similar to: (Another) Queue log analyser

hi I really want a good queue_log analyser, but I don't want to waste > EUR 1000 for something like that, so I thought starting a small project for it. I started off in php just creating a basic parser for the log, and I'll go on extending it. See http://karlsbakk.net/asterisk/scripts/queue_log_analyser-0.0.1- pre1.php.txt Does anyone want to join this effort? roy

I was wondering if anyone knew how to do the following.... Call comes in, gets put into a Queue, say `Sales`. Then the queue member is presented with the option to exit the queue, yet have the phone system sit in their place for them. When the virtual member reaches the front, call back the caller and connect them to the agent. Any ideas? Did i explain that ok? :) Cheers, Ben Merrills

Hi, I m trying to use a custom analyser to add my french stop words... i m reading the tutorial at : http://projects.jkraemer.net/acts_as_ferret/wiki/AdvancedUsage My problem is that i ve no idea where to put my custom Analyser class like : class GermanStemmingAnalyzer < Ferret::Analysis::Analyzer include Ferret::Analysis def initialize(stop_words = FULL_GERMAN_STOP_WORDS)

> On Jul 28, 2016, at 10:53 AM, Justin Lebar <jlebar at google.com> wrote: > > Thanks again for your thoughts, Chris. > >> As a straw man I would suggest the following criteria for inclusion into the mono-repo: >> >> (1) Projects in the mono-repo must be tightly coupled to specific versions or commits of other projects in the mono-repo > > I'm fine

Has anyone tried integrating Asterisk and Rhetorical's rVoice software? We're evaluating different approaches to system announcements via T2S. Has anyone gone down this route that could give some advice? I've installed festival and wasn't too impressed, the demo one the website seems far better quality and clarity then the defaults in the source package. However I must admit

Is there a UK Asterisk users group? Would be interested in contacting others in the UK who use asterisk for either home or business applications. If there is, could someone provide me with some contact details, else anyone who's also interested, contact me off list. Cheers, Ben Merrills -------------- next part -------------- An HTML attachment was scrubbed... URL:

To those who are/were interested :-) Ok, I've setup a mailman mailing list. I don't so this often, so I've done a bad job of it, please let me know. Oh, and I know the SSL cert has expired... was self signed :-) Now I've apologised for my crapness, the URL to signup to the mailing list is: https://xdev.net/cgi-bin/mailman/listinfo/asterisk-uk Hope this is ok with

> On Jul 28, 2016, at 12:05 PM, Justin Lebar <jlebar at google.com> wrote: > >>> The decision of whether or not to include these projects >>> affects only read-write consumers of these projects -- of which there >>> are relatively few people. >> >> Maybe there are few, but the impact is non-insignificant. Also I think the opinions of the

Hi All, I'm stuck and need some help. I have installed the Asterisk CDR Analyser Version 2.0.1. It mostly works except for the CDR Report. I get the following error even though it lists the CDR details. Database error: Invalid SQL: SELECT substring(calldate,1,10) AS day, sum(duration) AS calltime, count(*) as nbcall FROM cdr WHERE UNIX_TIMESTAMP(calldate) >=

Hello, We are looking for commercial solution SS7 with Asterisk. It does not need to be "build-in" with Asterisk. Could anybody suggest something? Thank you in advance. Bart

Is there a way of getting the dialled number from an AGI? Is it passed in the initial variables, or can it be pulled out or passed across from the dial plan? Cheers, Ben Merrills Griffin Internet -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040630/71e0bcda/attachment.htm

Hi, With multiple proposals to improve the situation for newcomers in flight, let me add another brief proposal: Let’s add a CONTRIBUTING.md file to the top-level directory of llvm-project. I think this would make the information about how to contribute more visible and Github uses CONTRIBUTING.md to display ‘Contributing Guidelines’ at a few places, like when creating new issues or pull

Hi, How can I (through the manager interface) steal a call from one phone, and transfer it to another? Does asterisk provide for actions like this? It's a common action in Lucent systems it seems. Cheers, Ben -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040816/7a79aa0f/attachment.htm

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin

Hi Keno, Could you give a short high-level overview of the way Julia works now with MCJIT instead the JIT: What I gather so far... * Compiled IR functions are emitted to a shadow module. * Any used function is cloned into its own new module and the module is added to MCJIT. * Called functions or globalvars are only declared in that module. * Modules are never removed meaning "old"

Hi, I would like to know how suppress number for outside dialling in CDR table. For example, if I need press 9 key to make an outside call, I would like that the number in dst field in cdr table was the outside number without 9 key. It's possible? Thanks in advance, srsergio

Hi Duncan, > […] > 2. Those working on projects *outside* the monolithic repo will get the downsides of both: a monolithic repo that they are only using parts of, and multiple repos that are somehow version-locked. > > 3. For many (most?) developers, changing to a monolithic git repo is a *bigger* workflow change than switching to separate git repos. Many people (and at least some

hello all, i have two asterisk server for call transfer and one more asterisk server for agent login(server_X) where agent take the call. server_A and server_B server_A is connected with pri and configure with 60 channel for call transfer into server_X server_B is connected with pri and configure with 30 channel for call transfer into server_X my query is that some time two call originate same

Hi Everyone, I am using R 2.10.1. lmer function works properly, however pvals.fnc () does not despite the fact that I uploaded: - library(lme4) - library(coda) - library(languageR) This is the error message I get pvals.fnc(lexdec3.lmerE2, nsim=10000)$fixed Error in pvals.fnc(lexdec3.lmerE2, nsim = 10000) : MCMC sampling is not yet implemented in lme4_0.999375 for models with random

The attached patches allow the pciback module to perform a reset whenever a PCI device is: - attached to the pciback module, as so: echo "0000:01.07.0" > /sys/bus/pci/devices/pciback/bind - detached from the pciback module, as so: echo "0000:01.07.0" > /sys/bus/pci/devices/pciback/unbind - and when the guest is done with (internally when the guest is not using