similar to: Hardware manager

Hi, On Friday we changed our Telco-Provider (from German Telekom to Mnet) and recieved new Numbers. I changed the extensions in extension conf to match the new numbers. But i always get: Jul 18 12:10:39 WARNING[245776]: pbx.c:1780 ast_pbx_run: Channel 'CAPI[contr1/89064934]/0' sent into invalid extension 's' in context 'default', but no invalid handler I only changed the

Dear asterisk users! I want to control a hardware pbx with asterisk. What I need to do this is being able to press "hold" which can be done with capicommand(hold) and then send digits on a bri card which connects to my asterisk computer. So far I use Dial(CAPI/ISDN1/27:<<digits>>/bo,15) to do this. Are there better ways? Note that these are not dtmf, I'm afraid. I use

Hello everyone, My setup: Debian squeeze Asterisk 1.8, DAHDI, libpri, compiled from source TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800 I'm trying to get MWI for Voicemail working. In the same server I have also got an Eicon DIVA PRI card for testing purposes (it is integrated via CAPI and the chan-capi channel driver into my Asterisk). MWI works just fine there. I

hi all, i have a probably very stupid question/problem. for testing purpose i am trying to get asterisk running with two isdn cards. I'd only like to here the demo sound when i call the number - but nothing works. The output of show channels is not showing any channel - should there be 4 channels ? - capi info shows my two cards perfectly. The ISDN Controller's are attached to an PTMP

Hi, I've been learning asterisk for a couple of weeks now - and it worked for me as faar as standard configurations where concerned (sip/iax outbound/isdn4linux & capi with AVM Fritz!, Digium X100P FXO). Now I recently I'evaluating to use asterisk as a replacemnt for our companies (15 employees) legacy pbx system and I'm experiencing multiple problems with the hfc isdn cards:

When i send more than one messages shortly after the other, my log (/var/spool/asterisk/sms ) looks like this and only two of four messages arrive. What am i doing wrong ? I am using an AVM B1 PCI with chan-capi and 1.4.4. and also, when sending with smsq -x only two of the messages are handled. (i thought, asterisk itself handles the queues ? ) Here the log: 2007-07-09T15:04:14 YOM04 0 -

Hi chan_capi users, this thing is awesome, no delays like in modem_i4l! Plus, it got those nice ISDN features. Here's my question: Does my service provider (Deutsche Telekom) have to provide me with these Services (CD, ECT)? (the Readme in 0.2.5 says "does not relay on service CD") I know, that I don't have CFU,CFNR,CFBS (which I would have to order seperately). How likely

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

Hello Armin, hello List I'm trying to get chan_capi working with asterisk from debian stable (asterisk 1.0.7, the debian version number is 1:1.0.7.dfsg.1-2). I managed to get it compiled by providing my own version of ast_copy_string. This is an Austrian PTP line. I can do outgoing calls fine (no comprehensive tests yet). For incoming calls, I'm getting "No answer" on the

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

Hello All * users. I have been looking for a way to allow GSM termination through Asterisk to occur. I am not a Telekom fundi and I have set up IAX2/IAX/SIP on asterisk with the ZAP channels via the Digium TDM 400P. I am unable to find any place that can tell me the cost of the VoiceBlue with a currency to I can calculate the cost of buying one. Alternativly - or just out of interist - I

Hi list! I'm very new in Asterisk and VoIP, and of course I have a problem... :) Well, my problem is, that Deutsche Telekom wants me to change my ISDN to VoIP... :( I must do that, since I have no alternative. Well, I have now two VoIP-phones (Thomson ST2022 and KE1020A). I can configure my two numbers by Deutsche Telekom and I got now an extra number from Messagenet.it. Now the

Am 14.06.2020 um 17:05 schrieb Antony Stone: Hi Antony, > You mean that the Thomson phone is registering to Deutsche Telekom? > > I thought it was registering to your Asterisk server. Sorry, I didn't read correctly your test 2b... Normally my Thomson phone is registering to my Asterisk server. I tried to register the Thomson phone directly to Telekom's server, to check if the

I'm testing * (CVS-09/16/03-02:07:49 with zaprtc 0.0.1) with Fritz!PCI (chan_capi 0.3.0), and have a couple of funny things - I wonder if anyone else has seen them: - Now and then, * just exits. Until now I had lowish-level verbosity on, so all I saw was 'Executing last minute cleanups'. What can trigger * exits? (in other words, what should I pay attention to when attempting to

Hi list! I already had the problem last year, then it would be solved (surely from some technician by Deutsche Telekom on their servers), and now I have the problem again (but I didn't changed my Asterisk configuration). The problem: after 15 minutes will the call dropped, but only if the call is to another nation! If I just call another phone in Germany, I can speak longer than 15

Hi list! I think there are many german users in this ML, that use Asterisk with the new line of Deutsche Telekom (Magenta Zuhause). My ISDN will be converted in Juli (Kaboom-day at Juli, the 3rd...) and right now I can just hope, that I configured my Asterisk well to work with Deutsche Telekom, but I cannot be sure, since I can't test it... So my question: can someone using Asterisk with

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 list! Yesterday Deutsche Telekom had a really big problem and Asterisk couldn't connect to the remote Server (by Telekom) until today about 7:30. Well, it could happen... What I find really annoying was that I needed to restart Asterisk as I checked with sipsak that the Telekom-Server works... I think, this should not be normal... Can someone explain me why it happens and what I have to

Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi again, > 2b. Take your Thomson telephone to some other location with Internet access, > let it register to your home Asterisk server, and them make a call to the same > number yet again. I'm sure you can get the Thomson to connect to Asterisk via > some external network, since you say you can do this from your Android phone.

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!