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
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
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
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 ewiel...@nyigc.com wrote:
Often it is
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
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
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
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
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk: http://the-asterisk-book.com -
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..
___
-- Bandwidth and Colocation Provided by
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.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678)
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
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
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
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
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
PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
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
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,
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
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
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
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 MendozaStefano Corsi wrote:Hello,I've discovered that in Italy ISDN lines can be
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
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
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
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
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
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,
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:
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
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
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:
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:
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,
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?
-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
Try looking at enswitch. It is a paid solution.
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, November 30, 2006 6:29 PM
Subject: [asterisk-users] Billing Software
We are
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
asterisk-users@lists.digium.com
Sent: Wednesday,
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/
/
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
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
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
___
--Bandwidth and Colocation provided by Easynews.com --
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
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
___
[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
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
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
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
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.
___
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
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
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
-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
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
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
[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
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
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 :-)
--
Jon Farmer
Telford,
: 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
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
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
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
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
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
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:
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,
On Monday 06 February 2006 09:25, JP Carballo wrote:
snip
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
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
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,
[EMAIL PROTECTED] wrote:
On Monday 06 February 2006 09:25, JP Carballo wrote:
snip
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
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
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
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
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
Yes.
[EMAIL PROTECTED] wrote:
Hello All,
Have anybody test ISP BILLING SYSTEM ?
http://ibs.sourceforge.net/index.html
Regards
Harry
___
Nouveau : téléphonez moins cher avec Yahoo!
Dinesh Nair ha scritto:
On 10/10/05 22:30 Waldo Rubinstein said the following:
1) When are asterisk CDR logs _normally_ generated? When the call
arrives, when the call hangs up, or both? I have looked at the records
when the call hangs up.
But if you use a h extension, at the end of
Have a look at http://astbill.com it is FREE and Open
SOURCE.
4) Because this (item 3) has already happened to me, is there any
free tool out there that will allow me to parse the CDR logs in order
to determine the maximum number of simultaneous calls that a
particular SIP peer has made within
You mean to say that it will ONLY log if I have an h extension or if
I don't? Shouldn't it be logged no matter what?
- Waldo
On Oct 11, 2005, at 5:31 AM, Simone Cittadini wrote:
Dinesh Nair ha scritto:
On 10/10/05 22:30 Waldo Rubinstein said the following:
1) When are asterisk CDR logs
Thanks.I understand your POV. However, in addition to usage-based billing (which is what you refer to), I need to bill for the account. So, if the user placed two simultaneous calls with the same account, that may be fine because it could have been the call-waiting feature. However, if the user
Waldo Rubinstein ha scritto:
You mean to say that it will ONLY log if I have an h extension or if
I don't? Shouldn't it be logged no matter what?
No, of course it logs no matter whats, I was meaning that if you have
exten = h,1,...
exten = h,2,
ecc ...
don't expect the h extension to
On 10/10/05 22:30 Waldo Rubinstein said the following:
1) When are asterisk CDR logs _normally_ generated? When the call
arrives, when the call hangs up, or both? I have looked at the records
when the call hangs up.
certain calls, I was thinking of doing something with FastAGI so that
The way I do it is to make a list of internal extensions and set those
to no charge. They get billed at no charge that way and it works fine.
/Plug Starts/ This is done using ASTPP www.aleph-com.net/astpp/ /Plug Ends/
Darren Wiebe
[EMAIL PROTECTED]
Chris Bagnall wrote:
Hi all,
I have
This billing is also able to set accounts balance and for each call. Now I
need to disable accounts which balance gets a determined value. I was
thinking on changing account pass for that specif account which we need to
disable. And then in the sip.com reload info.
Can you help me with new
Thanks Darren,
i applied the patch you mentioned and now i have billing cost.
I have to check it more, in the following days but i think that the patch
did the right thing!
Panos.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Can you post the output from the console when a call goes through? You
might also want to try the patch @ http://bugs.digium.com/view.php?id=4479
We need feedback on this patch.
Darren Wiebe
[EMAIL PROTECTED]
Panayiotis Kolyvas wrote:
Hi to everybody,
i tried to find an asnwer before
To breifly recap
Your main asterisk box runs linux, asterisk, ASTCC and MySQL
Another box runs linux, mysql, apache
The two sql servers are joined, updating each other?
or have I missed something?
___
Asterisk-Users mailing list
Why not to use one of the existing CallingCard solutions such AstCC
AreskiCC! There are pretty mature already and perhaps it would be better
to add your efforts on one of them!
BTW you can look on the sources to see how we manage
ANSWEREDTIME DIALEDTIME!
Rgds, Areski
On Tue, 2005-04-19 at
http://www.asterisk-support.ru/Members/litnimax/HowTo/DTLHowto/
My howto for using asterisk with any billing.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
Rizwan Chaudhry wrote:
Hey
I want to implement billing in Asterisk for a calling card type application.
My scenario is like this: PSTN = Asterisk =(IAX)= Asterisk = PSTN.
I used the ${ANSWEREDTIME} and ${DIALEDTIME} variables but
${ANSWEREDTIME} always gives a value even if the call is not
card systems out there.
Cheers
Sathya
-Original Message-
From: Maxim Litnitsky [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 19, 2005 3:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Billing
http://www.asterisk-support.ru/Members
Just a note on scaling astcc, you can have a database with server
replication, so that it scales well, and doesnt subtract from the cpu
power of the asterisk boxes. This is regardless of medium for the voice
calls.
If you then distribute the load across multiple asterisk boxes you build
in a
www.flexcom.lu
- Original Message -
From: Ronald Wiplinger [mailto:[EMAIL PROTECTED]
Sent: 12/5/2004 5:13:07 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Billing - which program are you using?
I want to play around with post billing. List of all phone calls, ...
Which program
On Sun, Dec 05, 2004 at 12:13:07PM +0800, Ronald Wiplinger spake thusly:
I want to play around with post billing. List of all phone calls, ...
Which program is useful for that?
All what I have seen are not based on CDR, but on Radius.
What are you using?
Everyone pretty much writes their
On Mon, 2004-11-29 at 10:58 +0100, Rastislav Lukac wrote:
[snip]
Maybe there is a way to catch the billing information
from D-channel. Is there any standalone application
for linux, which is able to filter these charging informations
when the Asterisk can't do that?
I don't know. Seems
Hello!
With 'PSTN' lines - do I understand correctly that you use ISDN lines? If so, I would probably not
of much help. Otherwise - I just had a billing problem with an analog line and solved it for our
telco. See the thread
Billing (itemized) in the UK in this months mailing.
(But if it is
On Fri, 2004-11-26 at 10:35 +0100, Rastislav Lukac wrote:
Hello all,
I would like to get billing/charging informations of all
outgoing calls of any PSTN numbers made with my IP-Phone via asterisk.
Asterisk automatically generates CDR's (Call Detail Records). They are
stored in cdr-csv (or
Hi Peter
You need to first of all ask your Telco what mechanism it uses with your
current switch. The most likely ways are
1) Two stage dialling. 1xxx pause PIN exten dialled number
2) access code 1xxx exten dialled number
You need to get the specs for this from Your
To give the extension information to the telco...
How can I configure Asterisk to do send extension information?
[Senad Jordanovic]
This greatly depends on your provider...
What signalling do I have to provide for outgoing calls to give
extension
information the telco?
[Senad Jordanovic]
Pete,
I am also in the UK and I have added an include in my extensions.conf for
the file listed bellow.
exten = _15X,1,Dial,${TRUNK}/BYEXTENSION
exten = _147X,1,Dial,${TRUNK}/BYEXTENSION
exten = _NX,1,Dial,${TRUNK}/BYEXTENSION
exten = _01.,1,Dial,${TRUNK}/BYEXTENSION
exten =
You just need to do something like
exten = _9.,1,Dial(Zap/g1/1666$CALLERIDNUM${EXTEN:1})
You can also do some useful translations like
exten = _9[2-8]XX,1,Dial(Zap/g1/1666$CALLERIDNUM0113${EXTEN:1})
This will look for 9, then a local number beginning 2,3,4,5,6,7,8 , and
dial out the
Thank you very much for the answers - I have hooked up a special adapter and active loudspeaker on
each of the three BT lines, but when I got a line and dial a number I cannot hear any other digits
than those I dial - I would have expected something like seven DTMF bursts/digits (16662xx) before
1 - 100 of 125 matches
Mail list logo