similar to: IPSwitchBoard new Release

I have been working on solving a major issue with IPSwitchBoard. It was reported that IPS would use all available memory and get the PC to grind to a halt. I could not understand this as I had it running on many different PC's in Denmark. I now found the bug: IPS would crash on any PC that had "." configured as decimal point (in Denmark we use ",") this meant

I am proud to announce that I have just released a brand new IPSwitchBoard totally rewritten from scratch. The new version has been built on the experiences gained by the previous versions. Download for FREE: http://ipsoftware.thorben.dk IPSwitchBoard is totally customizable and will give you, among other things: Unattended/attended transfers. Park calls and retrieve/forward them again.

Hi all, Just to let you know that I have just released a new version, 0.65, of IPSwitchBoard-BETA. This is some of the fixes: . Load IAX2 extensions as well as make calls/transfers To IAX2 (I will add ZAP next) . You can monitor Parked Calls (I have a lot of problems getting parking to work) . Possible to create a trace file by checking Trace on the config page . lot's of bugs fixed

IPSwitchBoard Version 0.67 Release notes: CRM integration, can call a web page with callerid when there's an incoming call. You can specify the min. and max. length of the callerid. Drop any active call. Help file integrated in IPSwitchBoard. Play button for sound files. Bug fixes - thank you for all your feedback. Download IPSwitchBoard for FREE here:

Release 0.66 of IPSwitchBoard is now available for FREE download at: http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA Enhancements: Support for Call Parking and retrieve/forward them again. Last Call on the Queues Page now displays a date-time in human readable format. Added CallerID on the Queue Members listing on the Queue page. New page with Agent information. Minor bug

IPSwitchBoard Version 0.123 - 17 July 2005 * Queue Analysis has been added to IPSwitchBoard. In order to properly manage ACD queues, it is important to be able to keep track of details of call setups and teardowns in much greater detail than traditional call detail records provide. In order to support this, extensive and detailed tracing of every queued call is stored in the queue log, located

IPSwitchBoard Version 0.123 - 17 July 2005 * Queue Analysis has been added to IPSwitchBoard. In order to properly manage ACD queues, it is important to be able to keep track of details of call setups and teardowns in much greater detail than traditional call detail records provide. In order to support this, extensive and detailed tracing of every queued call is stored in the queue log,

Hi all, I have just published my last few weeks of hard work: IPSwitchBoard BETA. Please let me know what you think and post comments on the Wiki. http://www.voip-info.org/wiki-IPSwitchBoard+BETA Thank you

I am rewriting IPSwitchBoard at the moment. I want to make IPSwitchBoard "Skinable" meaning that you can design your own skins with company logo etc. You will also be able to add graphical extension buttons, and led's that will light up ex. DND, busy/free, message waiting and much more. If you are graphically minded and would like to help with making test skins using PhotoShop or

IPSpeedDial creates speed dial numbers for Asterisk. Download from: <http://ipsoftware.thorben.dk/> http://ipsoftware.thorben.dk Use this to create speed dial numbers that can be used by all extensions on your Asterisk server. This program will create entries in the asterisk database which you then can lookup in you dial plan get the number to dial. All extensions can now dial

Hi, I'm using IPSwitchboard v 1.8.10, a sort of Operator Panel, to monitor my Asterisk's extensions. Recently I noticed that on the official site (http://ipswitchboard.thorben.dk/), where I downloaded the software some weeks ago, this project is no longer supported. Is there anyone that can say me where I can find the Italian version of IPswitchboard or if there is a way to translate the

Release 0.68 of IPSwitchBoard is ready for download at: http://www.voip-info.org/tiki-index.php?page=IPSwitchBoard+BETA Release Notes: Import/Export extensions to "cidname" in Asterisk Server, integrate this in your dial plan to see who's calling by using the Asterisk app. "LookUpCidName()." Set "Do Not Disturb" for an Extension, and type a reason for DND

IPSwitchBoard Version 0.114 - 6. may 2005. * IPS will now delete recordings on the Asterisk server after it's transferred to your PC. * You can now configure call Rates as?X of a Currency Unit. If your Currency is?Dollars then?the smallest?Call Rate?is?Dollar/X. X would typically be 1000. * IPS will now save the last?position and size and set that the next time your start IPS * bug fixes

On the Mambo CMS site there are vulnerabilities found. Whilst this is not a CentOS problem, people rent/deploy servers (CentOS) on the net with Mambo. A guy in one of the user forums on the net, had his Mambo 4.5.2 server hacked and they installed some interesting stuff in /tmp . When a server is hacked it gives bad PR for the underlying OS. <----announcement on

Hello everyone, So, this isn't exactly what it seems. I am not looking to integrate Asterisk and Mambo. I am the maintainer/creator of AstLinux, and I have recently decided that I should really have a better web site for it. I would like to use Mambo so that I can do updates easily, from anywhere, without having to waste time learning PHP/HTML/etc. Mambo CMS seems the best and most

I'm trying to get a script working on a website to send out automatic email reminders to customers reminding them monthly to change furnace filters. I haven't got one running successfully, yet. That made me think - could it be done with a phone call using Asterisk? A monthly automated phone call to remind people to change their furnace filter? I have no ability to figure this out

Hi All when I try to use auto-dial to connect to outside phone , my applications get executed before the caller attend the calls , this happens only when I call outside no , ie when I use Channel: ZAP/1/0507451111 in my sample.call file , if I use Channel:SIP/326 , it works fine my ?sample.call? file contains Channel: ZAP/1/0507451111 Callerid: Asterisk MaxRetries: 2 RetryTime: 10

This changes the message_age_timer calculation to use the BPDU's max age as opposed to the local bridge's max age. This is in accordance with section 8.6.2.3.2 Step 2 of the 802.1D-1998 sprecification. With the current implementation, when running with very large bridge diameters, convergance will not always occur even if a root bridge is configured to have a longer max age. Tested

How do I get a flash plugin to work with firefox? I thought that I installed it correctly, but I have yet to see any flash videos through firefox. I keep being told that I need an additional plugin. When I folow directions, I'm told I already have it? At least once, I was told I needed to upgrade. In case it helps, here is the output of yum provides '*flash*' | grep -e lash -e dobe

I'm in the process of writing a simple autodialer to dial a list of numbers and play a message. One of the options I want to give them is a way to "dial X to have a customer service representative call you" Looking for a simple way to pass the number that I dialed to a script in extensions.conf... something like this: [serviceinterruption] exten =>