similar to: Asterisk queues : command to run when a call is being bridged

GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can

Hi there, I am configuring a small voicemail server and I am facing the following problem. Executing this command: exten => 1234,1,VoiceMail(${NUMBER}@test) When a user does not have a customized temporary greeting vm-intro message is played asking for the message to the user but when the user has already a temporary greeting both the temporary greeting and vm-intro are played. Basically

I'm not sure I'm barking up the right tree here, but would I need to make use of groups to plot two separate datasets within ONE panel in xyplot? The desired end result is a single xy plot of two separate (but similar in values and ranges). Full code follows, xyplot code at bottom #########Determine Frequencies ##########coastal_slope #needs the maptools package to read ESRI grid

Hi, I've been reading all I can on Google (and Asterisk TFOT book) looking for ideas on how to implement an automated blacklist feature. I would like to automatically blacklist a incoming number based on timestamp and count information. For example, if I get a prank call from the same number 5 times within 15 minutes, I want my dialplan to automatically blacklist this number. Should I be

Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap] disconnect => ** My Dial command looks like this:

Hi, I am trying to use the attended transfer. So I put this in my feature.conf: [general] [featuremap] atxfer => *0 blindxfer => #0 I completly restart asterik, and not just make a RELOAD. But during a call, when I press # it runs a blind transfer and if I press * I am disconnected. I am using the CVS version of * get as explain here

Using version 1.2.35 built by root @ slate on a i686 running Linux on 2009-09-15 00:24:10 UTC Problem - I cannot get featuremap right. Have added a feature that I want to direct to an extension in extension.conf Extension is 521 In features.conf - [applicationmap] dumpcaller => #9,callee,goto(521|1) show features - Dynamic Feature Default Current ---------------

Is features.conf included in the cvs as of 8-1-04? I have updated, but am not seeing it?

Hi, I'm using the new AMP which provides a call recording. The options of recording call Always and Never are well working. But how to use the On-Demand option ? Should I press a pad ? Is this configured in the featuremap of features.conf ? Why my modifications in that features.conf have no effects ? Please advice me. Alexis.

Also, if password-based auth is not allowed, WTF would you want to log passwords? This whole idea is ugly, and smacks of a teenage-level prank attempt. I would strongly object against any such modification of the main source (though I'm sure the maintainers are sane enough to never let such a crap in). Of course the original poster is free to hack his own copy in whatever way he wants.?

Dear All, zlib has been removed from R-devel src/extra recently, and building R requires zlib >= 1.2.5. Ubuntu 12.04 LTS (also used on Travis CI) only has 1.2.3. This means that the next version of R will probably not available on Ubuntu 12.04. (Unless I am missing something of course.) Which is probably fine, it is almost three years old now. I guess R-core is aware of this. Just wanted to

Hi, Does anyone knows of a place where one can download through HTTP (and only HTTP) the CentOS 4.1 DVD ISO ? I'm behind a firewall and have no chance of either getting it through bittorrent or FTP. Any help would be apreciated. Warm regads, M?rio Gamito

All, Well I got a new laptop with vista. Got it on monday and I have seen 4 blue screens on it. I am getting ready to dual boot it and centos 5 (any day!!!). I just smile..... Please no flames on why vista. I had to for checking other things out... Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I was wondering wether there's a function in R that takes two vectors (of same length) as input and computes mean values for bins (intervals) or even a sliding window over these vectros. I've several x/y data set (input/response) that I'd like plot together. Say the x-data for one data set goes from -5 to 14 with 12,000 values, then I'd like to bin the x-vector in steps of

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon =>

Hi folks, I was using the following featuremap: blindxfer => *1 disconnect => *9 atxfer => *2 parkcall => *7 automixmon => *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I

I have set up an operator queue for our receptionist. That way, if she takes a break or is out, by logging out of the queue, calls to the "Operator" can be handled by other agents. I have set strategy = roundrobin in queues.conf. According to "the book" ATFoT, roundrobin always starts with the first agent in the queue. This is the desired result. I want all calls to start

Hi List. I have a small problem in using the transfer key transfer of IP Phone in Asterisk 1.6, I think I spend some detail in the configuration but can not find. What happens is, when I do a transfer using the Transfer button, the phone, does not play the music on hold, which is waiting on the phone is silent, and I have the same settings on a 1.4 server, and the music plays correctly when

Would somebody perhaps have some background on these tar files with such interesting names? To my understanding, OpenBSD=4.1 == OpenSSHD-4.6. Of course, this could just be a prank. Google knows about the mail address this (presumably) came from since about mid-2007, so it apparently has not been created just for the occasion.. Thanks in advance Jan -------------- next part -------------- An

Hello all, First of all, let me apologize about the length of this message, but I suppose it was necessary to include the details. I've spent quite some time already trying to get the call transfer function to work on my Asterisk installation. Let me first describe the general situation of the setup I am using, so you might be able to pinpoint the cause of the problem. I'm currently