.
> > >
> > > Yuan Liu
> >
> >Thanks, Yuan
> >
> >
> >But my confusion came from the original post stating the use of ISDN
> >circuits for this  implementation.  Id ISDN is in fact the circuit of
> >choice for this app, I agree why wouldn't he simply use the cause codes
> >for billing purposes.  We have a lot of experience in telecommunications
> >billing, and have always found cause codes to be more than sufficient
> >even for weird tiers, and bizarre rounding functions.
> 
> 

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Re: [asterisk-users] Billing pulses

2007-02-08 Thread George Camilleri
There are two types of ISDN line, Primary Rate Access (PRI) and Basic Rate 
Access (BRI). PRI has 30 (+ 1) channels, BRI has 2 (+1) channels. You are 
talking about BRI which consists of two 64 kbit/s data channels and 1 
signalling channel. In telephony, the two data channels are decoded and used 
as two voice channels. At the end of the decoding process and after passing 
through some interfacing hardware the voice channels end up in an analogue 
device such as a telephone set so that we humans can hear it.


The FXS hardware you invested in can be used for your analogue extensions. 
The FXO hardware is used to interface with analogue telco lines so if you 
want ISDN telco lines you will have to invest in BRI interface cards. 
(Google Asterisk ISDN BRI)


You then ask the telco to include Advice of Charge (AOC) in your ISDN setup. 
The AOC then is included somewhere in the Asterisk CDR, but I don't have 
direct experience of this. You can then get appropriate software to issue 
bills to telephone users.


This is as far as I know and have personal experience of. If anyone can add 
to it it will be appreciated.


George

- Original Message - 
From: "Stefano Corsi" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Asterisk Users Mailing List - Non-Commercial 
Discussion" 

Sent: Thursday, February 08, 2007 3:13 PM
Subject: Re: [asterisk-users] Billing pulses


I must clarify my original message. Maybe confusion is due to my poor 
english. So I'll make a list of statements:


- Each ISDN line in Italy can be splitted in two analog lines
- You can use those analog lines as normal analog lines
- I have already invested in analog hardware (my fault of course) for both 
FSX and FXO
- ISDN hardware installed by the telco can, in Italy, be programmed to 
send a "billing pulse".
- I guess this billing pulse is sent on each of the two analog lines in 
which a single ISDN line can be splitted (so there's no risk, I guess, for 
double billing).
- I'm considering if there's a small chance for me to avoid buying 
additional hardware (ISDN cards or gateways) and have an accurate billing 
using those analog lines resulting from splitting an ISDN line.
- To get an accurate billing, I'm wandering if it's possibile to use 
"billing pulse" provided by those analog lines.

- I have full specifications of the "billing pulse" provided:

frequency 
 12 
kHz ± 1%
level 
.. 
200 mVrms on 200
distortion... 
< 5%
pulse duration 
.125 ± 25 ms
pause duration 
> 180 ms
period 
...> 
300 ms


Do you think it's worth considering it?

Rgds
Stefano


> Bill them both.  We are talking about mere BRI's, right:-)  Good catch,
> David.  As others noted, billing pulse really applies to analogue lines
> only, and ISDN providers should always send status.
>
> Yuan Liu

Thanks, Yuan


But my confusion came from the original post stating the use of ISDN
circuits for this  implementation.  Id ISDN is in fact the circuit of
choice for this app, I agree why wouldn't he simply use the cause codes
for billing purposes.  We have a lot of experience in telecommunications
billing, and have always found cause codes to be more than sufficient
even for weird tiers, and bizarre rounding functions.



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Re: [asterisk-users] Billing pulses

2007-02-08 Thread Stefano Corsi
I must clarify my original message. Maybe 
confusion is due to my poor english. So I'll make a list of statements:


- Each ISDN line in Italy can be splitted in two analog lines
- You can use those analog lines as normal analog lines
- I have already invested in analog hardware (my 
fault of course) for both FSX and FXO
- ISDN hardware installed by the telco can, in 
Italy, be programmed to send a "billing pulse".
- I guess this billing pulse is sent on each of 
the two analog lines in which a single ISDN line 
can be splitted (so there's no risk, I guess, for double billing).
- I'm considering if there's a small chance for 
me to avoid buying additional hardware (ISDN 
cards or gateways) and have an accurate billing 
using those analog lines resulting from splitting an ISDN line.
- To get an accurate billing, I'm wandering if 
it's possibile to use "billing pulse" provided by those analog lines.

- I have full specifications of the "billing pulse" provided:

frequency 
 
12 kHz ± 1%
level 
.. 
200 mVrms on 200
distortion... 
< 5%
pulse duration 
.125 ± 25 ms
pause duration 
> 180 ms
period 
...> 300 ms


Do you think it's worth considering it?

Rgds
Stefano


> Bill them both.  We are talking about mere BRI's, right:-)  Good catch,
> David.  As others noted, billing pulse really applies to analogue lines
> only, and ISDN providers should always send status.
>
> Yuan Liu

Thanks, Yuan


But my confusion came from the original post stating the use of ISDN
circuits for this  implementation.  Id ISDN is in fact the circuit of
choice for this app, I agree why wouldn't he simply use the cause codes
for billing purposes.  We have a lot of experience in telecommunications
billing, and have always found cause codes to be more than sufficient
even for weird tiers, and bizarre rounding functions.



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Re: [asterisk-users] Billing pulses

2007-02-07 Thread David Boyd
On Wed, 2007-02-07 at 14:49 -0800, Yuan LIU wrote:
> >From: David Boyd <[EMAIL PROTECTED]>
> >Date: Wed, 07 Feb 2007 15:24:04 -0500
> >
> >On Wed, 2007-02-07 at 10:14 -0800, Yuan LIU wrote:
> > > From:  Jorge Mendoza <[EMAIL PROTECTED]>
> > > >Funny that a digital line have a analogue pulse.
> > > >Normally the billing pulse is used on payphones. IMO you only need
> > > >the answer supervision to trigger your own billing system.
> > > >
> > > >Jorge Mendoza
> > > >
> > > >Stefano Corsi wrote:
> > > >>Hello,
> > > >>
> > > >>I've discovered that in Italy ISDN lines can be programmed to
> > > >>generate a "billing pulse" every n seconds (it dipends from the
> > > >>pricebook). The pulse has these figures:
> > >
> > >
> > > Whatever reason, if telco provides them, there's a good chance
> > > that some ISDN interface cards can use them.  (Just googled to confirm
> > > that some non-Digium cards can be used in Asterisk.)  This doesn't
> > > mean that Asterisk can use them.  So you may need significant
> > > programming to get going.
> > >
> > > If they are truly analogue pulses, it could be cheaper to produce a
> > > little dedicated circuit to feed an AGI or something.
> > >
> > > Yuan Liu
> > > ...
> >How would you be able to determine which call was being billed for if
> >the pulse is sent down the wire on an ISDN circuit with multiple
> >channels in use?
> >
> >db
> 
> Bill them both.  We are talking about mere BRI's, right:-)  Good catch, 
> David.  As others noted, billing pulse really applies to analogue lines 
> only, and ISDN providers should always send status.
> 
> Yuan Liu
> 
> 
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Thanks, Yuan


But my confusion came from the original post stating the use of ISDN
circuits for this  implementation.  Id ISDN is in fact the circuit of
choice for this app, I agree why wouldn't he simply use the cause codes
for billing purposes.  We have a lot of experience in telecommunications
billing, and have always found cause codes to be more than sufficient
even for weird tiers, and bizarre rounding functions.

db

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Re: [asterisk-users] Billing pulses

2007-02-07 Thread Yuan LIU

From: David Boyd <[EMAIL PROTECTED]>
Date: Wed, 07 Feb 2007 15:24:04 -0500

On Wed, 2007-02-07 at 10:14 -0800, Yuan LIU wrote:
> From:  Jorge Mendoza <[EMAIL PROTECTED]>
> >Funny that a digital line have a analogue pulse.
> >Normally the billing pulse is used on payphones. IMO you only need
> >the answer supervision to trigger your own billing system.
> >
> >Jorge Mendoza
> >
> >Stefano Corsi wrote:
> >>Hello,
> >>
> >>I've discovered that in Italy ISDN lines can be programmed to
> >>generate a "billing pulse" every n seconds (it dipends from the
> >>pricebook). The pulse has these figures:
>
>
> Whatever reason, if telco provides them, there's a good chance
> that some ISDN interface cards can use them.  (Just googled to confirm
> that some non-Digium cards can be used in Asterisk.)  This doesn't
> mean that Asterisk can use them.  So you may need significant
> programming to get going.
>
> If they are truly analogue pulses, it could be cheaper to produce a
> little dedicated circuit to feed an AGI or something.
>
> Yuan Liu
> ...
How would you be able to determine which call was being billed for if
the pulse is sent down the wire on an ISDN circuit with multiple
channels in use?

db


Bill them both.  We are talking about mere BRI's, right:-)  Good catch, 
David.  As others noted, billing pulse really applies to analogue lines 
only, and ISDN providers should always send status.


Yuan Liu


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Re: [asterisk-users] Billing pulses

2007-02-07 Thread Patrick
On Wed, 2007-02-07 at 21:00 +0100, George Camilleri wrote:
> Hi
> 
> "Billing Pulses" only apply to analogue lines. You need special hardware in 
> the PBX interface to detect them and pass them on to the Billing software. 
> To my knowlege there is no Asterisk compatible hardware that does this.

ISDN has AOC (advice of charge) and does not require special hardware.
Iirc a while back there was some development of AOC support for Asterisk
but I am not aware of the current status.

Regards,
Patrick

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Re: [asterisk-users] Billing pulses

2007-02-07 Thread David Boyd
On Wed, 2007-02-07 at 10:14 -0800, Yuan LIU wrote:
> From:  Jorge Mendoza <[EMAIL PROTECTED]>
> >Funny that a digital line have a analogue pulse.
> >Normally the billing pulse is used on payphones. IMO you only need 
> >the answer supervision to trigger your own billing system.
> >
> >Jorge Mendoza
> >
> >Stefano Corsi wrote:
> >>Hello,
> >>
> >>I've discovered that in Italy ISDN lines can be programmed to 
> >>generate a "billing pulse" every n seconds (it dipends from the 
> >>pricebook). The pulse has these figures:
> 
> 
> Whatever reason, if telco provides them, there's a good chance
> that some ISDN interface cards can use them.  (Just googled to confirm
> that some non-Digium cards can be used in Asterisk.)  This doesn't
> mean that Asterisk can use them.  So you may need significant
> programming to get going.
> 
> If they are truly analogue pulses, it could be cheaper to produce a
> little dedicated circuit to feed an AGI or something.
> 
> 
> Yuan Liu
> 
> >>frequency 
> >> 
> >>12 kHz ?1%
> >>
> >>level 
> >>.. 
> >>200 mVrms on 200
> >>
> >>distortion...
> >> 
> >>< 5%
> >>pulse duration 
> >>.125 ?
> >>25 ms
> >>pause duration 
> >>> 
> >>180 ms
> >>period 
> >>...> 
> >>300 ms
> >>
> >>Does someone know if these values can be used somehow to get an 
> >>accurate billing using asterisk with these lines? Could be a matter 
> >>of configuration or programming?
> >>
> >>Thanks
> >>Stefano
> 
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How would you be able to determine which call was being billed for if
the pulse is sent down the wire on an ISDN circuit with multiple
channels in use?

db



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Re: [asterisk-users] Billing pulses

2007-02-07 Thread George Camilleri

Hi

"Billing Pulses" only apply to analogue lines. You need special hardware in 
the PBX interface to detect them and pass them on to the Billing software. 
To my knowlege there is no Asterisk compatible hardware that does this.


George
- Original Message - 
From: "Stefano Corsi" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, February 07, 2007 4:04 PM
Subject: [asterisk-users] Billing pulses



Hello,

I've discovered that in Italy ISDN lines can be programmed to generate a 
"billing pulse" every n seconds (it dipends from the pricebook). The pulse 
has these figures:


frequency 
 12 
kHz ± 1%


level 
.. 
200 mVrms on 200


distortion... 
< 5%
pulse duration 
.125 ± 25 ms
pause duration 
> 180 ms
period 
...> 
300 ms


Does someone know if these values can be used somehow to get an accurate 
billing using asterisk with these lines? Could be a matter of 
configuration or programming?


Thanks
Stefano
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Version: 7.5.432 / Virus Database: 268.17.29/673 - Release Date: 2/6/2007 
5:52 PM





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Re: [asterisk-users] Billing pulses

2007-02-07 Thread Yuan LIU

From:  Jorge Mendoza <[EMAIL PROTECTED]>>Funny that a digital line have a analogue pulse.>Normally the billing pulse is used on payphones. IMO you only need >the answer supervision to trigger your own billing system.>>Jorge Mendoza>>Stefano Corsi wrote:>>Hello,I've discovered that in Italy ISDN lines can be programmed to >>generate a "billing pulse" every n seconds (it dipends from the >>pricebook). The pulse has these figures:
Whatever reason, if telco provides them, there's a good chance that some ISDN interface cards can use them.  (Just googled to confirm that some non-Digium cards can be used in Asterisk.)  This doesn't mean that Asterisk can use them.  So you may need significant programming to get going.
If they are truly analogue pulses, it could be cheaper to produce a little dedicated circuit to feed an AGI or something.
Yuan Liu
>>frequency >> >>12 kHz ?1%level >>.. >>200 mVrms on 200distortion... >>< 5%>>pulse duration >>.125 ?>>25 ms>>pause duration >>> >>180 ms>>period >>...> >>300 msDoes someone know if these values 
can be used somehow to get an >>accurate billing using asterisk with these lines? Could be a matter >>of configuration or programming?Thanks>>Stefano

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Re: [asterisk-users] Billing pulses

2007-02-07 Thread Jorge Mendoza

All digital lines (BRI or PRI) provides answer and release supervision.
The drivers will send to * this information, and this information will
be registered into the CDR automatically. You only need setup your
billing system.

As said before you do not need to intercept the billing pulse.

Jorge Mendoza

Stefano Corsi wrote:

At 16.22 07/02/2007, you wrote:

Funny that a digital line have a analogue pulse.
Normally the billing pulse is used on payphones. IMO you only need 
the answer supervision to trigger your own billing system.


Yes, it's strange. But I find no mention on answer supervision in the 
NT1Plus manual (NT1Plus is the hardware device the Telco installs when 
you ask for an ISDN line). Where should I ask for answer supervision? 
The Telco? That sounds very difficult in Italy... they have no 
technical call centers. Almost only sales.


But if the line should provide those "analog" billing pulses... do you 
think could be possible to intercept them?


Rgds
Stefano 


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Re: [asterisk-users] Billing pulses

2007-02-07 Thread Stefano Corsi

At 16.22 07/02/2007, you wrote:

Funny that a digital line have a analogue pulse.
Normally the billing pulse is used on payphones. IMO you only need 
the answer supervision to trigger your own billing system.


Yes, it's strange. But I find no mention on answer supervision in the 
NT1Plus manual (NT1Plus is the hardware device the Telco installs 
when you ask for an ISDN line). Where should I ask for answer 
supervision? The Telco? That sounds very difficult in Italy... they 
have no technical call centers. Almost only sales.


But if the line should provide those "analog" billing pulses... do 
you think could be possible to intercept them?


Rgds
Stefano 


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Re: [asterisk-users] Billing pulses

2007-02-07 Thread Jorge Mendoza

Funny that a digital line have a analogue pulse.
Normally the billing pulse is used on payphones. IMO you only need the 
answer supervision to trigger your own billing system.


Jorge Mendoza

Stefano Corsi wrote:

Hello,

I've discovered that in Italy ISDN lines can be programmed to generate 
a "billing pulse" every n seconds (it dipends from the pricebook). The 
pulse has these figures:


frequency 
 
12 kHz ± 1%


level 
.. 
200 mVrms on 200


distortion... 
< 5%
pulse duration 
.125 ± 25 ms
pause duration 
> 180 ms
period 
...> 
300 ms


Does someone know if these values can be used somehow to get an 
accurate billing using asterisk with these lines? Could be a matter of 
configuration or programming?


Thanks
Stefano
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[asterisk-users] Billing pulses

2007-02-07 Thread Stefano Corsi

Hello,

I've discovered that in Italy ISDN lines can be 
programmed to generate a "billing pulse" every n 
seconds (it dipends from the pricebook). The pulse has these figures:


frequency 
 
12 kHz ± 1%


level 
.. 
200 mVrms on 200


distortion... 
< 5%
pulse duration 
.125 ± 25 ms
pause duration 
> 180 ms
period 
...> 300 ms


Does someone know if these values can be used 
somehow to get an accurate billing using asterisk 
with these lines? Could be a matter of configuration or programming?


Thanks
Stefano 


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Re: [asterisk-users] Billing solution

2006-12-27 Thread kjcsb




Can anyone recommend a call accounting solution with rating for post
paid billing that works well with asterisk using the account code or
any other info from the CDR?

I don't want the billing software to any phone calls for me, therefore
any solution that modifies my extensions.conf is out, nor does it have
to allow for customers the ability to log in to check their
usage/balances.
I have looked at astbill but it looks to be way overcomplicated for
what I want, as well as it requires realtime.
Thank you


CDRTool does call rating

Cameron
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Re: [asterisk-users] Billing solution

2006-12-20 Thread C F

Giedrius, did you read my post?
Doesn't seem  so as I ask for solution that does NOT require to modify
my dialplan.

On 12/20/06, Giedrius Augys <[EMAIL PROTECTED]> wrote:


2006/12/20, C F <[EMAIL PROTECTED]>:
>
> Well I did:
> astpp
> http://www.astpp.org/
>
>
> On 12/20/06, Vicky <[EMAIL PROTECTED]> wrote:
> > I am looking for exactly same kind of billing stuff but i dont think you
> > will get it without letting ur billing program make some changes in
asterisk
> > .
> >
> >
> > On 20/12/06, Carlos Rojas <[EMAIL PROTECTED]> wrote:
> > > a2billing
> > >
> > > Is very good
> > >
> > >
> > >
> > > On 12/19/06, Giedrius Augys < [EMAIL PROTECTED]> wrote:
> > > >
> > > >
> > > >
> > > > 2006/12/19, C F <[EMAIL PROTECTED]>:
> > > >
> > > > > Can anyone recommend a call accounting solution with rating for
post
> > > > > paid billing that works well with asterisk using the account code
or
> > > > > any other info from the CDR?
> > > > >
> > > > > I don't want the billing software to any phone calls for me,
therefore
> > > > > any solution that modifies my extensions.conf is out, nor does it
have
> > > > > to allow for customers the ability to log in to check their
> > > > > usage/balances.
> > > > > I have looked at astbill but it looks to be way overcomplicated
for
> > > > > what I want, as well as it requires realtime.
> > > > > Thank you
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> > > > >
> > > >
> > > >
> > > > Mor and Mcc
> > > >
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As I said , MCC would the best solution for you ( http://www.kolmisoft.com/
). You will compile app mcc2 , and you use this app as Dial command in
extensions.conf .

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Re: [asterisk-users] Billing solution

2006-12-19 Thread Andrew Joakimsen

Astpp runs two cron jobs, it writes the rate to the CDR, does it by the
accountcode.

On 12/18/06, C F <[EMAIL PROTECTED]> wrote:


Can anyone recommend a call accounting solution with rating for post
paid billing that works well with asterisk using the account code or
any other info from the CDR?

I don't want the billing software to any phone calls for me, therefore
any solution that modifies my extensions.conf is out, nor does it have
to allow for customers the ability to log in to check their
usage/balances.
I have looked at astbill but it looks to be way overcomplicated for
what I want, as well as it requires realtime.
Thank you
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Re: [asterisk-users] Billing solution

2006-12-19 Thread Giedrius Augys

2006/12/20, C F <[EMAIL PROTECTED]>:


Well I did:
astpp
http://www.astpp.org/


On 12/20/06, Vicky <[EMAIL PROTECTED]> wrote:
> I am looking for exactly same kind of billing stuff but i dont think you
> will get it without letting ur billing program make some changes in
asterisk
> .
>
>
> On 20/12/06, Carlos Rojas <[EMAIL PROTECTED]> wrote:
> > a2billing
> >
> > Is very good
> >
> >
> >
> > On 12/19/06, Giedrius Augys < [EMAIL PROTECTED]> wrote:
> > >
> > >
> > >
> > > 2006/12/19, C F <[EMAIL PROTECTED]>:
> > >
> > > > Can anyone recommend a call accounting solution with rating for
post
> > > > paid billing that works well with asterisk using the account code
or
> > > > any other info from the CDR?
> > > >
> > > > I don't want the billing software to any phone calls for me,
therefore
> > > > any solution that modifies my extensions.conf is out, nor does it
have
> > > > to allow for customers the ability to log in to check their
> > > > usage/balances.
> > > > I have looked at astbill but it looks to be way overcomplicated
for
> > > > what I want, as well as it requires realtime.
> > > > Thank you
> > > > ___
> > > > --Bandwidth and Colocation provided by Easynews.com --
> > > >
> > > > asterisk-users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > >
> > > Mor and Mcc
> > >
> > > ___
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> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > >
> >
> >
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> >
>
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As I said , MCC would the best solution for you (
http://www.kolmisoft.com/). You will compile app mcc2 , and you use
this app as Dial command in
extensions.conf .
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Re: [asterisk-users] Billing solution

2006-12-19 Thread C F

Well I did:
astpp
http://www.astpp.org/


On 12/20/06, Vicky <[EMAIL PROTECTED]> wrote:

I am looking for exactly same kind of billing stuff but i dont think you
will get it without letting ur billing program make some changes in asterisk
.


On 20/12/06, Carlos Rojas <[EMAIL PROTECTED]> wrote:
> a2billing
>
> Is very good
>
>
>
> On 12/19/06, Giedrius Augys < [EMAIL PROTECTED]> wrote:
> >
> >
> >
> > 2006/12/19, C F <[EMAIL PROTECTED]>:
> >
> > > Can anyone recommend a call accounting solution with rating for post
> > > paid billing that works well with asterisk using the account code or
> > > any other info from the CDR?
> > >
> > > I don't want the billing software to any phone calls for me, therefore
> > > any solution that modifies my extensions.conf is out, nor does it have
> > > to allow for customers the ability to log in to check their
> > > usage/balances.
> > > I have looked at astbill but it looks to be way overcomplicated for
> > > what I want, as well as it requires realtime.
> > > Thank you
> > > ___
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >
http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> > Mor and Mcc
> >
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
>
>
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>
>


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Re: [asterisk-users] Billing solution

2006-12-19 Thread Vicky

I am looking for exactly same kind of billing stuff but i dont think you
will get it without letting ur billing program make some changes in asterisk
.

On 20/12/06, Carlos Rojas <[EMAIL PROTECTED]> wrote:


a2billing

Is very good

On 12/19/06, Giedrius Augys <[EMAIL PROTECTED]> wrote:
>
>
>
> 2006/12/19, C F <[EMAIL PROTECTED]>:
> >
> > Can anyone recommend a call accounting solution with rating for post
> > paid billing that works well with asterisk using the account code or
> > any other info from the CDR?
> >
> > I don't want the billing software to any phone calls for me, therefore
> >
> > any solution that modifies my extensions.conf is out, nor does it have
> > to allow for customers the ability to log in to check their
> > usage/balances.
> > I have looked at astbill but it looks to be way overcomplicated for
> > what I want, as well as it requires realtime.
> > Thank you
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> Mor and Mcc
>
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>
>

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Re: [asterisk-users] Billing solution

2006-12-19 Thread Carlos Rojas

a2billing

Is very good

On 12/19/06, Giedrius Augys <[EMAIL PROTECTED]> wrote:




2006/12/19, C F <[EMAIL PROTECTED]>:
>
> Can anyone recommend a call accounting solution with rating for post
> paid billing that works well with asterisk using the account code or
> any other info from the CDR?
>
> I don't want the billing software to any phone calls for me, therefore
> any solution that modifies my extensions.conf is out, nor does it have
> to allow for customers the ability to log in to check their
> usage/balances.
> I have looked at astbill but it looks to be way overcomplicated for
> what I want, as well as it requires realtime.
> Thank you
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>


Mor and Mcc

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Re: [asterisk-users] Billing solution

2006-12-19 Thread Giedrius Augys

2006/12/19, C F <[EMAIL PROTECTED]>:


Can anyone recommend a call accounting solution with rating for post
paid billing that works well with asterisk using the account code or
any other info from the CDR?

I don't want the billing software to any phone calls for me, therefore
any solution that modifies my extensions.conf is out, nor does it have
to allow for customers the ability to log in to check their
usage/balances.
I have looked at astbill but it looks to be way overcomplicated for
what I want, as well as it requires realtime.
Thank you
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Mor and Mcc
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[asterisk-users] Billing solution

2006-12-18 Thread C F

Can anyone recommend a call accounting solution with rating for post
paid billing that works well with asterisk using the account code or
any other info from the CDR?

I don't want the billing software to any phone calls for me, therefore
any solution that modifies my extensions.conf is out, nor does it have
to allow for customers the ability to log in to check their
usage/balances.
I have looked at astbill but it looks to be way overcomplicated for
what I want, as well as it requires realtime.
Thank you
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Re: [asterisk-users] Billing Software

2006-12-03 Thread lists
In reality, this is the one I've found that has exactly what our client
needs. However, it seems to be a closed system so we are evaluating it
further.

AstBill and MOR don't seem to have the feature to offer referral "credits"
out-of-the-box. Maybe we missed something?

Thanks,
Daniel


-Original Message-
From: "Guillermo Salas M." <[EMAIL PROTECTED]>
Sent: Sun, December 3, 2006 11:43 am
To: [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Billing Software

Have you found any solution ?


I'm looking for the same product. Seems like astbill [1] and MOR [2] can
manage reseller accounts.

Regards,


[1] www.astbill.com
[2] www.kolmisoft.com


On Thu, 2006-11-30 at 11:29 -0500, [EMAIL PROTECTED] wrote:
> We are looking for an offline billing solution. We have a couple of
> particular requirements:
>
> 1) Since it's offline, we need to be able to import the CDR.
> 2) A way to support account credits based on referrals. Meaning, that if a
> member refers a new account, that member would get a free month of
> service, or similar type credits.
> 3) Generate invoices in either HTML or PDF format so they can be printed
> or emailed to the actual customers.
>
> Does anyone know of a package that supports this? Would prefer open source.
>
> Thanks,
> Daniel
>
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-- 
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Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
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Re: [asterisk-users] Billing Software

2006-12-01 Thread Dovid B

Try looking at enswitch. It is a paid solution.

- Original Message - 
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Thursday, November 30, 2006 6:29 PM
Subject: [asterisk-users] Billing Software


We are looking for an offline billing solution. We have a couple of
particular requirements:

1) Since it's offline, we need to be able to import the CDR.
2) A way to support account credits based on referrals. Meaning, that if a
member refers a new account, that member would get a free month of
service, or similar type credits.
3) Generate invoices in either HTML or PDF format so they can be printed
or emailed to the actual customers.

Does anyone know of a package that supports this? Would prefer open source.

Thanks,
Daniel

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[asterisk-users] Billing Software

2006-11-30 Thread lists
We are looking for an offline billing solution. We have a couple of
particular requirements:

1) Since it's offline, we need to be able to import the CDR.
2) A way to support account credits based on referrals. Meaning, that if a
member refers a new account, that member would get a free month of
service, or similar type credits.
3) Generate invoices in either HTML or PDF format so they can be printed
or emailed to the actual customers.

Does anyone know of a package that supports this? Would prefer open source.

Thanks,
Daniel

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Re: [asterisk-users] Billing software with reseller accounts

2006-11-29 Thread Dovid B
I have been using Enswitch. Has some bugs but over all works great. It's not 
open source but worth the money.


- Original Message - 
From: "Guillermo Salas M." <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, November 29, 2006 3:12 AM
Subject: [asterisk-users] Billing software with reseller accounts



Hello,


Can you recommend a good billing software for asterisk that supports
reseller accounts? Will be better if it haves opensource licence.

Best regards,

--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
  http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
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[asterisk-users] Billing software with reseller accounts

2006-11-28 Thread Guillermo Salas M.
Hello,


Can you recommend a good billing software for asterisk that supports
reseller accounts? Will be better if it haves opensource licence.

Best regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting

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[asterisk-users] billing

2006-11-06 Thread Khaled








 

Dear

 

How can I charge the incoming call to the destination call
,using a2billing 

I used to make setaccount but it didn’t work such a
loopback detected 

 

 

 

Please help 






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Re: [asterisk-users] billing

2006-11-03 Thread Don MacArthur
Jeremy, how about including a link to the appropriate forum?  (I know
you won't make me ask a second time...)

On Fri, 2006-11-03 at 10:57 -0500, Jeremy McNamara wrote:
> Khaled wrote:
> > Dear
> > 
> >  
> > 
> > How can I charge the incoming call to the destination call ,using a2billing
> > 
> > I used to make setaccount but it didn’t work such a loopback detected
> 
> 
> 
> 
> This is not the a2billing support forum.
> 
> 
> Is there an echo in here?
> 
> 
> 
> Jeremy McNamara
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Re: [asterisk-users] billing

2006-11-03 Thread Jeremy McNamara

Khaled wrote:

Dear

 


How can I charge the incoming call to the destination call ,using a2billing

I used to make setaccount but it didn’t work such a loopback detected





This is not the a2billing support forum.


Is there an echo in here?



Jeremy McNamara
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Re: [asterisk-users] billing

2006-11-03 Thread Doug Lytle

Khaled wrote:


Dear

How can I charge the incoming call to the destination call ,using 
a2billing


I used to make setaccount but it didn’t work such a loopback detected



You should ask this on the a2billing forums.

http://forum.asterisk2billing.org/
/
Doug/
/

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[asterisk-users] billing

2006-11-03 Thread Khaled








Dear

 

How can I charge the incoming call to the destination call ,using
a2billing 

I used to make setaccount but it didn’t work such a
loopback detected 

 

 

 

Please help 






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Re: [asterisk-users] Billing Solution ?

2006-10-30 Thread Noc Phibee

Thanks all for your answer ;=) i start test this week a2billing



Noc Phibee a écrit :

Hi

what is the best billing solution for Asterisk ?

With WWW manager interface for user can see the real invoice...

Thanks bye
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Re: [asterisk-users] Billing Solution ?

2006-10-30 Thread Guillermo Salas M.
On Mon, 2006-10-30 at 18:31 +0100, Noc Phibee wrote:
> Hi
> 
> what is the best billing solution for Asterisk ?
> 
> With WWW manager interface for user can see the real invoice...
> 


I'm using a2billing and works like a charm for me :)


Regards,

> Thanks bye
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-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular  : +593 9 985 5138
e-mail   : [EMAIL PROTECTED]
www  : http://www.manta.telconet.net
   http://www.telcocarrier.net

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting

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Re: [asterisk-users] Billing Solution ?

2006-10-30 Thread R.R. Libera

Try www.asterisk2billing.org




Noc Phibee escribió:

Hi

what is the best billing solution for Asterisk ?

With WWW manager interface for user can see the real invoice...

Thanks bye
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[asterisk-users] Billing Solution ?

2006-10-30 Thread Noc Phibee

Hi

what is the best billing solution for Asterisk ?

With WWW manager interface for user can see the real invoice...

Thanks bye
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[asterisk-users] Billing

2006-10-10 Thread Khaled Chehab








Dear 

I am using
a2billing accounting software, how can I charge on the destination target not
at the caller side

Ex: if user
A have 10$  and user B have 10$ ,and the onnet call charge cost 1$ 

When user A
call user B for 1 minute ,user A amount remains 10$ and user B amount be 9$

 

Regards  

 






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[Asterisk-Users] billing

2006-06-23 Thread Khaled Chehab








 

 

I am using
trixbox ,please any ont knows how to confiure billing on it,

I want to
make a billing ,I created an account at http://x.X.X.X/a2billing but it does work 

 






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This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.

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[Asterisk-Users] Billing when forwarding incomming calls from SIP phone

2006-05-09 Thread Garth van Sittert

Hi All

Does anyone know how to track calls that come in from the PSTN, go to 
the user SIP phone and then get forwarded from the user SIP phone back 
out to PSTN (typically the users mobile number)?
It is a headache trying to bill the users who do this.  There is only 
the src PSTN number and dst forwarded number in the CDR.  No SIP details 
at all.  How do we track this?

Billing for standard calls from the SIP user to the PSTN is fine.

Regards
Garth

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RE: [Asterisk-Users] billing realtime

2006-05-03 Thread Senad Jordanovic
[EMAIL PROTECTED] wrote:
>  Hi Senad
> 
> i looking for same thing, that is consider absolutetimeout as a
> timer, everytime is  near t zero, 3 secs for example, renew it,
> reacalculate real credit, and start again until some of the parties
> hangup.
> 
> The problem is how to iterate in asterisk config, or in deadagi,
> you will need some time values from asterisk anyway, CDR{billsec} and
> CDR{duration}, because i think we have to give this control to
> asterisk, he really knows the timing of calls. Now the problem number
> two. Asterisk set those values above, when the call is completely
> finished, i have tried with deadagi in php whit sleep function,
> nothing, the values of the varialbles are set after hangup extension,
> after deadagi final execution.

If I understood well, when each call is made u give him duration time 
based on the billing.
Its wrong direction at start. The only possible solution is in the 
asterisk. You need global variable with total time for all channels, 
then you need the timer.
Timer can be one by each channel, and each channel timer decrements same 
global time variable when it becomes a zero or less terminate all active 
channels for that account.

The other way would be to have one timer who decrements global time 
variable based on number of active channels. Timer is inactive when 
there is no active channels for account.
To explain this, if timer decrement cycle is  n second  then  he should 
decrement global remained time variable  ACCOUNT_TIME = ACCOUNT_TIME- (n 
active channels at the moment) x (timer cycle in seconds).
Then check condition ACCOUNT_TIME <= 0 if true hangup all active 
channels for that account.
Then check condition (n active channels for account == 0) if true stop 
the timer.
The "n active channels" should be checked on asterisk.

If you create account time variable when first channel of account 
becomes active  like AV_{some id} and timer who will process this 
remaining time.
Then on each new channel for that account you just increment other 
variable NAC_{some id} or decrement.
The best is that this variables be asterisk variables (global).

We have not tried above, so be my guest if you have free time :)



Senad
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Re: [Asterisk-Users] billing realtime

2006-05-02 Thread Dovid Bender
How about AstRTB ? Asterisk Real Time Billing
--- Thameem Ansari <[EMAIL PROTECTED]> wrote:

> Hello All,
> 
> I had the same question when I was writing my own
> billing software in java.
> Here is what I am doing to track multiple calls at a
> time from the prepaid
> account.
> 
> 1. Keep on db table for balance and
> reserver_balance.
> 2. First call coming to agi, check the balance - Sum
> of all the
> reserve_balance of that account code.
> 3. Check the destination and allowed minutes for
> that balance amount from
> step 2.
> 4. Reserve balance table will contain destination,
> amount, reserved secs
> columns
> 5. If the avaialable balance is <= 0 then announce
> not enough credit and
> hangup.
> 6. If the available balance is > 0 but seconds
> allowed to talk is less than
> reserved secs (see step 8 for more details about
> what this is)
>then set absolutetimeout for those seconds.
> 7. Otherwise the allowed seconds is more than the
> allowed seconds, set
> absolute time out for the reserved seconds and make
> the call.
> 8. Reserved secs is a custom constant seconds, say
> you can reserve fund for
> 3 minutes (180 seconds). if the account has balance
> for only 2 minutes (120
> seconds) then the absolute time out will be 120
> seconds.
> 9. Once the channel status changed to reserved,
> insert an record to
> reserve_balance table with uniqueid, accountcode,
> amount, reserved_secs
> information.
> 
> The above steps will handle one call so far
> now...and lets see how the dial
> plan should be,
> 10. In your dial plan, add an AbsoluteTimeout
> extension "T" and call another
> AGI script which will just to reset the absolute
> timeout.
> 11. When the particular timeout is reached asterisk
> will transfer the call
> to 'T" extension which will in turn call another
> agi.
> 12. The agi will receive all the information about
> the channel including
> uniqueid, repeat the steps 2- 7 (except dial) and
> reset the abstimeout and
> this process will repeat until the channel hangup.
> 13. Once the channel hangup, you can either use
> Manager to receive the cdr
> event or you can set "h" extension (not reliable and
> not recommended) to
> calculate the real balance and update the balance
> table. Once you update the
> balance table, remove the record from
> reserve_balance table for the
> uniqueid, channel and accountcode. (these three are
> enough to find out the
> entry in that table).
> 
> Now lets take the scenario for  second call when the
> first call was  active,
> 
> 14. When the second call comes in, start from step
> 2. In step 2, we are
> doing finalBalance = Balance - Sum(reserve_balance)
> for that account code.
> If there is already a call on this accountcode, then
> this table will have
> one entry and the reserved amount. Get the
> finalBalance by subtracting the
> amounts. Follow step 3 and allow or deny the caller.
> The above said solution is very stable and doesn't
> overflow the memory or
> session and not using any threads. The only
> restriction here is, if we have
> the scenario,
> 
> Call -1
> balance = $0.10
> destination= 1 (which is US)
> rate = $0.02 per minute
> reserveSecs  = 10 minutes (600secs)
> finalBalance = $0.10 - $0 (consider this is first
> call and no entry in
> reserve_balance table) = $0.10
> allowedMints = $0.10/$0.02 = 5 minutes  = 300
> seconds.
> AbsoluteTimeout = 300 seconds (this is less than the
> default reserveSecs so
> set this as abstimeout)
> 
> Call -2
> balance = $0.10
> destination= 1 (which is US)
> rate = $0.02 per minute
> reserveSecs  = 10 minutes (600secs)
> finalBalance = $0.10 - $0.10 (consider this is
> second call and already an
> entry in reserve_balance table) = $0.0
> allowedMints = 0 seconds.
> announce the denied ivr.
> 
> So, the reserveSecs is critical to avoid how much
> threshold amount the
> caller should have to make two calls. If they have
> $10 in their account as
> per the above two algorithms, they can make as many
> simultaneous calls.
> 
> I hope this solves most of your problems. I looked
> at ASTCC, A2Billing etc
> and they are not doing this way and not know whether
> they work properly. But
> this works for me. Shoot me your questions if you
> have one.
> 
> I am developing my own billing and routing app (in
> java) and I need a name
> for that.. guys pls suggest one.. i may put that in
> sourceforge if i feel
> confident.
> 
> Thanks,
> Thameem
> 
> 
> On 4/27/06, JP Carballo <[EMAIL PROTECTED]> wrote:
> >
> > Dovid Bender wrote:
> >
> > >A while back some one posted some code that he
> used
> > >that took out the flag in astcc that kept track
> if
> > >there was a call in progress for that pin or not.
> Dont
> > >know if it wil work for real time though.
> > >
> > >Dovid
> > >
> > >
> > I don't know if you were pertaining to what I
> posted in the message
> > "ASTCC: How to reset "in-use" flag automatically
> ?".
> > The setinuse() routine already exists in ASTCC.
> > One simply has to use that routine to disable th

Re: [Asterisk-Users] billing realtime

2006-05-01 Thread Thameem Ansari
Hello All,

I had the same question when I was writing my own billing software in
java. Here is what I am doing to track multiple calls at a time from
the prepaid account. 

1. Keep on db table for balance and reserver_balance.
2. First call coming to agi, check the balance - Sum of all the reserve_balance of that account code. 
3. Check the destination and allowed minutes for that balance amount from step 2. 
4. Reserve balance table will contain destination, amount, reserved secs columns
5. If the avaialable balance is <= 0 then announce not enough credit and hangup. 
6. If the available balance is > 0 but seconds allowed to talk is
less than reserved secs (see step 8 for more details about what this is)
   then set absolutetimeout for those seconds.
7. Otherwise the allowed seconds is more than the allowed seconds, set
absolute time out for the reserved seconds and make the call.
8. Reserved secs is a custom constant seconds, say you can reserve fund
for 3 minutes (180 seconds). if the account has balance for only 2
minutes (120 seconds) then the absolute time out will be 120 seconds. 
9. Once the channel status changed to reserved, insert an record to
reserve_balance table with uniqueid, accountcode, amount, reserved_secs
information. 

The above steps will handle one call so far now...and lets see how the dial plan should be,
10. In your dial plan, add an AbsoluteTimeout extension "T" and call
another AGI script which will just to reset the absolute timeout.
11. When the particular timeout is reached asterisk will transfer the
call to 'T" extension which will in turn call another agi. 
12. The agi will receive all the information about the channel
including uniqueid, repeat the steps 2- 7 (except dial) and reset the
abstimeout and this process will repeat until the channel hangup. 
13. Once the channel hangup, you can either use Manager to receive the
cdr event or you can set "h" extension (not reliable and not
recommended) to calculate the real balance and update the balance
table. Once you update the balance table, remove the record from
reserve_balance table for the uniqueid, channel and accountcode. (these
three are enough to find out the entry in that table).

Now lets take the scenario for  second call when the first call was  active, 
14. When the second call comes in, start from step 2. In step 2, we are
doing finalBalance = Balance - Sum(reserve_balance) for that account
code. If there is already a call on this accountcode, then this table
will have one entry and the reserved amount. Get the finalBalance by
subtracting the amounts. Follow step 3 and allow or deny the caller.
The above said solution is very stable and doesn't overflow the memory
or session and not using any threads. The only restriction here is, if
we have the scenario, 

Call -1 
balance = $0.10 
destination= 1 (which is US) 
rate = $0.02 per minute
reserveSecs  = 10 minutes (600secs)
finalBalance = $0.10 - $0 (consider this is first call and no entry in reserve_balance table) = $0.10 
allowedMints = $0.10/$0.02 = 5 minutes  = 300 seconds. 
AbsoluteTimeout = 300 seconds (this is less than the default reserveSecs so set this as abstimeout)

Call -2
balance = $0.10 

destination= 1 (which is US) 

rate = $0.02 per minute

reserveSecs  = 10 minutes (600secs)

finalBalance = $0.10 - $0.10 (consider this is second call and already an entry in reserve_balance table) = $0.0

allowedMints = 0 seconds. 
announce the denied ivr. 

So, the reserveSecs is critical to avoid how much threshold amount the
caller should have to make two calls. If they have $10 in their account
as per the above two algorithms, they can make as many simultaneous
calls. 

I hope this solves most of your problems. I looked at ASTCC, A2Billing
etc and they are not doing this way and not know whether they work
properly. But this works for me. Shoot me your questions if you have
one. 

I am developing my own billing and routing app (in java) and I need a
name for that.. guys pls suggest one.. i may put that in sourceforge if
i feel confident. 

Thanks,
Thameem
On 4/27/06, JP Carballo <[EMAIL PROTECTED]> wrote:
Dovid Bender wrote:>A while back some one posted some code that he used>that took out the flag in astcc that kept track if>there was a call in progress for that pin or not. Dont>know if it wil work for real time though.
>>Dovid>>I don't know if you were pertaining to what I posted in the message"ASTCC: How to reset "in-use" flag automatically ?".The setinuse() routine already exists in ASTCC.
One simply has to use that routine to disable the inuse flag when a callbegins and ASTCC will allow multiple calls for the same account.However, I too have no idea if this will work for real time.--
JP Carballohttp://www.netfone2x.comBringing the world closer.It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.___
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Re: [Asterisk-Users] billing realtime

2006-04-27 Thread JP Carballo

Dovid Bender wrote:


A while back some one posted some code that he used
that took out the flag in astcc that kept track if
there was a call in progress for that pin or not. Dont
know if it wil work for real time though.

Dovid
 

I don't know if you were pertaining to what I posted in the message 
"ASTCC: How to reset "in-use" flag automatically ?".

The setinuse() routine already exists in ASTCC.
One simply has to use that routine to disable the inuse flag when a call 
begins and ASTCC will allow multiple calls for the same account.


However, I too have no idea if this will work for real time.

--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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Re: [Asterisk-Users] Billing Server Open Source

2006-04-27 Thread Dovid Bender
astcc. it comes with asterisk.

--- [EMAIL PROTECTED] wrote:

> Any know of any working smart open source billing?
> Something smart that can do prepay/postpay and
> disconnect customers when they owe or a disconnect a
> call in progress for low balance.>
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Re: [Asterisk-Users] billing realtime

2006-04-27 Thread Dovid Bender

> JP Carballo wrote:
> 
> > Yes, certainly, through deadagi.
> > I just have one question though, why reinvent the
> wheel?
> > There are prepaid systems that work with asterisk.
> > 
> 
> I have yet to find a prepaid system that allows
> multiple concurrent
> calls per account. Most seem to be based on a pin
> number also which I
> don't want. Anyone know of a system that allows
> concurrent calls?


A while back some one posted some code that he used
that took out the flag in astcc that kept track if
there was a call in progress for that pin or not. Dont
know if it wil work for real time though.

Dovid

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RE: [Asterisk-Users] billing realtime

2006-04-26 Thread Josh McAllister
You keep eluding to the answer yourself. Asterisk Manager is the way to
go. Check out
http://search.cpan.org/~xantus/POE-Component-Client-Asterisk-Manager/.
Relatively simple event based method for using Asterisk manager. What I
would do is register a handler to track new calls, and calls ending.
Every time you get a new call, add it to a hash with the customer_id as
the key. Seperately register a callback that keeps re-calling itself at
X second intervals. It would cycle through the hash of active calls
decrementing remaining time for each, and then kick anyone with < 1
second remaining.

I have a single script running 12 instances of
POE::Component::Client::Asterisk::Manager (1 for each of 12 servers)
under a single POE kernel to track > 2500 channels (comings and goings
of MeetMe users) and it's had no problem keeping up. Just make sure that
you avoid any long running loops as POE is not multi-threaded.

For something like this, I think you'll find 1 instance of a single
script much easier to track and debug than a whole bunch of instance of
an AGI script.

Josh McAllister


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Farmer
Sent: Wednesday, April 26, 2006 7:27 AM
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] billing realtime



Nick Hoffman wrote:

> Hi Jon. If a customer has 10 minutes of call credit left and he makes
2 
> concurrent calls, how do you know to cut off the 2 calls at the 5
minute 
> mark rather than cut off both calls after 10 minutes?

That is the problem I am asking about :-)


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Telford, Shropshire, UK
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Re: [Asterisk-Users] billing realtime

2006-04-26 Thread Jon Farmer


Nick Hoffman wrote:

> Hi Jon. If a customer has 10 minutes of call credit left and he makes 2 
> concurrent calls, how do you know to cut off the 2 calls at the 5 minute 
> mark rather than cut off both calls after 10 minutes?

That is the problem I am asking about :-)


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Re: [Asterisk-Users] billing realtime

2006-04-26 Thread random cluster
 Hi Senad

i looking for same thing, that is consider absolutetimeout as a
timer, everytime is  near t zero, 3 secs for example, renew it,
reacalculate real credit, and start again until some of the parties
hangup.

The problem is how to iterate in asterisk config, or in deadagi,
you will need some time values from asterisk anyway, CDR{billsec} and
CDR{duration}, because i think we have to give this control to
asterisk, he really knows the timing of calls. Now the problem number
two. Asterisk set those values above, when the call is completely
finished, i have tried with deadagi in php whit sleep function,
nothing, the values of the varialbles are set after hangup extension,
after deadagi final execution.









 The solution that I looking for  is to take a average-time-call, and
create a timer with it.
 Then base on this value, and the price for destination call, every
time the average-time-call pass substract the consume credit from the
real credit, and set absolute timeout, for this average-time-call.

  But I dont know how to implement this is asterisk. With pseudo-code

while




2006/4/26, Senad Jordanovic <[EMAIL PROTECTED]>:
> [EMAIL PROTECTED] wrote:
> > On Wed April 26 2006 16:31, Jon Farmer <[EMAIL PROTECTED]> wrote:
> >> JP Carballo wrote:
> >>> Yes, certainly, through deadagi.
> >>> I just have one question though, why reinvent the wheel?
> >>> There are prepaid systems that work with asterisk.
> >>
> >> I have yet to find a prepaid system that allows multiple concurrent
> >> calls per account. Most seem to be based on a pin number also which I
> >> don't want. Anyone know of a system that allows concurrent calls? --
> >> Jon Farmer
> >> Telford, Shropshire, UK
> >
> >
> > Hi Jon. If a customer has 10 minutes of call credit left and he makes
> > 2 concurrent calls, how do you know to cut off the 2 calls at the 5
> > minute mark rather than cut off both calls after 10 minutes?
>
> The way we solved this is:
>
> 1/ Each account has incoming/outgoing channels
> 2/ Once call is started then the total balance is divided by number of
> outgoing channels for that account. This sets the time limit.
> 3/ If more calls are made then each new call has same absolute timeout.
>
> Above is not perfect, since we are limiting each call to less talk time then
> total balance allows, hence why we are currently
> looking into possibility in changing the value of absolute timeout in memory
> for each of the calls.
>
>
> Senad
>
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RE: [Asterisk-Users] billing realtime

2006-04-26 Thread Senad Jordanovic
[EMAIL PROTECTED] wrote:
> On Wed April 26 2006 16:31, Jon Farmer <[EMAIL PROTECTED]> wrote:
>> JP Carballo wrote:
>>> Yes, certainly, through deadagi.
>>> I just have one question though, why reinvent the wheel?
>>> There are prepaid systems that work with asterisk.
>>
>> I have yet to find a prepaid system that allows multiple concurrent
>> calls per account. Most seem to be based on a pin number also which I
>> don't want. Anyone know of a system that allows concurrent calls? --
>> Jon Farmer
>> Telford, Shropshire, UK
>
>
> Hi Jon. If a customer has 10 minutes of call credit left and he makes
> 2 concurrent calls, how do you know to cut off the 2 calls at the 5
> minute mark rather than cut off both calls after 10 minutes?

The way we solved this is:

1/ Each account has incoming/outgoing channels
2/ Once call is started then the total balance is divided by number of
outgoing channels for that account. This sets the time limit.
3/ If more calls are made then each new call has same absolute timeout.

Above is not perfect, since we are limiting each call to less talk time then
total balance allows, hence why we are currently
looking into possibility in changing the value of absolute timeout in memory
for each of the calls.


Senad

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Re: [Asterisk-Users] billing realtime

2006-04-26 Thread JP Carballo

Nick Hoffman wrote:

Hi Jon. If a customer has 10 minutes of call credit left and he makes 2 
concurrent calls, how do you know to cut off the 2 calls at the 5 minute 
mark rather than cut off both calls after 10 minutes?

-- Nick
e: [EMAIL PROTECTED]
p: +61 7 5591 3588
f: +61 7 5591 6588

If you receive this email by mistake, please notify us and do not make any 
use of the email.  We do not waive any privilege, confidentiality or 
copyright associated with it.
 

There's an application (sorry, which one, escapes me at the moment), 
that gets around this by reserving a certain amount of credit per call.
Say the amount is 10 minutes, if you have 30 minutes worth of credit, 
you can have 3 concurrent calls good for 10 minutes each.
The way I understand it, if you only have 15 minutes left in your 
account, the first call will last for 10 and the next concurrent one for 
5 minutes.


--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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Re: [Asterisk-Users] billing realtime

2006-04-25 Thread Nick Hoffman
On Wed April 26 2006 16:31, Jon Farmer <[EMAIL PROTECTED]> wrote:
> JP Carballo wrote:
> > Yes, certainly, through deadagi.
> > I just have one question though, why reinvent the wheel?
> > There are prepaid systems that work with asterisk.
>
> I have yet to find a prepaid system that allows multiple concurrent
> calls per account. Most seem to be based on a pin number also which I
> don't want. Anyone know of a system that allows concurrent calls?
> -- 
> Jon Farmer
> Telford, Shropshire, UK


Hi Jon. If a customer has 10 minutes of call credit left and he makes 2 
concurrent calls, how do you know to cut off the 2 calls at the 5 minute 
mark rather than cut off both calls after 10 minutes?
-- Nick
e: [EMAIL PROTECTED]
p: +61 7 5591 3588
f: +61 7 5591 6588

If you receive this email by mistake, please notify us and do not make any 
use of the email.  We do not waive any privilege, confidentiality or 
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Re: [Asterisk-Users] billing realtime

2006-04-25 Thread Darren Wiebe
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Jon, we can do that using ASTPP.  The downside is that we don't
currently have a way to limit the call lengths so that when they have
multiple calls in progress they still can't go over their prepaid limit.
 On postpaid accounts this is not usually an issue but on prepaid it
still is.

Darren Wiebe
[EMAIL PROTECTED]

Jon Farmer wrote:
> JP Carballo wrote:
> 
>> Yes, certainly, through deadagi.
>> I just have one question though, why reinvent the wheel?
>> There are prepaid systems that work with asterisk.
>>
> 
> I have yet to find a prepaid system that allows multiple concurrent
> calls per account. Most seem to be based on a pin number also which I
> don't want. Anyone know of a system that allows concurrent calls?
> 
> 

-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.5 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFETxRg4DADnh+tnOQRAuhJAJ9kzGiQYh4Z6WPXXes6TKtwusBliwCeMvHG
3nrqsxdXNrfJbCZ3uzlpd5w=
=+fV+
-END PGP SIGNATURE-
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Re: [Asterisk-Users] billing realtime

2006-04-25 Thread Jon Farmer

JP Carballo wrote:

> Yes, certainly, through deadagi.
> I just have one question though, why reinvent the wheel?
> There are prepaid systems that work with asterisk.
> 

I have yet to find a prepaid system that allows multiple concurrent
calls per account. Most seem to be based on a pin number also which I
don't want. Anyone know of a system that allows concurrent calls?


-- 
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Telford, Shropshire, UK
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Re: [Asterisk-Users] billing realtime

2006-04-25 Thread JP Carballo

random cluster wrote:


  Now, the question, can I access somehow in a deadagi, or
whatever the CDR function
in order to update the credit when the call has just finished.

 


Yes, certainly, through deadagi.
I just have one question though, why reinvent the wheel?
There are prepaid systems that work with asterisk.

--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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[Asterisk-Users] billing realtime

2006-04-25 Thread random cluster
Hi all

 I think this could be en old question. I would like to do a
realtime billing prepaid system, mainly using asterisk.
  I have found few things;

 I can not get CDR function into agi because asterisk set them
once the call is absolutely finish (at least main values for the main
porpouse, billsec,duration, etc..)

 There is a patch that allow you to use CDR function in hangup
extension, but it seems to have some troubles, haven't it??
   Finally, another solution I have found reading somewhere
could be use triggers in the database when there is and CDR insert.
   A cron job could do something similar.

   Now, the question, can I access somehow in a deadagi, or
whatever the CDR function
in order to update the credit when the call has just finished.


Thank you very much
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RE: [Asterisk-Users] Billing Server Open Source

2006-04-19 Thread Mindaugas Kezys








http://www.paskambink.lt/mcc

 



Regards/Pagarbiai,

Mindaugas Kezys











From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of [EMAIL PROTECTED]
Sent: Monday, April 17, 2006 8:13
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Billing
Server Open Source



 



Any know of any working smart open source billing? Something smart that
can do prepay/postpay and disconnect customers when they owe or a disconnect a
call in progress for low balance.








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RE: [Asterisk-Users] Billing Server Open Source

2006-04-17 Thread William Piper








FYI, this is more of a question for the
asterisk-biz list.

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, April 17, 2006 1:13
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Billing
Server Open Source



 



Any know of any working smart open source billing? Something smart that
can do prepay/postpay and disconnect customers when they owe or a disconnect a
call in progress for low balance.





__ NOD32 1.1492 (20060416) Information __

This message was checked by NOD32 antivirus system.
http://www.eset.com






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RE: [Asterisk-Users] Billing Server Open Source

2006-04-17 Thread William Piper








http://www.asterisk2billing.org/

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, April 17, 2006 1:13
PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Billing
Server Open Source



 



Any know of any working smart open source billing? Something smart that
can do prepay/postpay and disconnect customers when they owe or a disconnect a
call in progress for low balance.





__ NOD32 1.1492 (20060416) Information __

This message was checked by NOD32 antivirus system.
http://www.eset.com






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[Asterisk-Users] Billing Server Open Source

2006-04-17 Thread broadbandvoice

Any know of any working smart open source billing? Something smart that can do prepay/postpay and disconnect customers when they owe or a disconnect a call in progress for low balance.

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RE: [Asterisk-Users] billing with PostgreSQL

2006-04-14 Thread Mindaugas Kezys
You can try:

http://www.paskambink.lt/mcc


Regards/Pagarbiai,
Mindaugas Kezys

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira
Sent: Wednesday, April 12, 2006 3:15 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] billing with PostgreSQL

Hello to all
Im looking for a billing tool for Asterisk, that works with PostgreSQL.
All the tools I found in www.asteriskbilling.com just work with MySQL :(

Do you know a nice billing tool for Asterisk with PostgreSQL?

Thanks
Joao Pereira

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Re: [Asterisk-Users] billing with PostgreSQL

2006-04-12 Thread Andy Tan
Hi Joao,

some billing solutions are listed here ->
http://www.voip-info.org/wiki/view/Open+Source+Billing+Systems

IIRC, none works with PGSQL. My opinion is that considering the
importance of billing, it's better to develop a customised solution.
That way, you would have full understanding and confidence in it.
References to other systems can be  useful also. Hope it helps.

Regards
Andy Tan

On Wed, 12 Apr 2006 11:15:24 +0100, "Joao Pereira"
<[EMAIL PROTECTED]> said:
> Hello to all
> Im looking for a billing tool for Asterisk, that works with PostgreSQL.
> All the tools I found in www.asteriskbilling.com just work with MySQL :(
> 
> Do you know a nice billing tool for Asterisk with PostgreSQL?
> 
> Thanks
> Joao Pereira
> 
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[Asterisk-Users] billing with PostgreSQL

2006-04-12 Thread Joao Pereira

Hello to all
Im looking for a billing tool for Asterisk, that works with PostgreSQL.
All the tools I found in www.asteriskbilling.com just work with MySQL :(

Do you know a nice billing tool for Asterisk with PostgreSQL?

Thanks
Joao Pereira

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[Asterisk-Users] Billing from CDR files

2006-03-23 Thread Chris Mason (Lists)
I have a Astlinux system that can't run mysql. I want to bill calls by 
extension but all I have is the raw CDR files. Is there any software 
that cal produce a simple call accounting from text CDR files?


--
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NetConcepts
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Re: [Asterisk-Users] billing - different tarif per phone

2006-02-27 Thread Darren Wiebe
I think that the feature you're looking for is called "pricelists" in 
ASTPP but I could misunderstand what you want.  Feel free to post the 
question either on the astpp-users mailing list or the astpp forum.  
Visit www.astpp.org for more info.


Darren Wiebe
[EMAIL PROTECTED]

Pavel Jezek wrote:

Hello, I would like apply different call rate (tarif) per outgoing 
number (or group of phones, based on prefixes),
I'm playing with astpp, but seems, that this feature isn't available 
here,
can you recommend any other open-source billing (A2billing, AstBill?), 
that this can do?

thank you!
PJ

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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing & Calling Cards
www.aleph-com.net/astpp

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[Asterisk-Users] billing - different tarif per phone

2006-02-27 Thread Pavel Jezek
Hello, I would like apply different call rate (tarif) per outgoing 
number (or group of phones, based on prefixes),

I'm playing with astpp, but seems, that this feature isn't available here,
can you recommend any other open-source billing (A2billing, AstBill?), 
that this can do?

thank you!
PJ

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Re: [Asterisk-Users] Billing inbound calls per minute

2006-02-20 Thread Darren Wiebe
Hours of struggling later, I have found the problem.  Here is the 
correct format for those outgoing calls.


SIP/[EMAIL PROTECTED]||L(54081429:6:3)|Hj

I'll try to get a patch done up one of these days.

Darren Wiebe
[EMAIL PROTECTED]

[EMAIL PROTECTED] wrote:


On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
 


I've been playing with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the "Connect fee"(if I put one)
and keeps it that way no matter how long
the call is ...( if no "Connect fee" -stays empty).
i.e.
[inbound]
exten => 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten => 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten => 1122334455,3,Hangup
   


Michiel van Baak wrote:
   


DeadAGI is for hungup channels, not for active channels.
That might be a problem.

Try this:
exten => h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
 


On Monday 06 February 2006 09:25, JP Carballo wrote:
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
   


Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
 



 


On Saturday 11 February 2006 06:32, Darren Wiebe wrote:
Are you running a relatively recent version of ASTCC?  Say within the
last 6 months.  The answeredtime = 0 bug was supposed to have been fixed
by http://bugs.digium.com/view.php?id=4300  Unless something has changed
in Asterisk that affects this
   



Thanks Daren,
Yes, my version of astcc is the most recent one.
Asterisk-1.2.4
I have found you patch 0004300 from 16 May 2005.
Probably it's time to reverse it back since "something has changed
in Asterisk that affects this..." as you said.
My observation is:
If I keep:
$dialstr = "Local/[EMAIL PROTECTED]>{path}|30|HL/n(" . ($maxtime * 60 * 1000) . 
":6:3)";
Either the billseconds is empty(when dial out through Local), either there is 
a when dialing in. 
I put back the dialstring to:
"Local/$phone/$res->{path}|30|HL/n(" . ($maxtime * 60 * 1000) . 
":6:3)";

The only difference that it looks only for is a default context.

extensions.conf
[inbound]
; 10 digits DID = _XX = cardnumber
; 
exten => _XX ,1,Answer()

exten => _XX ,n,Set(DB(RCID/${CALLERIDNUM})=${CALLERIDNUM})
exten => _XX ,n,Set(realcid=${DB(RCID/${CALLERIDNUM})})
exten => _XX ,n,Noop(${REALCID})
;exten => _XX ,n,Set(TIMEOUT(digit)=4)
exten => _XX ,n,Set(CALLERID(number)=${EXTEN})
exten => _XX ,n,Set(CALLERID(name)= ${REALCID})
;exten => t,3,Goto(h|1)
;exten => _XX 2,Goto(s|1)
;exten => s,1,Wait,1 ; is this preventing HUP?
exten => h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${CALLERIDNUM},4) 
; must be "h,1" as per Michiel van Baak note(above).

exten => h,2,Hangup
[internal]
; i.e. 360 1234567 = DID = card
exten => 3601234567,1,Macro(stdexten,3601234567,sip/did_owner)
[default]
include => internal
[personal]
exten => t,1,Hangup
include => inbound

Result:
- ANSWEREDTIME is OK
- inbound call billed on the callee
- there is CALLERID(name) for callerid in the cdrs(kind of)
There is still a small but "looong" problem - Timeout about 10 
secs long while the IAX2/incoming Hangup in personal,t,1.

But CDR is updated after that and the call is billed as expected.

Sorry for the long explanation.
What do you think? Is there something "suspicious" in
that solution?
Thanks,
benchev

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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing & Calling Cards
www.aleph-com.net/astpp

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Re: [Asterisk-Users] Billing inbound calls per minute

2006-02-11 Thread bbench
> >>>On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
> I've been playing with astcc, but while
> 'billseconds' stays empty, 'billcost' has
> strange behavior - either stays ampty
> or takes ONCE the "Connect fee"(if I put one)
> and keeps it that way no matter how long
> the call is ...( if no "Connect fee" -stays empty).
> i.e.
> [inbound]
> exten => 1122334455,1,Set(CALLERID(number)=${EXTEN})
> exten => 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
> exten => 1122334455,3,Hangup
> >>>
> >>Michiel van Baak wrote:
> >>>DeadAGI is for hungup channels, not for active channels.
> >>>That might be a problem.
> >>>
> >>>Try this:
> >>>exten => h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
> >>
> >>On Monday 06 February 2006 09:25, JP Carballo wrote:
> >>ASTCC works fine here. The duration and billseconds fields in my cdrs as
> >>well as ASTCC's cdr are filled.
> >>I don't use the connect fee field though and all are set to 0.
> >
> >Would you share with me how'd you do billing on a DID
> >(if you do), and through what Technology?
> >Anything that goes Local here is ANSWEREDTIME zero.

>On Saturday 11 February 2006 06:32, Darren Wiebe wrote:
> Are you running a relatively recent version of ASTCC?  Say within the
> last 6 months.  The answeredtime = 0 bug was supposed to have been fixed
> by http://bugs.digium.com/view.php?id=4300  Unless something has changed
> in Asterisk that affects this

Thanks Daren,
Yes, my version of astcc is the most recent one.
Asterisk-1.2.4
I have found you patch 0004300 from 16 May 2005.
Probably it's time to reverse it back since "something has changed
in Asterisk that affects this..." as you said.
My observation is:
If I keep:
$dialstr = "Local/[EMAIL PROTECTED]>{path}|30|HL/n(" . ($maxtime * 60 * 1000) . 
":6:3)";
Either the billseconds is empty(when dial out through Local), either there is 
a when dialing in. 
I put back the dialstring to:
"Local/$phone/$res->{path}|30|HL/n(" . ($maxtime * 60 * 1000) . 
":6:3)";
The only difference that it looks only for is a default context.

extensions.conf
[inbound]
; 10 digits DID = _XX = cardnumber
; 
exten => _XX ,1,Answer()
exten => _XX ,n,Set(DB(RCID/${CALLERIDNUM})=${CALLERIDNUM})
exten => _XX ,n,Set(realcid=${DB(RCID/${CALLERIDNUM})})
exten => _XX ,n,Noop(${REALCID})
;exten => _XX ,n,Set(TIMEOUT(digit)=4)
exten => _XX ,n,Set(CALLERID(number)=${EXTEN})
exten => _XX ,n,Set(CALLERID(name)= ${REALCID})
;exten => t,3,Goto(h|1)
;exten => _XX 2,Goto(s|1)
;exten => s,1,Wait,1 ; is this preventing HUP?
exten => h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${CALLERIDNUM},4) 
; must be "h,1" as per Michiel van Baak note(above).
exten => h,2,Hangup
[internal]
; i.e. 360 1234567 = DID = card
exten => 3601234567,1,Macro(stdexten,3601234567,sip/did_owner)
[default]
include => internal
[personal]
exten => t,1,Hangup
include => inbound

Result:
- ANSWEREDTIME is OK
- inbound call billed on the callee
- there is CALLERID(name) for callerid in the cdrs(kind of)
There is still a small but "looong" problem - Timeout about 10 
secs long while the IAX2/incoming Hangup in personal,t,1.
But CDR is updated after that and the call is billed as expected.

Sorry for the long explanation.
What do you think? Is there something "suspicious" in
that solution?
Thanks,
benchev

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Re: [Asterisk-Users] Billing inbound calls per minute

2006-02-11 Thread bbench
> >On Monday 06 February 2006 09:25, JP Carballo wrote:
> >> 
> >>
> >>ASTCC works fine here. The duration and billseconds fields in my cdrs as
> >>well as ASTCC's cdr are filled.
> >>I don't use the connect fee field though and all are set to 0.
> >
> >Would you share with me how'd you do billing on a DID
> >(if you do), and through what Technology?
> >Anything that goes Local here is ANSWEREDTIME zero.
> >Thanks,
> >benchev
>
> That probably explains it.
> IIRC, from when I was still testing ASTCC, when calling a Local channel,
> the AGI suffers from short term memory loss and forgets the values of
> channel variables even if "/n" is used in the dial string.
> I checked my test server logs and while I can verify that ASTCC's CDR
> does have blank duration and billsec fields for the Local calls, *'s CDR
> records them.
Similar here, and I read the patch from Darren May, 2005
where "Local/$phone/$res->{path}|30|HL/n was changed to
"Local/[EMAIL PROTECTED]>{path}|30|HL/n


> I do billing based on account number so clients are free to call from
> any phone. I don't check callerid.
> Since each account is based on the phone number registered by the
> client, I can just chop off the 2 digit prefix and set their callerid
> with the result.
Yes, I do that also with another instance of astcc, I call astcc-disa.agi
to allow clients from outside to enter * and do things.
> [makecall]
> exten => s,1,Set(CALLERID(num)=${CARDNO:2})
> exten => s,n,DeadAGI(astcc.agi,${CARDNO})
> exten => s,n,Goto(nf2xsubmenu,s,1)
>
> All my calls are routed to IAX2 or SIP or Zap.
And this is my problem because my target is to use Local, but
please follow my answer, within that thread, to Darren.

Thanks very much for your help.
benchev
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Re: [Asterisk-Users] Billing inbound calls per minute

2006-02-10 Thread Darren Wiebe
Are you running a relatively recent version of ASTCC?  Say within the 
last 6 months.  The answeredtime = 0 bug was supposed to have been fixed 
by http://bugs.digium.com/view.php?id=4300  Unless something has changed 
in Asterisk that affects this


[EMAIL PROTECTED] wrote:


On Monday 06 February 2006 09:25, JP Carballo wrote:
 


Michiel van Baak wrote:
   


On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
 


Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?

I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the "Connect fee"(if I put one)
and keeps it that way no matter how long
the call is ...( if no "Connect fee" -stays empty).

i.e.
[inbound]
exten => 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten => 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten => 1122334455,3,Hangup
   


DeadAGI is for hungup channels, not for active channels.
That might be a problem.

Try this:
exten => h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
 


ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
   


Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
Thanks,
benchev
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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing & Calling Cards
www.aleph-com.net/astpp

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Re: [Asterisk-Users] Billing inbound calls per minute

2006-02-10 Thread JP Carballo

[EMAIL PROTECTED] wrote:


On Monday 06 February 2006 09:25, JP Carballo wrote:
 




ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
   


Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
Thanks,
benchev
 


That probably explains it.
IIRC, from when I was still testing ASTCC, when calling a Local channel, 
the AGI suffers from short term memory loss and forgets the values of 
channel variables even if "/n" is used in the dial string.
I checked my test server logs and while I can verify that ASTCC's CDR 
does have blank duration and billsec fields for the Local calls, *'s CDR 
records them.

If it's also true for you, you might want to use *'s CDRs for rating.

I do billing based on account number so clients are free to call from 
any phone. I don't check callerid.
Since each account is based on the phone number registered by the 
client, I can just chop off the 2 digit prefix and set their callerid 
with the result.


[makecall]
exten => s,1,Set(CALLERID(num)=${CARDNO:2})
exten => s,n,DeadAGI(astcc.agi,${CARDNO})
exten => s,n,Goto(nf2xsubmenu,s,1)

All my calls are routed to IAX2 or SIP or Zap.

--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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Re: [Asterisk-Users] Billing inbound calls per minute

2006-02-10 Thread bbench
On Monday 06 February 2006 09:25, JP Carballo wrote:
> Michiel van Baak wrote:
> >On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
> >>Hi,
> >>Does anyone have a neat idea as how to
> >>bill inbound calls per minute(second) real time?
> >>
> >>I've been pplaying with astcc, but while
> >>'billseconds' stays empty, 'billcost' has
> >>strange behavior - either stays ampty
> >>or takes ONCE the "Connect fee"(if I put one)
> >>and keeps it that way no matter how long
> >>the call is ...( if no "Connect fee" -stays empty).
> >>
> >>i.e.
> >>[inbound]
> >>exten => 1122334455,1,Set(CALLERID(number)=${EXTEN})
> >>exten => 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
> >>exten => 1122334455,3,Hangup
> >
> >DeadAGI is for hungup channels, not for active channels.
> >That might be a problem.
> >
> >Try this:
> >exten => h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
>
> ASTCC works fine here. The duration and billseconds fields in my cdrs as
> well as ASTCC's cdr are filled.
> I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
Thanks,
benchev
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Re: [Asterisk-Users] Billing inbound calls per minute

2006-02-06 Thread bbench
> >>Does anyone have a neat idea as how to
> >>bill inbound calls per minute(second) real time?
> >>
> >>I've been pplaying with astcc, but while
> >>'billseconds' stays empty, 'billcost' has
> >>strange behavior - either stays ampty
> >>or takes ONCE the "Connect fee"(if I put one)
> >>and keeps it that way no matter how long
> >>the call is ...( if no "Connect fee" -stays empty).
> >>i.e.
> >>[inbound]
> >>exten => 1122334455,1,Set(CALLERID(number)=${EXTEN})
> >>exten => 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
> >>exten => 1122334455,3,Hangup
> >
> >DeadAGI is for hungup channels, not for active channels.
> >That might be a problem.
> >
> >Try this:
> >exten => h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
>
Thanks, tried that several ways but no help since ${EXTEN}=h.
Probably will try with CHANISAVAIL or ${CHANNEL} or something...
> ASTCC works fine here. The duration and billseconds fields in my cdrs as
> well as ASTCC's cdr are filled.
> I don't use the connect fee field though and all are set to 0.
Sure ASTCC works, but I am talking about inbound calls
where 1122334455 is a DID as well as a card number being
charged for the incoming calls.  Thus ${EXTEN}=DID=card i.e.
exten => 1122334455,1,Set(CALLERID(number)=${EXTEN})

Mayby I should not assosiate DID from card(user) and create a separate
peer for the DID on a different port.

Any other ideas? Thanks.
benchev

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Re: [Asterisk-Users] Billing inbound calls per minute

2006-02-05 Thread JP Carballo

Michiel van Baak wrote:


On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
 


Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?

I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the "Connect fee"(if I put one)
and keeps it that way no matter how long
the call is ...( if no "Connect fee" -stays empty).

i.e.
[inbound]
exten => 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten => 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten => 1122334455,3,Hangup
   



DeadAGI is for hungup channels, not for active channels.
That might be a problem.

Try this:
exten => h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
 

ASTCC works fine here. The duration and billseconds fields in my cdrs as 
well as ASTCC's cdr are filled.

I don't use the connect fee field though and all are set to 0.

--
JP Carballo

http://www.netfone2x.com
Bringing the world closer.

It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. 


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Re: [Asterisk-Users] Billing inbound calls per minute

2006-02-05 Thread Michiel van Baak
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
> Hi,
> Does anyone have a neat idea as how to
> bill inbound calls per minute(second) real time?
> 
> I've been pplaying with astcc, but while
> 'billseconds' stays empty, 'billcost' has
> strange behavior - either stays ampty
> or takes ONCE the "Connect fee"(if I put one)
> and keeps it that way no matter how long
> the call is ...( if no "Connect fee" -stays empty).
> 
> i.e.
> [inbound]
> exten => 1122334455,1,Set(CALLERID(number)=${EXTEN})
> exten => 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
> exten => 1122334455,3,Hangup

DeadAGI is for hungup channels, not for active channels.
That might be a problem.

Try this:
exten => h,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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[Asterisk-Users] Billing inbound calls per minute

2006-02-05 Thread bbench
Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?

I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the "Connect fee"(if I put one)
and keeps it that way no matter how long
the call is ...( if no "Connect fee" -stays empty).

i.e.
[inbound]
exten => 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten => 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten => 1122334455,3,Hangup

Thanks in advance,
benchev




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