similar to: CAPI didn't get a frame | avoiding initial deadlock | multiple instances of Asterisk

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 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

Hi, I am having a problem with (Asterisk is crashing) with a Fritz card PCI / chan_capi. Receiving Calls from PSTN works, but outbound calls make asterisk crash (Speicherzugriffsfehler/Segmentation fault). The crash occurs upon dialing with the other phone not even ringing. I hereby ask if somebody reading this list can confirm or disprove my issue. Does anbody run a recent asterisk 2.6 with

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

1. If the other end picks up the phone the conversation is successful and when the conversation ends, the channel is properly released: -- CAPI/ISDN1#02/2104988888-0 answered SIP/1000-081d91b0 == ISDN1#02: CAPI Hangingup for PLCI=0x101 in state 2 == Spawn extension (internal, 2104988888, 1) exited non-zero on 'SIP/1000-081d91b0' > ISDN1#02: CAPI INFO 0x3490: Normal call

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'm currently using chan_capi-cm-0.6, with the following capi.conf: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 language=de [ISDN1] msn=8304490 incomingmsn=8304490 isdnmode=msn group=1 controller=1 softdtmf=1 context=demo echosquelch=1 echocancel=yes echotail=64 callgroup=1 devices=2 Each user has a different numer, e.g. 83044910, 83044911, 83044912 and so

[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

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 =>

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

Hi, I have an asterisk with a DIVA Server V-BRI-2 card, connected to a Siemens PABX. From a SIP phone, I can call other internal SIP phones, external numbers (to PSTN), but I can't call internal phones connected to the internal phone network. When I call 107, which is an internal phone, heres the logs from asterisk: -- Executing Dial("SIP/Greg-081f5a10",

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

Is there a reason the variable ${DNIS} does not get set with incoming calls via chan_capi ? Is it related to the MSN=X in capi.conf ? version = chan_capi-cm-0.6.3 example; exten => _95555555XX,1,NoOp, ${EXTEN}, ${DNIS} == ISDN1: Incoming call '0400000000' -> '95555555' -- Executing SetCDRUserField("CAPI/ISDN1/955555 55-135", "Incoming") in new

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 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, 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 =

I'm setting up an asterisk server to allow our PBX to make calls out via VoIP, but when it calls out I get this message: chan_capi.c: did not find device for msn = (eg no msn) Which would be correct because at that point I've only asked for an outside line. I'm using CAPI obviously, and my config is: [ISDN1] ntmode=yes isdnmode=did incomingmsn=* immediate=yes controller=1 group=1

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Ahoy I'm trying to setup Asterisk on debian etch (with the debian packages) with a "Fritz!Card PCI" ISDN card and chan_capi. Everything seems to be configured the right way (excerpts below), Asterisk seems to see the ISDN-card but if i try to place a test-call from the outside i don't see anything on the asterisk-console (set debug

Hi all, I try to connect two asterisk-server together. There is a server (obelix) which receives a call. This call should be transfered to another server. In my dialplan at obelix I have the following: exten => 920622201,1,Dial(SIP/outbound:geheim at asterix.local:${EXTEN}) exten => 920622201,n,Hangup exten => i,1,Congestion exten => t,1,Congestion If I call the number 920622201