similar to: LCR / Trollphone Rate Engine

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

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

Has anyone had any luck getting the rate engine app from trollphone.org to work? I've been trying to find documentation on how to set up the information in the mysql tables. Could anyone point me in the right direction? -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V.

Is anyone familiar with the Trollphone's LCR package? There is a field in the egress table labeled substitue. Placing a $1 there results in the correct dial extension being passed. However how is this field used to substitute replacement dial extensions... in other words as an example, lets say my EXTEN as passed to and back from the rate_engine is 01144123456789 and I want to have a

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking

Hi, After a few extra days of hard work, debugging, and many coffees, I am pround to announce that Asterisk::LCR has been released on CPAN. Asterisk::LCR is an open-source, Perl-based collection of tools to help you manage efficiently multiple VoIP providers with your Asterisk installation. It is capable of importing providers rates from multiple providers, comparing these rates, and

Hi all, I am planning to implement LCR routing on my already running asterisk server. Uptill now i have found out that asterisk has no support for lcr, i have to do something about it myself, for example using the AGI. Im looking for ideas here. Whats the best way to start implementing lcr in asterisk. Should i use agi and start implementing my own lcr script or is there any plugin available which

Hello, I short question: I want to connect Asterisk to OpenBSC with mISDN, mISDNuser and LCR. Do I need chan_lcr? I have: Asterisk 1.8 mISDN .v2 integrated in Kernel 3.0.22 mISDNuser lcr 1.7 HFC-E1 Evaluation board from cologne chip I tried to configure Asterisk with <./configure --prefix=/usr/src/lcr --with-gsm-bs> and it runs without errors, But make and make install didn't run

I have got Areskicc installed with AMP and I can't stress out how good and excellent this software is. I must say the author desire every credit for this software and I would like to say Thank you to Arezqui who wrote the software. Now there is one last thing before my whole asterisk calling card become prefect. And I am wondering whether anybody may know the answer of this. I

Hi, This is what I want to do: 1. Asterisk to answer calls via DID's, currently using SIPGATE 2. Provide a menu, and allow users to dial out. 3. According to the country and area they dial, the call should connect via one of up 4 carriers depending on cost. 4. If the carrier is busy it should go to the next one in line and so forth. I have tried to set this up, but it never answers the

Hello asterisk-users. I have the following dial-plan: [test] exten => 482,1,Dial(OH323/106@192.168.2.73,10) exten => 482,n,Dial(OH323/102@192.168.2.73,10) exten => 482,n,Dial(OH323/103@192.168.2.73,10) exten => 482,n,Dial(OH323/104@192.168.2.73,10) exten => 482,n,Dial(OH323/105@192.168.2.73,10) exten => 482,n,Dial(OH323/106@192.168.2.73,10) When I call exten =>

Dear All, I have two questions. 1. How can I distribute equally the outgoing calls between channels in a group? I'd like to send outgoing calls on 2 ISDN GSM adapter. I need to use the GSM subscriptions approximately equally. 2. When all of the channels are busy in the first outgoing channel group. How can I send out the call on another group? N+101 is for this problem? Regards, Repas

Hi all, Got an interesting question here. We have 2 incoming ISDN channels (linked to a local phone number that our customers call), and an account with a local IP provider. We use the ISDN for local, 13 calls - 13 is a call who's destination depends on the originating call (no point putting it through the IP provider as it will think we are calling from the wrong city and may not get

Ladies and Gents.... Please be patient as I try to explain what I am trying to achieve.. I have a PSTN line and a Freshtel account, what I want to do is have the PSTN line as the first choice for outgoing calls for local calls and Freshtel as the second choice. The problem is that it's easy enough to set up both individually but how do I get the "second choice" drop the leading

Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random "calls" so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? Thanks, Daniel

Hiyall. Just wondering how people test your emergency dialing in the UK. Obviously you need to dial the 999 for emergency services, but am a bit unsure if this would go down too well with the operator with a 'sorry just testing' call. (you do all /test/ your emergency dialing dont you!?:-) ) As another thing, what is the correct method when using least cost routing... If you have a

Hello, How do you guys implement LCR in Asterisk? Thanks,

Hi List, I am working on least cost routing code on the moment, and I am stumbling on a problem. Say you have provider A having: Prefix XXX 0.10 Prefix XXXYYY 0.20 And provider B having Prefix XXX 0.15 You're stuck, because you cannot decide if provider B's "XXX" prefix also covers XXXYYY numbers or not. If it doesn't, it would be a waste

Ok, so you've read the Wiki and gotten call routing using ENUM to work (http://www.voip-info.org/tiki-index.php?page=Asterisk%20E164%20Call%20Routing) with your own ENUM-alike domain, e164.example.com. But how do you populate it with data? You can do it manually, but that gets very tedious very quickly. Or you can use the nifty DDNS updating program that comes with bind9. The first thing is

Hi all, Here''s my situation: I have a little route creation app that lets a user create a route. I''m using the mapstraction version of ym4r... here is the workflow: first page has a map displaying the start point of existing routes. if they click "new route: creates a route_object clears all polylines in case there are any clears all the markers and adds an onclick