similar to: Diva Server V-BRI-2 and internal numbers

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

Hi, the first thing I did with Asterisk is listening to `demo-congrats' by Xlite on the same machine. This works perfectly. The config files are those shipped with the package. Now I want to listen to it over ISDN/Capi but I don't succeed. My `capi.conf' is like show in many tutorial on the web. In `extensions.conf' I just added the following lines: [capi-in] exten =>

Hi, I like to forward an incoming call on an ISDN line to my mobile phone. Since ISDN offers two channels, I thought that this should work, but Asterisk tells me, that there is no channel available. There is no one else using this line, so guess I made a mistake in the configuration or it might not work for another reason. Here's the CLI output , the capi.conf and extensions.conf. 83086921 is

Hi! I am playing around with Asterisk and have a problem :-) (Asterisk-version: 1.2.4, chan_capi-cm-version: 0.6.4) I have a sip-phone at my desk and an ISDN-phone (independent of the Asterisk-server) in my living room, when I'm not at my desk, the sip-phone is switched off. I would like to be able to accept calls at both phones (when available) and have Voicemail kick in if I don't

[Something seems to have went wrong with my previous posting. It appears on the archive page in another thread. I did not receive anything myself. So I may give it another try:] Hi, the first thing I did with Asterisk is listening to `demo-congrats' by Xlite on the same machine. This works perfectly. The config files are those shipped with the package. Now I want to listen to it over

I have a problem receving calls via the ISDN line, using the followin components Asterisk 1.0.9 with asterisk@home chan_capi-cm-0.6 AVM Fritz card datalink protocol = point to multimode I can make calls out with no problems so the issue is only incoming calls. When I make the call from an external line to the ISDN line connected to asterisk, I get a busy signal after about 5 seconds. I have

Hello, I have a noticed strange behavior in Asterisk 1.6.18.2 with ReceiveFax Digium FAX Driver: 1.6.2.0_1.3.0 (optimized for i686_32). I use a context [capi-in] for icoming ISDN calls: ====== [capi-in] ; Faxe fuer Ruben exten => 12345,1,Macro(faxin,ruben.roegels at jumping-frog.org,${EXTEN}) ====== My macro for the fax receiving looks like that: ====== [macro-faxin] ; Faxe ; ARG1 =

Hello everyone, I'm using very simple setup to make and receive external ISDN calls through a softphone (x-lite version 3.0 - Win32) via an asterisk box. Hardware setup: - Dialogic Diva BRI (lspci yields: Network controller: Eicon Technology Corporation DIVA Server BRI-2M/-2F (rev 01)) - ISDN BRI line Software setup: - Redhat 9 - asterisk-1.4.16.2 - chan_capi-1.0.2 Asterisk configuration

Hi, I have done my best and tired of searching the net about the problem. If anybody could help would be a great favour. Description of Problem ------------------------ I am trying to install two Netpci cards(Traverse Technology Netjet ISDN-s) on Trixbox 2 and aim is to use in Asterisk as dailin and dialout. I compliled the driver as directed in the manufacture manual. After installation dmesg

Hi, after I fixed my problem with the playback() application, I now have the next strange one. When I dial the number of our client, located in another town, I get a connection to the asterisk server, I can talk to my client or listen to his mailbox. If some in the town of this client calls him, he gets the ISDN error "service not available". Out office is connected to he office

Hi all, I have a Diva Server V-BRI-2 card, which support, as written in reference guide: Extended tone processing (human talker detection, generation and detection of country-specific tones) I would like to detect human speech and fax tone with asterisk. I think that the diva card transmit a DTMF code when detecting voice, but chan_capi doesn't receive this DTMF code. I verbose it

I seem to be having a problem with B3 on my ISDN line, as you can see from the dial string I am having to have asterisk generate ringing else there is no progress indication. -- Executing Dial("SIP/0014A8ACCB83-fd9f", "CAPI/g1/142392203000/ b|40|r") in new stack -- Called g1/142392203000/b -- CAPI/ISDN1/142392203000-0 is proceeding passing it to SIP/

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

I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm 0.6.4 When making outgoing calls I don't seem to have any control over the CLI that is presented to the called party -- it can be any one of the MSNs allocated to the line, allocated on what seems to be a random basis. This is on a BT Business Highway line (which is essentially an ISDN2e line with two built-in

Hi everyone, I have Asterisk SVN-trunk-r7498 running for a few months and I'm quite happy with it. However, I am experiencing a quality issue with my AVM Fritz!card PCI which is used with chan_capi. When somebody calls me on this line he hears a lot of noise and I hear "scratches" and "plops". It is very annoying. Below is is my /etc/asterisk/capi.conf I've tried to

I have a Diva Server BRI installed in a Debian system running Asterisk 1.2.22. When a call comes in the Background application plays a couple of times. Halfway through the second time, the call is disconnected (total time connected about 25 seconds). I've posted various configs below. Any advice on how to debug would be appreciated. We are located in New Zealand. CLI showing capi debug (just

Hi everybody, I have a problem which I cannot eliminate on my own. Has anybody any idea for the following: I am using the asterisk-version from Debian-Testing (1.2.13) with the latest chan_capi (also tried an older version). When using the Capi-Channel, everything works fine except from the sound it sounds extremely choppy and is unusable :-( When e.g. capisuite is used for fax, everything

Hello everyone, I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz, 256MB) with a TDM40b a TDM04b and an avm fritz!card pci. I experience a problem with voicemail: my messages are good unless the incoming call comes from isdn, which means via the avm fritz!card. In this case (and in this case only) the message is disjointed and I can hear at most 1 second out of a 1 minute

Hi! I'm trying to enable echo cancelation with my Eicon Diva Server 4 Bri card. I've enabled it from zapata.conf, (as I read from www.voip-info.org) ------------------------------- ; Enable echo cancellation echocancel=64 echocancelwhenbridged=yes echotraining=2000 ---------------------------- but nothing seems to change, echo is still present. Any idea? Should I enable/change something

Hi Guys! I got a Diva passive ISDN card and I can't get it work with asterisk 1.4, It is supported in the kernel as an isdn4linux device but I can't find "Modem" channel type when i type in: core show channeltypes. I'm guessing it is removed in asterisk 1.4. Tried with capi interface but it does not work :( Anybody got some idea how can i make it work or got a link to a