similar to: Least Cost Routing

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

Hi all, I compiled rate-engine version rate-engine-0.5.2 with Asterisk CVS-HEAD-07/14/04-18:25:12 with some warnings. When I load the asterisk in debug mode, I got an error: [rate_engine.so]Jul 29 13:51:19 WARNING[1024]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/rate_engine.so: undefined symbol: ast_channel_walk Jul 29 13:51:19 WARNING[1024]: loader.c:423 load_modules: Loading

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

Hi, all: I am a newbie to the asterisk and its architecture. :( After reading some help in the tarball of Asterisk, I am still in the mess. So I want to know where I can find a detailed explanation of the Asterisk which including the Architecture, Install, Configure, usage example document. Maybe what I want is too much, after all it is a open project, not commercial product. If I want to get

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

Well, Once upon a time, I had problems receiving callerid, and then one day, Mark was logged into my asterisk box helping with something else, and I asked him about this, and he showed me a nice tweak to some source file that made it work. Some time later, I must have done hundreds of CVS updates, and along the way, lost the above patch (one liner) and so callerID hasn't worked for a long

I've tried to compile rate-engine 0.5.2 on Fedora Core 1, Redhat 9 and OpenNA Linux 1.0 and all give me an "Error 1" after typing "make" but with no real reason given. Just a few standard/non-critical warning messages, and then suddenly "Error 1" Anybody successfully compile Rate Engine? The least cost routing module for Asterisk? Thanks in advance.

Dear Duane: Thanks for the steer (presently I route calls to 1800 via iaxtel, but I'll turn that off for that test) I came up with a thought for an interesting e164 service last night - distributed custom ringtones, and custom announcements on a callerid lookup via enum. You'd be able to embed a RINGTONE - an url pointing to either a ringtone definition or a wav file that contains

works, but there are some issues. I've had asterisk up and running on a suse 9.0 beta 7 x86_64 for about two weeks now. I used then-current cvs, and the compile went smoothly with only a dozen complaints about 64 bit casting of pointer types. The good news : Latency is effectively non-existent (especially when compared to the lame c3 itx box I normally use), and asterisk has not crashed

We currently have an office which has a happily working Asterisk setup and all is good. We are looking to deploy the same solution to our other 5 offices, my question relates to the interconnections between all those sites. The requirement is seamless dialing of extensions and the ability to have members of multiple offices participate in call queues. I wish to keep the configuration as

Guys.. Im new to asterisk but is it possible to simulate a dialtone for example, in other PBX when you pick up the phone you can hear a certain dialup, which is the PBX dialtone, and when you hit 9, you can hear the PSTN dialtone, is this possible? __________________________________________________________________ Anton Krall

Hello all, I am planning to connect my Asterisk with the FWD and/or iaxtel networks. Two mounths ago, I just used the iaxtel network, and i remember I have trouble with this network, I can not place a call. The service do not wotk. With FWD I alwais can place a call, I never get an error from this network. But my experience in both are very short, just a few test time. Could somebody

I'm setting up a private enum zone to simplify/centralize dialplans for a number of Asterisk servers. In several dialed number situations, there are a handful of possible destinations for the call, and I'd like to have * try ENUMENTRY1, ENUMENTRY2, .., ENUMENTRYN just in case the first result is temporarily unable to handle the call. In at least some cases, I'd also like the order in

For those interested, I'm giving a talk about VoIP/enum.164/asterisk tonight in Sydney at the Sydney LUG meeting which is about 7pm in the UTS build #2, 4th floor, room 10. Sorry for the late notice, it didn't occur to me that there might be people on this list interested and able to attend etc... -- Best regards, Duane http://www.cacert.org - Free Security Certificates

e164.org is a public name service which provides ENUM.164, a method devised by the IETF and ITU to allow an ordinary telephone to be connected to an Internet type network and provided dialling service from other, regular telephones. Unlike many other "free" voice over IP systems, e164.org allows users who have a regular telephone line, to also hook themselves up to the Internet

Hi folks, I think my little agi script is ready for the big one-oh-oh. Available at http://muware.com/asterisk is cid_rewrite-1.0.0. This agi-script does the following: - Standardize incoming caller-id numbers to adhere to US dialing code; NANPA numbers are reformatted to 1+10, international numbers become 011<country-code><number> (this is customizable with a little PHP knowledge).

Hi. Thank you to Olle, the Wiki, and Trollphone. I found the Trollphone Rate Engine listed on the Wiki at: http://www.voip-info.org/wiki-Asterisk+addon+rate-engine I'm not familiar with LCR yet, but this is something that I need to do. I have it all installed but am not familiar with the terminology. Can someone help me maybe figure out what some of these fields are for? mysql>

Can someone send me a quick snippet of a dialplan for international dialing via NuFone? I'm having a hard time getting any help from them this week. Scott

Apart from the "virbiage" FT-201 which hasn't seen the light of day yet, just wondering if there are any other IAX phonesets around available NOW. Don't get me wrong, the Virbiage unit seem to be in the right price range, its just not available yet... Gary .

Hello, all. I have some problem when testing my IP phone with Windows Messenger. My IP phone supports such codecs as g729, g711. And Windows Messenger supports red, g711, g723 as you know. The problem comes up when testing with this sip.conf file. ([general] context displayed only) =================================================================================== [general] port=5060