similar to: segfault with chan_capi-cm 0.5.4

Hi, I've just installed Asterisk 1.2.0-beta1 for the first time and downloaded chan_capi and compiled it in and run it. (For comparison purposes, I've tried this with both chan_capi-0.4.0PRE1 and chan_capi-cm-0.5.4 and chan_capi-cm latest CVS). Whilst most things are fine, it seems that if I specify the 'b' parameter in the dial string before the number, sometimes the early

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, the following szenario leads to a problem: I connect an CAPI channel to an AGI-Script per Manager API. This Agi script starts the MeetMe-Application. The Person on the Capi Channel is now able to speak with the other conferess in the MeetMe-Room. But if the CAPI channel hangs up, the busy tone is streamed into the MeetMe Room for several seconds, until the CAPI HANGUP-Signal is finally

Hi! I have problems with the ToN configurations in chan_capi-cm. I understand how incoming calls are rewritten using national and international prefix. But for outgoing calls - what is the ToN? Further, is there any debug info of the Ton? "capi debug" and the divactrl dchannel both show the CLI and CDLI, but do not show the ToN. How can I set the ToN? I suspect chan_capi to use the

Hi, I tried to use the version 0.6 of chan_capi-cm for outgoing calls it works perfectly but for incoming calls it will not work: --- snip --- *CLI> capi debug CAPI Debugging Enabled -- CONNECT_IND (PLCI=0x101,DID=97,CID=0179903xxxx,CIP=0x1,CONTROLLER=0x1) == reventix: Incoming call '0179903xxxx' -> '97' -- reventix: info element CALLED PARTY NUMBER --

I'm getting a strange error on one of the two controllers on an AVM C2 card under chan_capi-cm-0.6.3. I have two ISDN controllers defined, both in the same group, both connections are UK ISDN2e Point to Point: On the third outbound call (both of the first two calls are handled by the second controller "ISDN2",) I get this error : chan_capi.c conf_error 0x2001 PLCI=0x301

Hi all, i have asterisk 1.0.9 with an Eicon Diva 4bri and chan_capi-cm-0.6. I have 2 NTBAs (one with did and one without). When using the one without did, i am able to place outgoing and incoming calls. When i use the NTBAs with did i have a layer 2 error. Anyone an idea? -- Executing Dial("SIP/2004-9634", "CAPI/g1/43XXXXXX") in new stack > data = g1/43XXXXXX

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

Hello, I am using a system with an AVM ISDN PCI card (fcpci) and asterisk with chan_capi-cm-0.6. The hardware is connected to a Siemens Hipath 3550 PBX. As a BRI connection has 2 channels and allows 2 simultaneous calls, numbers/MSNs 6391 and 6392 were for provisioned for each channel. The system is working (partly, read on), the trick is the correct cable wiring and setup the PBX's port

Hi all, on sourceforge.net I added the fixup release 0.5.4 of chan_capi-cm driver. The changes from 0.5.3 to 0.5.4 are: - fixed 'group' setting according to Asterisk defaults. - use SetCallerPres(prohib_not_screened) instead of CallingPres(32) for CLIR. - full CallingPres support added. - use mutex when debug/verbose messages are printed. - set dnid on incoming call. - catch errors in

Hello List, I'm trying to get a PrimuX Card (www.primuxisdn.de) working. The Manufacturer says that chan_capi (the older one) used to work. Now I'm trying with chan_capi-cm and have got the following problems: Outgoing calls "ausgehend_ueber_ntba_primux.txt" the other phone rings but when I answer the call asterisk crashes. Incoming: first tried with immediate=no

Hi all, I would like to announce the first release of the chan_capi channel driver on sourceforge.net The package is available for download with name chan_capi-cm-0.5 and is the current CVS HEAD. It is derived from the chan_capi-0.4.0PRE1 of kapejod. The main changes are: - complete rework - fix race-conditions - fix call state handling - rework of debug/verbose messages - added capiFax

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 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

Hello all, I'm having problems getting chan_capi 0.3.5 to work well with an Eicon Diva Server card using using the driver from linux kernel both 2.6.10 and 2.6.12 (vanilla versions). I have a (really two) producci?n system(s) running chan_capi in another identical Eicon Card using kernel 2.4.27 and the Diva Server drivers from Eicon. I installed an compiled the source level rpm

I'm trying to compile chan_capi-0.5.4 on Slackware 10, but I have bunch of errors. (By the way, can I use chan_capi for ISDN card with winbond w6692cf chipset?) I'm not a linux expert, still :-) Before compiling, when I type "modprobe capi" to load capi module, and then "lsmod", i get list of modules: capi 6208 0 kernelcapi 30496 1

Hi all, it took a while, but on sourceforge.net I added the new release 0.6 of chan_capi-cm driver. Note: dial string and capi.conf has changed. The main changes are: - added 'relaxdtmf'. - more BSD compatibility - correct PROGRESS handling - added verbose text for capi info/reason error messages. - use correct facility-selector for echo-cancel - added application capicommand() for CAPI

O.K. thanks a lot, Felix and Peer Oliver. But somehow asterisk keeps telling me while startup: [chan_capi.so] => (Common ISDN API for Asterisk) Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr1 Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr2 Jan 24 14:30:47 NOTICE[9796]: chan_capi.c:3271 load_module: Unused contr3 Jan 24 14:30:47

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/

Dear list, today I installed a new asterisk machine, bound to replace my current pbx. I am using a Fritz ISDN card; on the old machine with the drivers coming together with the super-old rpm asterisk installation of SUSE 9.2. The new machine is finally on asterisk 1.0.9, with chan_capi 0.5.4; now I am doing a nightly test. Apparently I can receive calls, but I can't dial out. I seem to