[asterisk-users] Billing for minutes
All; I was wondering what people are doing to bill customers for minutes. I know that A2Billing is a popular option, but I was wondering if there are other good alternatives. They don't necessarily have to be free, but they need to be cost effective. Any insight at all would be greatly appreciated. Thanks; John V. -- _ -- 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
[asterisk-users] Billing
Hi I have the following topology: Sever A IAX2 - SERVER B SIP ITSP Server A : Branch Server ( with billing cdr ) : Billing module for auditing calls at branch level Server B : Billing Server ( with billing cdr ) : Official statements get generating from this server I am trying to match my billsecs on server A TO that of server B The problem seems to be that server A is counting the ringing as well, not just the ANSwered ( person talking ) If the phone rings for 25 secs and we speak for 30 secs, billsec is 55 secs and duration is 55 secs CDR values for deposition are correct if if the call has not been answered or busy. ie NO ANSWER or BUSY on Server A. Asterisk 13 on branch server any ideas on how to trouble shoot this problemthanks Zakir -- _ -- 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 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
[asterisk-users] Billing software: Other than A2Billing because of the problem with the analogue channels
Hello; I am facing a trouble with A2Billing when using analogue lines because the channels are not closing properly when dialing happen through A2Billing (it seems the dialing scenario including the hangup is not handled properly through A2Billing but I do not have control on this). But when I do dialing from asterisk and using analogue lines, I do not face a trouble because I can write the script in the extensions.conf in professional way to confirm that the channel is closed successfully. Is there alternative Billing solution than A2Billing which has another working mechanism? How I can resolve such problem which is related to the analogue channels? Regards Bilal-- _ -- 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
[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 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 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: sip:2325551212@1.2.3.4 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
Gateway computers rejects calls like this. I was informed that their carrier rejects the calls because they cannot accurately bill. It seems pretty silly with voip and number portability. Thanks, Steve T On Mar 17, 2014 5:19 PM, Eric Wieling ewiel...@nyigc.com wrote: Often it is 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 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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[asterisk-users] billing based on local access number
Hi all, I am configuring asterisk as a prepaid calling card. I am getting different local rate from my ISDN provider e.g 0.002 for landline and 0.13 for mobile etc. In this case I thing I have to say my asterisk/a2billing to bill based on local access number. so How can I retrieve called number (eg. 03-6832-1040 and 0120-272-060 is our ISDN PRI access number) to my asterisk server so i can trigger different rates. Regards Umesh _ Hotmail: Powerful Free email with security by Microsoft. http://clk.atdmt.com/GBL/go/201469230/direct/01/-- _ -- 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 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
[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, ___ -- 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
[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
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
[asterisk-users] Billing software for 1.6?
Hi, is there a Asterisk 1.6 billing software that i can use? Prepaid supporting ones are more acceptable for me right now but, post-paids are also welcome if they are available. As i see most of the softwares are designed for 1.2 and 1.4 ___ -- 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
[asterisk-users] Billing and Soft Switch.
Looking for a Free VOIP Billing and Soft Switch. Any suggestions ? ___ -- 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 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/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
[asterisk-users] Billing/Call Control engine : AGI scripts/ AstMan API
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
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 ___ --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
[asterisk-users] Billing Telephone Number (BTN)
I have Asterisk setup with two PRI's one going to my telco and the other going to a Norstar Meridian system. The Norstar Meridian is sending a BTN number that needs to be passed to the Telco. Is there a way to pass the BTN as a variable in the dial plan? Like CallerID(num)? What is the variable for BTN if so? Many Thanks. -- *** Forrest Beck IAXTEL: 17002871718 [EMAIL PROTECTED] ___ --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 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
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 asterisk-users@lists.digium.com Sent: Thursday, February 08, 2007 3:13 PM Subject: Re: [asterisk-users] Billing pulses I must clarify my original message. Maybe confusion is due to my poor english. So I'll 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
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
[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
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 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
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
From:Jorge Mendoza [EMAIL PROTECTED]Funny that a digital line have a analogue pulse.Normally the billing pulse is used on payphones. IMO you only need the answer supervision to trigger your own billing system.Jorge MendozaStefano Corsi wrote:Hello,I've discovered that in Italy ISDN lines can be 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 thatsome ISDN interface cards can use them. (Just googled to confirm that somenon-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. Ifthey aretruly 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 mspause duration 180 msperiod ... 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?ThanksStefano ___ --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 asterisk-users@lists.digium.com 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
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
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
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 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 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
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 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
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
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
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
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
[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 ___ --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 asterisk-users@lists.digium.com 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
[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
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 asterisk-users@lists.digium.com 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
[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
[asterisk-users] billing
Dear How can I charge the incoming call to the destination call ,using a2billing I used to make setaccount but it didnt work such a loopback detected Please help * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
[asterisk-users] billing
Dear How can I charge the incoming call to the destination call ,using a2billing I used to make setaccount but it didnt work such a loopback detected Please help * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
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
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
[asterisk-users] Billing Solution ?
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 ?
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 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 ?
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
[asterisk-users] Billing
Dear I am using a2billing accounting software, how can I charge on the destination target not at the caller side Ex: if user A have 10$ and user B have 10$ ,and the onnet call charge cost 1$ When user A call user B for 1 minute ,user A amount remains 10$ and user B amount be 9$ Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
[Asterisk-Users] billing
I am using trixbox ,please any ont knows how to confiure billing on it, I want to make a billing ,I created an account at http://x.X.X.X/a2billing but it does work * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
[Asterisk-Users] Billing when forwarding incomming calls from SIP phone
Hi All Does anyone know how to track calls that come in from the PSTN, go to the user SIP phone and then get forwarded from the user SIP phone back out to PSTN (typically the users mobile number)? It is a headache trying to bill the users who do this. There is only the src PSTN number and dst forwarded number in the CDR. No SIP details at all. How do we track this? Billing for standard calls from the SIP user to the PSTN is fine. Regards Garth ___ --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 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
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 usedthat took out the flag in astcc that kept track ifthere was a call in progress for that pin or not. Dontknow if it wil work for real time though. DovidI don't know if you were pertaining to what I posted in the messageASTCC: 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 listTo UNSUBSCRIBE or update
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 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
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 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 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
-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
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
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
[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
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
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
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
[Asterisk-Users] billing realtime
Hi all I think this could be en old question. I would like to do a realtime billing prepaid system, mainly using asterisk. I have found few things; I can not get CDR function into agi because asterisk set them once the call is absolutely finish (at least main values for the main porpouse, billsec,duration, etc..) There is a patch that allow you to use CDR function in hangup extension, but it seems to have some troubles, haven't it?? Finally, another solution I have found reading somewhere could be use triggers in the database when there is and CDR insert. A cron job could do something similar. Now, the question, can I access somehow in a deadagi, or whatever the CDR function in order to update the credit when the call has just finished. Thank you very much ___ --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
[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
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 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 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
[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
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
[Asterisk-Users] Billing from CDR files
I have a Astlinux system that can't run mysql. I want to bill calls by extension but all I have is the raw CDR files. Is there any software that cal produce a simple call accounting from text CDR files? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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
[Asterisk-Users] billing - different tarif per phone
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
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
aZOMBIE 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},${CALLERIDNUM},4)
; must be h,1 as per Michiel van Baak note(above).
exten = h,2,Hangup
[internal]
; i.e. 360 1234567 = DID = card
exten = 3601234567,1,Macro(stdexten,3601234567,sip/did_owner)
[default]
include = internal
[personal]
exten = t,1,Hangup
include = inbound
Result:
- ANSWEREDTIME is OK
- inbound call billed on the callee
- there is CALLERID(name) for callerid in the cdrs(kind of)
There is still a small but looong problem - Timeout about 10
secs long while the IAX2/incoming Hangup in personal,t,1.
But CDR is updated after that and the call is billed as expected.
Sorry for the long explanation.
What do you think? Is there something suspicious in
that solution?
Thanks,
benchev
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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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Re: [Asterisk-Users] Billing inbound calls per minute
On Monday 06 February 2006 09:25, JP Carballo wrote:
snip
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
Thanks,
benchev
That probably explains it.
IIRC, from when I was still testing ASTCC, when calling a Local channel,
the AGI suffers from short term memory loss and forgets the values of
channel variables even if /n is used in the dial string.
I checked my test server logs and while I can verify that ASTCC's CDR
does have blank duration and billsec fields for the Local calls, *'s CDR
records them.
Similar here, and I read the patch from Darren May, 2005
where Local/$phone/$res-{path}|30|HL/n was changed to
Local/[EMAIL PROTECTED]{path}|30|HL/n
snip
I do billing based on account number so clients are free to call from
any phone. I don't check callerid.
Since each account is based on the phone number registered by the
client, I can just chop off the 2 digit prefix and set their callerid
with the result.
Yes, I do that also with another instance of astcc, I call astcc-disa.agi
to allow clients from outside to enter * and do things.
[makecall]
exten = s,1,Set(CALLERID(num)=${CARDNO:2})
exten = s,n,DeadAGI(astcc.agi,${CARDNO})
exten = s,n,Goto(nf2xsubmenu,s,1)
All my calls are routed to IAX2 or SIP or Zap.
And this is my problem because my target is to use Local, but
please follow my answer, within that thread, to Darren.
Thanks very much for your help.
benchev
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Re: [Asterisk-Users] Billing inbound calls per minute
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
I've been playing with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${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
aZOMBIE 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},${CALLERIDNUM},4)
; must be h,1 as per Michiel van Baak note(above).
exten = h,2,Hangup
[internal]
; i.e. 360 1234567 = DID = card
exten = 3601234567,1,Macro(stdexten,3601234567,sip/did_owner)
[default]
include = internal
[personal]
exten = t,1,Hangup
include = inbound
Result:
- ANSWEREDTIME is OK
- inbound call billed on the callee
- there is CALLERID(name) for callerid in the cdrs(kind of)
There is still a small but looong problem - Timeout about 10
secs long while the IAX2/incoming Hangup in personal,t,1.
But CDR is updated after that and the call is billed as expected.
Sorry for the long explanation.
What do you think? Is there something suspicious in
that solution?
Thanks,
benchev
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Re: [Asterisk-Users] Billing inbound calls per minute
On Monday 06 February 2006 09:25, JP Carballo wrote:
Michiel van Baak wrote:
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${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
[EMAIL PROTECTED] wrote:
On Monday 06 February 2006 09:25, JP Carballo wrote:
snip
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
Thanks,
benchev
That probably explains it.
IIRC, from when I was still testing ASTCC, when calling a Local channel,
the AGI suffers from short term memory loss and forgets the values of
channel variables even if /n is used in the dial string.
I checked my test server logs and while I can verify that ASTCC's CDR
does have blank duration and billsec fields for the Local calls, *'s CDR
records them.
If it's also true for you, you might want to use *'s CDRs for rating.
I do billing based on account number so clients are free to call from
any phone. I don't check callerid.
Since each account is based on the phone number registered by the
client, I can just chop off the 2 digit prefix and set their callerid
with the result.
[makecall]
exten = s,1,Set(CALLERID(num)=${CARDNO:2})
exten = s,n,DeadAGI(astcc.agi,${CARDNO})
exten = s,n,Goto(nf2xsubmenu,s,1)
All my calls are routed to IAX2 or SIP or Zap.
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
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Re: [Asterisk-Users] Billing inbound calls per minute
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},${EXTEN},4)
ASTCC works fine here. The duration and billseconds fields in my cdrs as
well as ASTCC's cdr are filled.
I don't use the connect fee field though and all are set to 0.
Would you share with me how'd you do billing on a DID
(if you do), and through what Technology?
Anything that goes Local here is ANSWEREDTIME zero.
Thanks,
benchev
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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing Calling Cards
www.aleph-com.net/astpp
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Re: [Asterisk-Users] Billing inbound calls per minute
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${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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[Asterisk-Users] Billing inbound calls per minute
Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${EXTEN})
exten = 1122334455,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten = 1122334455,3,Hangup
Thanks in advance,
benchev
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Re: [Asterisk-Users] Billing inbound calls per minute
On 00:30, Mon 06 Feb 06, [EMAIL PROTECTED] wrote:
Hi,
Does anyone have a neat idea as how to
bill inbound calls per minute(second) real time?
I've been pplaying with astcc, but while
'billseconds' stays empty, 'billcost' has
strange behavior - either stays ampty
or takes ONCE the Connect fee(if I put one)
and keeps it that way no matter how long
the call is ...( if no Connect fee -stays empty).
i.e.
[inbound]
exten = 1122334455,1,Set(CALLERID(number)=${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=getsearch=0x7E0B9A2D
Why is it drug addicts and computer afficionados are both called users?
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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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