similar to: Passing DID to external number?

Hi all, I'm new to the forum. Oh no....newbie question coming, I hear you all cry! I'm playing around with Asterisk@home and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've

For the DID's the easiest way for you to trasmit the incoming DID is to create custom extensions for the external numbers that access the external trunk directly. (e.g. they should NOT go to Loca/xxxx. or it will not retain the orignal CID in freepbx- which is effectively how it is being sent when you put xxxxx# in the ring list). As far as why it is only ringing one of your external numbers,

Hi all, We use AAH to run our office telecoms registered with two Sipgate accounts. Today, Sipgate appeared to have had problems with their server with oneway audio on every call. In order to cause the Sipgate message service to pick up in stead of our AAH box, I simply unplugged the network cable. We now have problems where AAH does not seem to access the network. I plugged the network cable

I''m brand new to rails and am trying to build a simple ajax email collection form. It''s just a coming soon sort of thing where the user can enter his email address in the field, hit submit, and the ajax form dumps the email into the database and says "success" unless the email is blank or not unique in which case it says "failure." The trouble is, no

an application i am working on sends emails to an array of users. it is possible for the user''s email to actually be in the array more than once. i don''t have a real strong grasp of the ruby syntax yet, but is there an easy way to go through and either remove all of the duplicates or at least only send an email to that user once? -- Posted via http://www.ruby-forum.com/.

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to

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 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:

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

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

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

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

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

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

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 =>

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

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

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

Dears; I am facing the following problem: Already we requested from the service provider to enable the auto jumping service for our analoge telephone lines, so because we have 4 telephone lines from the service provider, then if you called line # 1 and it was busy, then the call will be sent to any available line #2 or #3 or #4, and if you call line # 3 and it was busy then the call will be sent

Hi All, As is obvious by my joining the list, I'm interested in learning more about Asterisk. I have downloaded the PDF manual (for version 1.4) and am beginning to go through it. What I'm looking for in the short- term, however, is a more concise reference for common Asterisk configurations and setups. I currently have a non-profit client to which I am donating work. They