Re: [asterisk-users] Billing software: Other than A2Billing because of the problem with the analogue channels
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. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1
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? > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1
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/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing number vs. CallerID number | Asterisk 11.5.1
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? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] billing based on local access number
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);
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
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Re: [asterisk-users] Billing applications
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing applications
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 > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing and Soft Switch.
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) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing and Soft Switch.
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 http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing and Soft Switch.
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 - http://das-asterisk-buch.de AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing/Call Control engine : AGI scripts/ AstMan API
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. > > > > > > > EMAIL DISCLAIMER : This email and any files transmitted with it are > confidential and intended solely for the use of the individual or entity to > whom they are addressed. Any unauthorised distribution or copying is strictly > prohibited. If you receive this transmission in error, please notify the > sender by reply email and then destroy the message. Opinions, conclusions and > other information in this message that do not relate to official business of > Mascon shall be understood to be neither given nor endorsed by Mascon. Any > information contained in this email, when addressed to Mascon clients is > subject to the terms and conditions in governing client contract. > > Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, > we can not guarantee that any email or attachment is free from computer > viruses and you are strongly advised to undertake your own anti-virus > precautions. Mascon grants no warranties regarding performance, use or > quality of any e-mail or attachment and undertakes no liability for loss or > damage, howsoever caused. > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- "Within C++, there is a much smaller and cleaner language struggling to get out." ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Billing/Call Control engine : AGI scripts/ AstMan API
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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Re: [asterisk-users] Billing Telephone Number (BTN)
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 ___ --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
Re: [asterisk-users] Billing pulses
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. ___ --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 ___ --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
Re: [asterisk-users] Billing pulses
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. > > ___ --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
Re: [asterisk-users] Billing pulses
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. ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.30/674 - Release Date: 2/7/2007 3:33 PM ___ --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
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. ___ --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
Re: [asterisk-users] Billing pulses
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 > > > ___ > --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 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 ___ --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
Re: [asterisk-users] Billing pulses
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 ___ --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
Re: [asterisk-users] Billing pulses
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 ___ --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
Re: [asterisk-users] Billing pulses
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 > > ___ > --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 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 ___ --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
Re: [asterisk-users] Billing pulses
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 ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.29/673 - Release Date: 2/6/2007 5:52 PM ___ --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
Re: [asterisk-users] Billing pulses
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 ___ --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
Re: [asterisk-users] Billing pulses
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 ___ --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
Re: [asterisk-users] Billing pulses
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 ___ --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
Re: [asterisk-users] Billing pulses
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 ___ --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 ___ --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
Re: [asterisk-users] Billing solution
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 ___ --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
Re: [asterisk-users] Billing solution
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 > > > > > ___ > > > > > --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 > > > > > > > > ___ > > > > --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 > > > > > > > > > > > > > > > > > > > > > ___ > > > --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 > > > > > > > > > > > > > > > ___ > > --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 > > > > > > > ___ > --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 > 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 . ___ --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 ___ --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
Re: [asterisk-users] Billing solution
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 ___ --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 ___ --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
Re: [asterisk-users] Billing solution
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 > > > > > > ___ > > > --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 > > > > > > > > > > > > > > > ___ > > --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 > > > > > > > > > ___ > --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 > > > ___ --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 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 . ___ --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
Re: [asterisk-users] Billing solution
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 > > > > ___ > > --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 > > > > > > > > > ___ > --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 > > > ___ --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 ___ --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
Re: [asterisk-users] Billing solution
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 > > ___ > --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 > > > ___ --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 ___ --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
Re: [asterisk-users] Billing solution
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 ___ --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 ___ --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
Re: [asterisk-users] Billing solution
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 ___ --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
Re: [asterisk-users] Billing Software
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 > > ___ > --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 -- 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 ___ --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
Re: [asterisk-users] Billing Software
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 ___ --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 ___ --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
Re: [asterisk-users] Billing software with reseller accounts
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 http://es.wikipedia.org/wiki/Top-posting ___ --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 ___ --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
Re: [asterisk-users] billing
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 > ___ > --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 ___ --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
Re: [asterisk-users] billing
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 ___ --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
Re: [asterisk-users] billing
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/ / ___ --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
Re: [asterisk-users] Billing Solution ?
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 ___ ___ --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
Re: [asterisk-users] Billing Solution ?
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 > ___ > --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 -- 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 ___ --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
Re: [asterisk-users] Billing Solution ?
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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
RE: [Asterisk-Users] billing realtime
[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
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
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.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing lis
Re: [Asterisk-Users] billing realtime
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. ___ --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
Re: [Asterisk-Users] Billing Server Open Source
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.> ___ > --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 > __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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
Re: [Asterisk-Users] billing realtime
> 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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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
RE: [Asterisk-Users] billing realtime
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 :-) -- Jon Farmer Telford, Shropshire, UK ___ --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 ___ --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
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 :-) -- Jon Farmer Telford, Shropshire, UK ___ --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
Re: [Asterisk-Users] billing realtime
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
[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 ___ --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
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? -- 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. ___ --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
Re: [Asterisk-Users] billing realtime
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 copyright associated with it. ___ --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
Re: [Asterisk-Users] billing realtime
-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- ___ --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
Re: [Asterisk-Users] billing realtime
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 ___ --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
Re: [Asterisk-Users] billing realtime
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. ___ --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
RE: [Asterisk-Users] Billing Server Open Source
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. ___ --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
RE: [Asterisk-Users] Billing Server Open Source
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 ___ --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
RE: [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 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 ___ --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
RE: [Asterisk-Users] billing with PostgreSQL
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 ___ --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 ___ --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
Re: [Asterisk-Users] billing with PostgreSQL
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 > > ___ > --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 -- Andy Tan [EMAIL PROTECTED] -- http://www.fastmail.fm - Faster than the air-speed velocity of an unladen european swallow ___ --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
Re: [Asterisk-Users] billing - different tarif per phone
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 ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing & Calling Cards www.aleph-com.net/astpp ___ --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
Re: [Asterisk-Users] Billing inbound calls per minute
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},${CALLERIDNUM},4)
; must be "h,1" as per Michiel van Baak note(above).
exten => h,2,Hangup
[internal]
; i.e. 360 1234567 = DID = card
exten => 3601234567,1,Macro(stdexten,3601234567,sip/did_owner)
[default]
include => internal
[personal]
exten => t,1,Hangup
include => inbound
Result:
- ANSWEREDTIME is OK
- inbound call billed on the callee
- there is CALLERID(name) for callerid in the cdrs(kind of)
There is still a small but "looong" problem - Timeout about 10
secs long while the IAX2/incoming Hangup in personal,t,1.
But CDR is updated after that and the call is billed as expected.
Sorry for the long explanation.
What do you think? Is there something "suspicious" in
that solution?
Thanks,
benchev
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Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing & Calling Cards
www.aleph-com.net/astpp
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Re: [Asterisk-Users] Billing inbound calls per minute
> >>>On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
> I've been playing with astcc, but while
> 'billseconds' stays empty, 'billcost' has
> strange behavior - either stays ampty
> or takes ONCE the "Connect fee"(if I put one)
> and keeps it that way no matter how long
> the call is ...( if no "Connect fee" -stays empty).
> i.e.
> [inbound]
> exten => 1122334455,1,Set(CALLERID(number)=${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},${CALLERIDNUM},4)
; must be "h,1" as per Michiel van Baak note(above).
exten => h,2,Hangup
[internal]
; i.e. 360 1234567 = DID = card
exten => 3601234567,1,Macro(stdexten,3601234567,sip/did_owner)
[default]
include => internal
[personal]
exten => t,1,Hangup
include => inbound
Result:
- ANSWEREDTIME is OK
- inbound call billed on the callee
- there is CALLERID(name) for callerid in the cdrs(kind of)
There is still a small but "looong" problem - Timeout about 10
secs long while the IAX2/incoming Hangup in personal,t,1.
But CDR is updated after that and the call is billed as expected.
Sorry for the long explanation.
What do you think? Is there something "suspicious" in
that solution?
Thanks,
benchev
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Re: [Asterisk-Users] Billing inbound calls per minute
> >On Monday 06 February 2006 09:25, JP Carballo wrote:
> >>
> >>
> >>ASTCC works fine here. The duration and billseconds fields in my cdrs as
> >>well as ASTCC's cdr are filled.
> >>I don't use the connect fee field though and all are set to 0.
> >
> >Would you share with me how'd you do billing on a DID
> >(if you do), and through what Technology?
> >Anything that goes Local here is ANSWEREDTIME zero.
> >Thanks,
> >benchev
>
> That probably explains it.
> IIRC, from when I was still testing ASTCC, when calling a Local channel,
> the AGI suffers from short term memory loss and forgets the values of
> channel variables even if "/n" is used in the dial string.
> I checked my test server logs and while I can verify that ASTCC's CDR
> does have blank duration and billsec fields for the Local calls, *'s CDR
> records them.
Similar here, and I read the patch from Darren May, 2005
where "Local/$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,${CARDNO})
> exten => s,n,Goto(nf2xsubmenu,s,1)
>
> All my calls are routed to IAX2 or SIP or Zap.
And this is my problem because my target is to use Local, but
please follow my answer, within that thread, to Darren.
Thanks very much for your help.
benchev
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Re: [Asterisk-Users] Billing inbound calls per minute
Are you running a relatively recent version of ASTCC? Say within the
last 6 months. The answeredtime = 0 bug was supposed to have been fixed
by http://bugs.digium.com/view.php?id=4300 Unless something has changed
in Asterisk that affects this
[EMAIL PROTECTED] wrote:
On Monday 06 February 2006 09:25, JP Carballo wrote:
Michiel van Baak wrote:
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the "Connect fee"(if I put one)
and keeps it that way no matter how long
the call is ...( if no "Connect fee" -stays empty).
i.e.
[inbound]
exten => 1122334455,1,Set(CALLERID(number)=${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
[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.
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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
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},${EXTEN},4)
>
> ASTCC works fine here. The duration and billseconds fields in my cdrs as
> well as ASTCC's cdr are filled.
> I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
Thanks,
benchev
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Re: [Asterisk-Users] Billing inbound calls per minute
> >>Does anyone have a neat idea as how to
> >>bill inbound calls per minute(second) real time?
> >>
> >>I've been pplaying with astcc, but while
> >>'billseconds' stays empty, 'billcost' has
> >>strange behavior - either stays ampty
> >>or takes ONCE the "Connect fee"(if I put one)
> >>and keeps it that way no matter how long
> >>the call is ...( if no "Connect fee" -stays empty).
> >>i.e.
> >>[inbound]
> >>exten => 1122334455,1,Set(CALLERID(number)=${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
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},${EXTEN},4)
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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Re: [Asterisk-Users] Billing inbound calls per minute
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
> Hi,
> Does anyone have a neat idea as how to
> bill inbound calls per minute(second) real time?
>
> I've been pplaying with astcc, but while
> 'billseconds' stays empty, 'billcost' has
> strange behavior - either stays ampty
> or takes ONCE the "Connect fee"(if I put one)
> and keeps it that way no matter how long
> the call is ...( if no "Connect fee" -stays empty).
>
> i.e.
> [inbound]
> exten => 1122334455,1,Set(CALLERID(number)=${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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Re: [Asterisk-Users] billing system
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! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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 ___ --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
Re: [Asterisk-Users] Billing/SPA-841/CDR Log
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 have at disposal the cdr line in the db, the actual INSERT is done at the end of all extension processing (lost a day trying to figure out what's wrong with an agi before understanding that) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing/SPA-841/CDR Log
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 placed more than 2 simultaneous calls, then I should bill the user for an additional account or something like CEILING(n/2) accounts where n is the number of simultaneous calls.Note that a simultaneous calls is a call which overlaps the another call. For example, call A starts at 10:30AM and lasts 15 minutes. Call B starts at 10:34AM and lasts for 20 minutes. Call C starts at 10:44AM and lasts for 3 minutes. In this case, I should bill the user for 2 accounts plus usage.Does astbill allow me to do this?Thanks,WaldoOn Oct 11, 2005, at 6:17 AM, Are wrote: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 a specific timeframe? That way, I could potentially bill the client for 2 accounts instead of 1. AstBill is a FREE real time billing engine for Asterisk. By using AstBill to process your Asterisk calls you get real time credit control of your customers. When any SIP or IAX client places a call AstBill will do a credit control. If the account has Sufficient founds to place a call then AstBill will calculate the amount needed for the duration of the call. (By default calls disconnect after max 60 minutes. You can change that.) AstBill will create a record in the MySQL table 'astcreditres' with the uniqueid, user and MAX COST of the call. Next time somebody are using the same account to make a call AstBill will check that there are founds available for the customer and deduct any entry reserved for that customer in the MySQL table 'astcreditres' before AstBill decide the max length of the call the client is allowed to use.The CDR in AstBill have two parts. When the call is initiated the CDR info available is stored in the table 'astcdr'. When the call is terminated the rest of the information about the call is updated in the CDR. The cost of the call is calculated and deducted from the customers balance. The record for reserved founds in 'astcreditres' is deleted There are some advantages to this as it allows us to query the database at any time to know how many calls are currently being processed without using the Asterisk Manager Interface. I hope this is useful information and if you have more questions about AstBill please feel free to use the forums at http://astbill.com/forum/3 Please download AstBill now and have a look. http://astbill.com Are Casilla http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and Drupal Consultants http://astbill.com - Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing/SPA-841/CDR Log
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 _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 that extension ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing/SPA-841/CDR Log
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 a specific timeframe? That way, I could potentially bill the client for 2 accounts instead of 1. AstBill is a FREE real time billing engine for Asterisk. By using AstBill to process your Asterisk calls you get real time credit control of your customers. When any SIP or IAX client places a call AstBill will do a credit control. If the account has Sufficient founds to place a call then AstBill will calculate the amount needed for the duration of the call. (By default calls disconnect after max 60 minutes. You can change that.) AstBill will create a record in the MySQL table 'astcreditres' with the uniqueid, user and MAX COST of the call. Next time somebody are using the same account to make a call AstBill will check that there are founds available for the customer and deduct any entry reserved for that customer in the MySQL table 'astcreditres' before AstBill decide the max length of the call the client is allowed to use. The CDR in AstBill have two parts. When the call is initiated the CDR info available is stored in the table 'astcdr'. When the call is terminated the rest of the information about the call is updated in the CDR. The cost of the call is calculated and deducted from the customers balance. The record for reserved founds in 'astcreditres' is deleted There are some advantages to this as it allows us to query the database at any time to know how many calls are currently being processed without using the Asterisk Manager Interface. I hope this is useful information and if you have more questions about AstBill please feel free to use the forums at http://astbill.com/forum/3 Please download AstBill now and have a look. http://astbill.com Are Casilla http://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk and Drupal Consultants http://astbill.com - Billing, Routing and Management software for Asterisk and VOIP AstBill DEMO: http://demo.astbill.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing/SPA-841/CDR Log
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 that extension ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing/SPA-841/CDR Log
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 only when certain calls terminate, I would write my custom CDR. you could use the cdr_custom module in CVS head. additionally, asterisk stores cdrs in Master.csv (all calls) as well as separate files based on account code. you could define a specific account code and only parse that file. customer is able to, potentially, establish 4 simultaneous calls and I'm only billing for one account. Is there a way to restrict the SPA-841 from Asterisk so that I don't depend on Line 2 being disabled on the SPA-841 (which the client could always change)? even if there's one account but 4 calls being made, you'd still see the cdr for all 4 calls. bill according to that. 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 you could limit simultaneous calls using the SetGroup() application. see voip-info.org wiki. -- Regards, /\_/\ "All dogs go to heaven." [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo "The opinions here in no way reflect the opinions of my $a $b." | | done; done | +=+ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing: amaflags and accountcode
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 about 10 SIP phones for different users defined in sip.conf, each with their own accountcode= entry. There is a global setting in sip.conf that states amaflags=documentation There are 3 IAX->PSTN gateways defined in iax.conf for outbound calls. These do not have an accountcode=, but do have amaflags=billing defined in each. The theory was that all calls should be logged, those calls either incoming or between SIP users should have amaflags=documentation (which they do, all well and good), but when a user makes an outgoing call via an IAX gateway, it gets amaflags=billing (so I know it's a chargeable call). However, this doesn't seem to work - all call logs, even those to the IAX gateways all have amaflags=documentation. Is there another way around this? How are you good people using amaflags and accountcode to apportion billing to different users, whilst not "billing" them for incoming calls or calls between SIP users? Thanks in advance. Regards, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing - Disable accounts when balance gets 0 value
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 (new ways for doing so) or programing ideas too once billing server has not the same public IP than Asterisk server. I ll appreciate your comments ok. I use ser+radius to do authentication, this way I can disable users or groups of users in a "standard" way, without using tricks like changing passwords. (when your customer pays he expect to have the same password as before, have you saved it ? where ? in a safe way ?) radius has a mysql backend, so also no need to reload config files. Asterisk and radius share the same db, with some not-too-complex agi before the actual Dial you can do stuff like setting the call timeout based on the remaining credit, blocking the call if the credit is too much in the red, and so on... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing works but i do no get full calling cost...!
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 http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing works but i do no get full calling cost...!
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 posting this, but most astcc billing issues i searched refer to the case when no billing occurs at all. In my case i get only initial charges and any subsequent minute does not count for billing. In my iax.conf i entered the "notransfer = yes" but nothing changed. My test card and test calls are the following TEST-CARD en N/A N/A 6 0 0 50 ^02* TRUNK-G1 500 10 25000 Caller*ID Called Number Trunk Disposition Billable Seconds Billed Cost "3600" <3600> 02203568459 TRUNK/G1ANSWER 105 502 "3600" <3600> 02203568459 TRUNK/G1ANSWER 11 502 "3600" <3600> 02203568459 TRUNK/G1ANSWER 31 502 "3600" <3600> 02203568459 TRUNK/G1ANSWER 79 502 "3600" <3600> 02203568459 TRUNK/G1ANSWER 252 502 "3600" <3600> 02203568459 TRUNK/G1ANSWER 76 502 "3600" <3600> 02203568459 TRUNK/G1ANSWER 233 502 "3600" <3600> 02203568459 TRUNK/G1ANSWER 126 502 Any ideas? Thank you, Panos. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing
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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Billing
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 system that is more fault tolerant and can scale better. While I havent looked specifically at astcc it would need to ensure that concurrent calls from the same account dont end up going over available minutes if prepaid. This can be accomplished by locking rows in the database, pulling a certain amount of minutes from the database (perhaps into a temp table incase something breaks) etc. Then at regular intervals pull more minutes or drop the call if none are present. I dont know how astcc deals with this particular issue. A scalable solution with redundancy could be implemented with astcc based on an overview of what it is. The fact that you have a realtime database for queries on calls could mean that you can have a easier time with a web interface than batch processing radius accounting logs later. It would also offer a prepaid solution, which would be almost impossible with radius alone. On Tue, 2005-04-19 at 09:16 -0700, Sathya Weerasooriya wrote: > Maxim, based on the info in the URL below, you claim to say that completely > asterisk based solution for calling card application may not scale. You > suggest that the alternative is to use gnugk just to use its AAA, or Radius. > In my opinion and experience, I would say by introducing Gnugk and OH323, > you take more horsepower out of the Server that you are running the "calling > card application". > I believe you can do lot better even with an application like ASTCC. Better > mean you will be able to handle more calls in the same box. I think one of > the best ways to handle large call volume is to make sure that asterisk do > the minimum and essential work and build your network around it. If you can > set up asterisk to Answer SIP calls, Authenticate the user based on mysql > database and then route, again using SIP with codec pass through, that will > be the most minimum and efficient way to use asterisk. This kind of setup > with a powerful processor based box, can easily handle 100 + concurrent > calls with millions of minutes. Then you face the situation where, your most > terminating parties are h.323. At this point is where a Cisco 2600XM come in > handy. Also, now you want all front end work to be done via web interface, > which would give customers real time account recharge, cdr etc. For that you > could have a backup mysql with replication. Your website would be talking to > this replicated server for real time data. I think this kind of a solution > can compete with most medium to large calling card systems out there. > > Cheers > > Sathya > > -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Billing
Maxim, based on the info in the URL below, you claim to say that completely asterisk based solution for calling card application may not scale. You suggest that the alternative is to use gnugk just to use its AAA, or Radius. In my opinion and experience, I would say by introducing Gnugk and OH323, you take more horsepower out of the Server that you are running the "calling card application". I believe you can do lot better even with an application like ASTCC. Better mean you will be able to handle more calls in the same box. I think one of the best ways to handle large call volume is to make sure that asterisk do the minimum and essential work and build your network around it. If you can set up asterisk to Answer SIP calls, Authenticate the user based on mysql database and then route, again using SIP with codec pass through, that will be the most minimum and efficient way to use asterisk. This kind of setup with a powerful processor based box, can easily handle 100 + concurrent calls with millions of minutes. Then you face the situation where, your most terminating parties are h.323. At this point is where a Cisco 2600XM come in handy. Also, now you want all front end work to be done via web interface, which would give customers real time account recharge, cdr etc. For that you could have a backup mysql with replication. Your website would be talking to this replicated server for real time data. I think this kind of a solution can compete with most medium to large calling 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/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 update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing
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 answered.
e.g. If I dial on a Zap Channel, Zap answers the call the moment the
channel starts ringing. So I get an answeredtime even if there has
only been ringing.
Has anyone encountered this before?
This is the way it works with ANALOG FXO ports.
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Re: [Asterisk-Users] Billing
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 update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing
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 09:33, 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 answered.
> e.g. If I dial on a Zap Channel, Zap answers the call the moment the
> channel starts ringing. So I get an answeredtime even if there has
> only been ringing.
>
> Has anyone encountered this before?
>
> Regards
>
> riz
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Re: [Asterisk-Users] Billing - which program are you using?
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 own. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgp8BioDMoC7L.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing - which program are you using?
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 is useful for that? > All what I have seen are not based on CDR, but on Radius. > > What are you using? > > bye > > Ronald > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing of outoging calls via CAPI
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 difficult to have a standalone app catch AOC info while Asterisk needs the ISDN link also. You can ask junghanns.net of they are willing to add AOC support to chan_capi (Possibly at a fee). Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing of outoging calls via CAPI
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 a database if you want it there). > Can I obtain in * an accurate charge information of outgoing call via CAPI > which > destination is any PSTN number? I think you are referring to AOC (Advice of Charge) and afaik chan_capi does not support that nor does the CDR generation code. > Does the ISDN "signal" contain charging informations ? Depends on your telco but if Asterisk does not support AOC, it will not matter... Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing of outoging calls via CAPI
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 ISDN, I apologize that I can't help so much...) P 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. Can I obtain in * an accurate charge information of outgoing call via CAPI which destination is any PSTN number? Does the ISDN "signal" contain charging informations ? Thank for advice Rastislav ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- There are 10 kinds of people in the world, those who understand binary, and those who don't. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billing (itemized) in the UK
Solution found!
I connected a recording device to the line and called the same number (near London) from different
devices. The recording I played into an audio editor (cool edit) and played the files with a program
that could decode DTMF signals. The program then showed the numbers I dialed. Except for the
recording from one or two devices (missed digit / double digit => audio not clean) with every
device the dialled number was:
1666
The first tests failed because I tried a local number and we have least-cost-routing as well, so
that 1666 only works with non-local numbers.
So many thanks for all your help, esp. Tim Robinson's comment which brought me on the right track!
Also for those who gave me different dial commands - I will look into that. We should now be able to
connect asterisk to the PSTN line and still get the itemized bills from our telco as before.
One post asked why we wouldn't move over to ISDN - the reason is that our analog lines have so far
sufficed for us in every respect and I try to follow the maxime: "If it ain't broke, don't fix it."
Peter
Peter Hoppe wrote:
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 my digits are audible. Near the pbx I have noticed a
small white box saying 'Smiths communications' and 'SC14' on the lid.
The box is connected to two cables - one to a power supply, the other is
a 4 pair telephone installation cable with 3 pairs connected. Next to
the box is a switch with some labels on it: one label says 'LINE 1'. The
other two labels describe the switch settings - 'SYSTEM' and 'A/PH MOD'.
I have the suspicion that the white box has something to do with the
billing and that it sends some fast data over one of the lines when an
outside call is initiated, but I am not sure. I'll continue to hunt.
I also asked the telecom provider but they were not very helpful and
couldn't (or didn't wish to)
give me any information as to the technical details.
I'll hunt on...
P
Robinson Tim-W10277 wrote:
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${EXTEN:1})
This will look for 9, then a local number beginning 2,3,4,5,6,7,8 , and
dial out the extension number, followed by the 0113 area code.
You will need to make sure that 999 and 112 go direct to BT by using
another line in the extensions file. E.g.
exten => ,1,Dial(Zap/g1/999)
exten => 9112,1,Dial(Zap/g1/112)
And probably
exten => 999,1,Dial(Zap/g1/999)
Just to be on the safe side!
You could also write a little macro to kick another user off their call
to allow the emergency call to get priority.
There is just so much cool stuff you can do. But do test well!
Rgds
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Hoppe
Sent: 25 November 2004 13:34
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Billing (itemized) in the UK
If the protocol is correct, I could construct a dial command such as
exten => _9.,1,Dial(Zap/g1/1666${EXTEN:1})
or so - I would just need a way to construct - and then any caller
from an inside device would just prepend a '9' before the real number.
I probably would also bar
simple '9' dialling to get an outside line... lets see.
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Re: [Asterisk-Users] Billing (itemized) in the UK
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
my digits are audible. Near the pbx I have noticed a small white box saying 'Smiths communications'
and 'SC14' on the lid. The box is connected to two cables - one to a power supply, the other is a 4
pair telephone installation cable with 3 pairs connected. Next to the box is a switch with some
labels on it: one label says 'LINE 1'. The other two labels describe the switch settings - 'SYSTEM'
and 'A/PH MOD'. I have the suspicion that the white box has something to do with the billing and
that it sends some fast data over one of the lines when an outside call is initiated, but I am not
sure. I'll continue to hunt.
I also asked the telecom provider but they were not very helpful and couldn't
(or didn't wish to)
give me any information as to the technical details.
I'll hunt on...
P
Robinson Tim-W10277 wrote:
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 extension number, followed by the 0113 area code.
You will need to make sure that 999 and 112 go direct to BT by using
another line in the extensions file. E.g.
exten => ,1,Dial(Zap/g1/999)
exten => 9112,1,Dial(Zap/g1/112)
And probably
exten => 999,1,Dial(Zap/g1/999)
Just to be on the safe side!
You could also write a little macro to kick another user off their call
to allow the emergency call to get priority.
There is just so much cool stuff you can do. But do test well!
Rgds
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Hoppe
Sent: 25 November 2004 13:34
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Billing (itemized) in the UK
If the protocol is correct, I could construct a dial command such as
exten => _9.,1,Dial(Zap/g1/1666${EXTEN:1})
or so - I would just need a way to construct - and then any caller
from an inside device would
just prepend a '9' before the real number. I probably would also bar
simple '9' dialling to get an
outside line... lets see.
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RE: [Asterisk-Users] Billing (itemized) in the UK
You just need to do something like
exten => _9.,1,Dial(Zap/g1/1666$CALLERIDNUM${EXTEN:1})
You can also do some useful translations like
exten => _9[2-8]XX,1,Dial(Zap/g1/1666$CALLERIDNUM0113${EXTEN:1})
This will look for 9, then a local number beginning 2,3,4,5,6,7,8 , and
dial out the extension number, followed by the 0113 area code.
You will need to make sure that 999 and 112 go direct to BT by using
another line in the extensions file. E.g.
exten => ,1,Dial(Zap/g1/999)
exten => 9112,1,Dial(Zap/g1/112)
And probably
exten => 999,1,Dial(Zap/g1/999)
Just to be on the safe side!
You could also write a little macro to kick another user off their call
to allow the emergency call to get priority.
There is just so much cool stuff you can do. But do test well!
Rgds
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Hoppe
Sent: 25 November 2004 13:34
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Billing (itemized) in the UK
If the protocol is correct, I could construct a dial command such as
exten => _9.,1,Dial(Zap/g1/1666${EXTEN:1})
or so - I would just need a way to construct - and then any caller
from an inside device would
just prepend a '9' before the real number. I probably would also bar
simple '9' dialling to get an
outside line... lets see.
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RE: [Asterisk-Users] Billing (itemized) in the UK
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 => _07.,1,Dial,${TRUNK}/BYEXTENSION
exten => _08.,1,Dial,${TRUNK}/BYEXTENSION
exten => _09.,1,goto(nogo,1)
You dont need a 9 for a line, you couls also add lines for barred numbers
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Peter Hoppe
Sent: 25 November 2004 13:34
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Billing (itemized) in the UK
Thank you very much for the answer! I think it is a good path to look at. I
have had a look through
our paperwork for the present pbx, and I found one document that seemed to
indicate we have to dial
1666
to give the extn info to the telco. The paper is a bit old (1999) and since
then we have changed our
telco, but I guess that this protocol is still valid. This afternoon I will
hook up a recording
device on the line and see which digits are actually dialled when I dial an
outside line. From the
recording I should be able to reconstruct which digits have actually been
dialled by the pbx.
If the protocol is correct, I could construct a dial command such as
exten => _9.,1,Dial(Zap/g1/1666${EXTEN:1})
or so - I would just need a way to construct - and then any caller from
an inside device would
just prepend a '9' before the real number. I probably would also bar simple
'9' dialling to get an
outside line... lets see.
Keep you posted, and so many thanks for all the help!
P
> 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
> 2) access code1xxx
>
> You need to get the specs for this from Your Communications. It is not
> clear from the web site...
>
> Asterisk will cope perfectly with either solution - you will just need
> to fiddle a bit with the dial plan. Once we know what you have to send
> to the telco there are tons of people here who will advise on the Dial
> command you should use to achieve what you want.
>
> Rgds
> Tim Robinson
> Ps. Any reason why you chose to stick with the analogue solution? Is
> this just risk mitigation in the early stages? (this is a valid reason,
> btw!)
>
>
>
> -Original Message-
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Peter
> Hoppe
> Sent: 25 November 2004 10:54
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Billing (itemized) in the UK
>
>
> Hello!
>
> We are located in the UK, and we are planning to replace our old pbx
> with an asterisk based pbx. For
> outgoing calls our present pbx is connected to three PSTN lines which
> all have the same number.
> Internally, the pbx caters for quite a few extensions, and each
> extension can make outbound phone calls.
>
> Our telecom provider (your communications) gives us monthly itemized
> bills that list all of the
> calls per extension, i.e. from the bill we are able to tell which
> internal extension made what call
> to which destination at which date/time, how long this call was in
> minutes and how much that
> particular call costs.
>
> We would like to reuse the three PSTN lines with the asterisk system,
> and at present there are no
> plans to utilize other connectiviy (such as ISDN) - we would like to
> stick with the three PSTN lines.
>
> My understanding is that when the asterisk system is running we won't
> get any itemized bills any
> more since the telecom provider has no way of telling from which
> extension a call originated.
>
>
> Questions:
>
> To give the extension information to the telco...
>
> How can I configure Asterisk to do send extension information?
>
> What signalling do I have to provide for outgoing calls to give
> extension information the telco?
>
> Is there a standard for sending extension numbers (i.e. do I have to
> send some DTMF digits)?
>
> Is there a software / asterisk extension (that works in the UK) that
> allows asterisk to send
> extension info?
>
> Do I need to buy some equipment that can provide this info to the telco?
> Which?
>
> Where could I find more information on that subject?
>
>
>
> Thank you very much for your consideration.
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RE: [Asterisk-Users] Billing (itemized) in the UK
>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] What PBX are you using currently? >Is there a standard for sending extension numbers (i.e. do I have to send >some DTMF digits)? > [Senad Jordanovic] On POTS lines no. On BRI/PRI yes... > >Where could I find more information on that subject? > [Senad Jordanovic] Try http://www.voip-info.org/tiki-index.php?page=Asterisk Senad Jordanovic Bicom Systems, The complete systems provider www.bicomsystems.com USA 1-212-400-7921 UK 0870 682 782 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
