similar to: Self Contained Least Cost Routing Solution

Hello, does anyone know of a good calling card solution for asterisk that is able to do lcr? Does astcc do this? I've been searching around and I can find some lcr modules/apps but none that incorporate prepaid card functionality. Regards, Igor H.

Can someone guide me on what package I can use to do least cost routing in asterisk-1.4 without going through the prepaid calling card platforms. I have tried Asterisk::LCR and LCDial without success, if more help on either too. I will be glad. I will be glad for good pointers. Thanks. Goksie

Greetings. I've been playing with the TrollPhone Rate Engine Addon for a week or so. I'm curious what is being used out there for LCR applications? I've run into a stump with the Rate Engine and that is the costing is done with an integer. With this, how do you put in say 0.014 / minute rate. And if your periods are not 60 seconds.. well.. then you have even more places to

We need an expert in least cost routing (LCR) for an Asterisk project. Please provide references and a resume of your experience. Contact us at voip3@nibble.net.

Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording

Anyone using the rate_engine from TrollPhone? There is absolutly no documentation on how to setup data in the tables. If someone could send sample data, or post it to the wiki, it would be helpful. If any others are successfully using another Least Cost Routing method, please pass it along. THanks, Matthew

I've got two patterns I want to match on making an outgoing call... (one day - to do Least Cost Routing for Cell/Mobile calls) Firstly - I prefer '0' rather than '9' to get an outside line... Either its a call to a mobile No... (072 -or- 082 -or- 083 -or- 084) or its just another number to dial... I added the following... the playback just advises me which 'route' is

Unsubscribe Elvin G. Nodalo -----Original Message----- From: asterisk-users-request at lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit

Unsubscribe Elvin G. Nodalo -----Original Message----- From: asterisk-users-request at lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit

I forgot to pay this month's phone bill, and never noticed until family (the in-laws, who are too cheap to try the cell phone if landline fails, because it is 'more expensive') told me they were unable to reach us... As it turns out, the phone company disconnected us, but because Asterisk routes all outgoing calls in the Netherlands over VoipBuster, I never noticed anything! ;-) If

Hello, We are glad to announce new release of our advanced billing and routing package for Asterisk - MOR v0.7 It is complete solution for VoIP billing and routing for advanced and start-up telecoms, carriers, voip calling card operators and ISPs. Demo available online, as LiveCD or as InstallCD. Contact us for more details. More info: http://www.kolmisoft.com What is new in

Hi! Maybe someone could help me out? When a call is routed via a2billing AGI and user does a transfer, the call is dropped. If the trunk is called directly everyhing works. Here's a direct scenario (working fine): [pbx000001] exten => 101,1,Set(__TRANSFER_CONTEXT=pbx000001) exten => 101,n,Dial(SIP/pozitel/37129238254,45,t) exten => 102,1,Dial(SIP/12345,60) so, when user calls ext

Hello, I'm planning to start using R. Before getting into it, I'd like to ask a couple of questions. Does R carry out loglinear model analysis? That is, will it provide the chi-squared goodness of fit test statistic for a given hierarchical loglinear model? Maybe even do a model selection procedure (like Brown's two-step procedure, or forward/backward selection)? Thanks

Not sure if a2billing can be shared here, but ill give a shot If the credit < min_credit the IVR play: sorry you have 0 credit and hangup, I want it to FW me to the IVR to add voucher, please let me know: here is log: [18/10/2010 07:01:12]:[file:a2billing.php - line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1] [18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE : standard]

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

hi all, I have a little problem. I'm trying to configure a2billing (asterisk2billing) with asterisk. Everything done successfully but when I try to call following error occur "WARNING[9690]: pbx.c:3170 pbx_extension_helper: No application 'DeadAGI,a2billing.php' for extension (a2billing, 456,3) and it hang ups the call. Can someone please tell me why this error occuring. My

I am not that good at regex and it's use in Asterisk. I am running Asterisk 1.4.13 Currently I have this in my extensions.conf for incoming calls on our house phone line: [housemenu] exten => s,1,GotoIF($["${CALLERID(num)}" = "815xxxxxxx"]?s|12); 815xxxxxxx is our home phone number, when caller id fails or is missing that is what is recorded. I want to expand this

Hello, im having trouble with the following: [Asterisk]<------>[ISP]<------>[ADSL Modem]<------>[Linksys Router]<------>[Grandstream ATA]<------>[Analog Phone] On server: - Asterisk 1.6 - A2Billing 1.4 A2Billing have 2 Trunks: - TrExt: Voip Provider - TrInt: Internal Calls This structure works on first day (Asterisk+A2Billing installation/configuration). But on

Hi Everyone, Trying to run a php script after DeadAGI for A2Billing does it's magic. This is the dialplan: [a2billing] exten => _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN} ${UNIQUEID}) exten => _X.,n,AGI(a2billing.php,1) exten => _X.,n,Hangup() *exten => h,1,Wait(5)* *exten => h,n,System(php post-call.php ${CALLERID(num)} ${UNIQUEID})* As you can see above, I even