similar to: combining calls from 2 queues

Hello all, I'm struck with a very strange problem today. I've an AGI with some code subroutine snippet as follows: sub enable_sbc($) { my $carrier = shift; my $tmp = substr($carrier,1); my $jkh = $tmp; $server_port = $ast_agi->get_variable("SIPPEER($jkh,port)"); $ser_ip = $ast_agi->get_variable("SIPPEER($tmp,ip)");

I'm having problems enabling stutter dialtone for users connected to channel banks. Half of our users are on iaxy's and the other half are connecting to channel banks. The users on ixay's are getting the stutter dialtone on new voicemails, but the ones on the channel banks are not. Currently, all users are in the default context in the voicemail.conf file. I've tried the

Hello, Raw stats: Version:1.8.3.2 OS:Centos 5.6 Special setup: postgre database I am having a few queue issues with Asterisk specifically relating to breaking out from queues while on hold. The intent is that while someone is on hold they can press a key (lets say *) to break from the queue and go elsewhere (in this case to leave a message). However In all of my testing I am unable to get

I am unable to ping NE.TW.RKB.IP1 from an outside network. Other machines which do not have access or routes for NET.WOR.KA.0 respond just fine. How do I get it to respond on both NET.WOR.KA.0 and NE.TW.RKB.0 given all default traffic should go through NET.WOR.KA.1 unless it is in reply to traffic from NE.TW.RKB.1 or there is an outage. [root at host20 ~]# route -n Kernel IP routing table

Hello, I have the following structure: - app - controllers - helpdesk - call_controller.rb (Helpdesk::CallController < ApplicationController) - models - employee.rb - department.rb (Department < ActiveRecord::Base) - helpdesk - call.rb - call_history.rb (Helpdesk::CallHistory < ActiveRecord::Base) Is it possible to have a relationship between

Hello, I've been planning to replace my aging CENTREX switch with a new PBX and am seriously considering Asterisk as my solution. I work at a college, and we currently support just under 300 regular analog lines to the offices and whatnot. I was wondering.. Is asterisk ready for such a job? Would I be making a mistake to deploy this across the campus at this time? It seems that

Forwarded message: > From R.v.Gogh@kappe-int.com Wed Mar 3 07:54:28 2004 > Message-ID: <0FDD52D38220D611B7CC0004763B37441B2572@HNTS-04> > From: "Gogh, Ruben van" <R.v.Gogh@kappe-int.com> > To: Darren Reed <avalon@caligula.anu.edu.au> > Subject: RE: IPFilter and FreeBSD (was Re: mbuf vulnerability) > Date: Tue, 2 Mar 2004 21:54:23 +0100 >

Hi, I have a dataset from ILO, originally in csv-format, that I have read into R. It is cross-sectional time-series data, so I have a bunch of variables and dummy variables that I need to extract data from for the entire time period. However, the years are separated by columns instead of rows, as is usually the case in R. This is what it looks like: > str(laborstafinMFBA)

Hello, My company is looking for someone familiar with Rails, but *really* familiar with front-end web development; CSS, Javascript (Ajax + +good), and one of the XXp templating languages. We''re migrating a rusty J2EE-ish app to Rails, and we need someone who can flip between controller work and view development with ease. Very relaxed, flexible work atmosphere--we value

Hi, I am trying to do a multiple regression on the dataset "Hdma", available in the Ecdat package. The data looks like this: > str(Hdma) 'data.frame': 2381 obs. of 13 variables: $ dir : num 0.221 0.265 0.372 0.32 0.36 ... $ hir : num 0.221 0.265 0.248 0.25 0.35 ... $ lvr : num 0.8 0.922 0.92 0.86 0.6 ... $ ccs : num 5 2 1 1 1 1 1 2 2 2

Hi, We are currently testing Asterisk with Wildcard 100XP and serveral Cisco ATA Box. Everything works great except that the card does not detect the hangup signal. We are using a standard Belgian PSTN line. I have not found anything about a be zone (only us, fr, de, nl, ...). Does someone experience the same problem? Do I need to create a new zone be (and how to do that)? Another small

I will be out of the office starting 22-03-2007 and will not return until 16-04-2007. I will be on holidays until the 16th of April. I will have no access to e-mail during this time. If this is a matter relating to IT support, please contact the IT Helpdesk by either opening a ticket via http://helpdesk.qad.com/request. Or calling the EMEA helpdesk on +31 20 654 7139. If this is an urgent

Switched from Asterisk 1.8 to 13.3.2. Now it logs to /var/log/asterisk/full (good) as well as /var/log/messages (not good). Anyone know why? # grep -v "^;" logger.conf [general] [logfiles] console => notice,warning,error messages => error full => notice,warning,error,debug,verbose,dtmf,fax Thankfully, the .../full logs are rotating properly now (thanks Dale) but we don't

Ok, commented out that line. It's still doing it. Reloaded dialplan. Please don't tell me I have to restart asterisk. Thomas M. Peters | Systems Administrator | tpeters at mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org >>> Tiago Geada <tiago.geada at gmail.com> 6/26/2015 12:07 PM >>> messages => error states to log error messages to

Duncan: You may have it right-I took one phone and set the ring time to 60 seconds. I now get about 4 rings on that one. I wonder how I can change the timing source. Thomas M. Peters | Sr. Systems Administrator | tpeters at mcts.org<mailto:tpeters at mcts.org> Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org<mailto:helpdesk at mcts.org> Milwaukee County Transit System

hi friends, I have configured a HelpDesk Ticketing System on Centos4.4. The problem I am facing is that there is a file called "site.xml" which contains the information about database connections and I don't want ppl to be able to read that file through browser. As per the readme.htm of that software if the below entries will be put in .htaccess then nobody can read the xml through

We have an old Asterisk 1.8.7.0 system desperately need to keep alive for another 6 months or so. We had all kinds of hardware problems, so we virtualized it. Now, random extensions ring once and go straight to voicemail. I went to one of the affected extensions and changed the ring time from the default (20) to 26. Still one ring. I changed it to 30. Now I get two rings. Other extensions ring

Here is what I am attempting to do (which works well on Cisco Call Manger). I have some 7960's that have multiple lines on them. The second line specifically is a "helpdesk" line that is shared among multiple phones. Here is how I am making that line ring on multiple phones, maybe you have other suggestions: exten => 135,1,Dial(SIP/135@100&SIP/135@101,20,rt) So this

Hi, folks, I''m attempting to run an e-mail server on Xen. The e-mail system is Novell GroupWise, and it serves about 250 users. The disk volume for the e-mail is on my SAN, and I''ve attached the FC LUN to my Xen host, then used the "phy:/dev..." method to forward the disk through to the domU. I''m running into an issue with high I/O wait on the box (~250%)

Which function implements the piecewise cubic Hermite interpolation? I am looking for equivalent of matlab's interp1 with the method = 'pchip' Here is the reference http://www.mathworks.com/access/helpdesk/help/techdoc/index.html?/access/helpdesk/help/techdoc/ref/interp1.html& -- View this message in context: