search for: switchboards

Switchboard is a simple, event-observing framework for ActiveRecord. It''s designed to make it easy to add observers for all models in your app, and to easily turn them on and off selectively. Intallation gem sources -a http://gems.github.com sudo gem install zilkey-switchboard Usage First, require switchboard above your rails initializer: # environment.rb require

I've been trying to setup AddQueueMember() as a replacement for the deprecated AgentCallbackLogin(), but I get _tree_ Queue()'s. Massaged extensions.conf (can provide the original if need be): ----- s n i p ----- [default] include => agent-loginout include => local ; ---------- [agent-loginout] exten => _100.,n,Macro(queue-addremove,I${EXTEN:3},dispatch,10)

Hello All I am looking for a PC based switchboard application. Cisco CallManager has a web attendant console that allows you to use the PC to transfer calls and the like and I was wondering if there was a similar program compatible with *. Thank you in advance Keith D'Atrio -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi there, I'm looking into developing an in-house switchboard application. Does anyone here know of a way to control a hard-phone from such an application. For example, the attendant forwards a call with another one in queue. Once the first call has been forwarded (by keyboard shortcuts or dragging-n-dropping) - she presses a button (on the computer) to answer the waiting call. Now, if the

Good day all I'm trying this switchboard demo (ftp://ftp.lanwan.fi/switchboard) But in the pdf howto they speak about a swb.txt and I dont have that file Thanks Altus

Hi, I am using wine-20050111 with Access 2000. When I start the Access application the switchboard (opening screen with buttons) is minimized. I can drag the window bigger and everything works fine. But when I maximize the switchboard with the window button, Access "hangs". In the console these messages are repeated every time: fixme:hook:NotifyWinEvent (32773,0x20078,159,0)-stub!

Hello, I am still researching my dive into Asterisk at my workplace, and I was wondering about how switchboard activities are handled.. Right now, a call comes into our switchboard, and the operator forwards them to the appropriate line, thus freeing up the primary number and allowing more calls in. Everyone on campus has a direct-dial line as it is right now. I want to eliminate most of

Hello, we are looking for someone to write a simple switchboard application for asterisk which can park, place on hold, and transfer to extensions. Standard switchboard/operator console features but we also need the following ability. We need to have a screen/panel which monitor the operators/main extension and displays a DID's owner. For instance, let's say we forward all of our

Hi, What's your favorite switchboard application software, for a 100-200 seats company attendant ? The setup is : PSTN ----------- Asterisk ------------- SIP Phones | --------------------- Attendant console Has anyone used Icecom switchboard, FOP, Netwise, Hud ? What do you think of those ? Regards -------------- next part

I'm new to this list. Reading the asterisk handbook pdf (good work) but but still have some questions. Using Trustix 2.1 and installed Asterisk via CVS, zaptel and libpri. We have a dedicated server which is connected to our telephone company. It makes us able to call ordinary phones via VOIP using Ericsson DRG22. Would like to make people able to call me - and get a message "dial 1

Hi Does anyone know a good, scalable switchboard solution for asterisk? I've been looking around and I've found a couple but I'm not sure yet... Have anyone here used one in large environments? We need usable GUI with the usual stuff like queues, transfer, meetme etc roy

Hi, I'm a "noob" to Asterisk and this list so I apologize if I'm out of place with my questions. I'm planning to utilise Asterisk to build a "switchboard" (of sorts) with it and was wondering if this would be feasible (from what I've read about Asterisk it seems possible). What I would like to achieve: All analog calls routed through Asterisk. Callers will

Hi all , I'm from Germany and need your help. I use a telephone switchboard from the German Telekom. The Software is only for Windows but I like to use it with Ubuntu 10.04 and wine. With Windows you have to install the Software this way. 1. install the capi-driver ( is for the switchboard) 2. install the Software( to handle the switchboard) 3. unplug the usb-cable and restart the Computer

Perhaps I've got the whole concept wrong, but here goes: Using 1.4, when someone from the outside dials my direct line (123456), I want it to call my extension at work (SIP/456), my extension in my home office (vpn connection to corporate lan, SIP/678) and my mobile (654321). So my dialplan is thus: exten => 123456,1,Dial(SIP/456&SIP/678&Zap/G3c/07803654321,30) exten =>

This one is for the archives. I got a call today that the * at one of my clients was not working. The switchboard is set up to ring for a while and then the rest of the phones start up if the switchboard doesn't pick up. This was not happening. Instead the mobile phone of one of the people there was ringing and after the delay the internals started ringing. When I connected to the web

I can't check the voicemail for the switchboard. Asterisk hangs up for some unknown reason... ----- s n i p ----- -- Executing [*500 at default:1] Wait("SIP/597-00f0c410", "1") in new stack -- Executing [*500 at default:2] VMAuthenticate("SIP/597-00f0c410", "500 at default|s") in new stack -- <SIP/597-00f0c410> Playing

Well I'm sure this issue has been bean up a few time since it's one of the only ones I can't find a real "simple" answer to. I'm trying to find away to do attended transfers through the manager interface, for a pc switchboard / Agent client solution, but so far coming up short. The action Originate is part of the solution, but what really I want is the phone being taken

Hi Guys I am trying to write a macro for a call return so for example Anyone in the company transfers a call to another extension and it is not answered etc it must return to the person who did the transfer I have got it working but if the call originates externally for example someone calls in to the switchboard and they transfer it then it tries to return to the outside caller. As

Dear All, I've got a working asterisk installation which I need minor help from. Currently, I'm running a Sales Queue, which is answered by a selected group of people. Here are my queues.conf [sales-hotline] strategy = roundrobin timeout = 10 member = SIP/602 member = SIP/603 member = SIP/701 member = SIP/604 After calls come in, it works fine, however, I notice that even when SIP/602

Hi there, could anyone drop a short line on what "cidinternalcontexts" exactly does in voicemail.conf? The Wiki explanation isn't sufficient - at least not for me... :-> Also: How/where do I define an Operator extension? Cheers, Philipp