similar to: capi_request: didn't find capi device with outgoing msn =

Hello, I have som problems with my BRI/capi setup. I manage to call in to the system (some rows below). ---------------- -- Executing Dial("CAPI[contr1/16453]", "SIP/BYEXTENSION@janm|10") in new stack -- Called s@janm -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing -- SIP/janm-63f5 is ringing ---------------- But I can't make outgoing calls from

hi when two calls are active on controller 2, chan_capi won't use controller 1. this is with AVM C2 roy -- Executing Goto("SIP/torgeir-b476", "capiring|BYEXTENSION|1") in new stack -- Goto (capiring,90044875,1) -- Executing Dial("SIP/torgeir-b476", "CAPI/22545066:bBYEXTENSION|120|Ttr") in new stack == data = 22545066:b90044875 ==

Vincent Guidoux schrieb: > Now i have a un new prob > > Executing Dial("SIP/2500-0bbb", "CAPI/@4202270:0796273153|30|r") in new > stack > Jan 17 13:14:39 NOTICE[4146]: chan_capi.c:1173 capi_request: didn't find > capi device with outgoing msn = 4202270. you should check your config well the error message says it all. 'you should check your

Hi all, I have installed a test machine with asterisk in order to try it. I have a problem with capi channel (chan_capi 0.3.4a). When an external call directed to an internal Ip phone is not answered I obtain this warning repeated many times: .... .... Jul 12 16:13:43 WARNING[1209214400]: app_dial.c:302 wait_for_answer: Unable to forward frame Jul 12 16:13:43 WARNING[1209214400]: app_dial.c:302

Hi all, I am trying to get Capi Dial to use a specific outgoing msn. I can't get it to work. If I make a test call to 0703241494 (same isdn line, just one of the other numbers) I don't get CLID at all. Any ideas? ; use 0703241432 as outgoing msn exten => _070.,1,Dial(CAPI/@0703241432:${EXTEN}|30|r) in capi.conf I have: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8

Hi there, I searched in mailinglist and in web, but no answer to my problem... Only this post with no answers: http://lists.digium.com/pipermail/asterisk-users/2004-March/038994.html I'm using CVS Asterisk (05/17/04) with chan_capi 0.3.1. (multiple controller support). In my Asterisk-box there are 2 Fritzcards (module for second card compiled with changes on sourcecode found in the web).

Hello, I'm using asterisk 1.0.7. with a somewhat advanced setup with IAX and CAPI as channels. A call comes in via IAX2 and should be redirected to CAPI. So I wrote the following dialplan: [fromiax] exten => _8XXX,1,Answer exten => _8XXX,2,Dial(CAPI/265:B${EXTEN:1},,r) [fromcapi] exten => 265,1,Answer exten => 265,2,Dial(IAX2/PoC/11@from-lw) exten => 265-BUSY,1,Busy exten

Good day, The unusual thing about this problem is that it doesn't occur just during a CAPI call, or just during an IAX/SIP call. Only during IAX/CAPI I'm having some trouble with the CAPI interface and it only occurs when a call comes in on an IAX channel and goes out the CAPI interface. The capi debug in the asterisk console is below as well as the relevent parts of .conf files from

hello, sometimes my capi_channel stop works - e.g. when i try to call number which does not exist ( typo error ) and i must restart asterisk. following lines appears in the log files : ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free channel on controller 1! will continue searching. ERROR[393234]: File chan_capi.c, Line 1050 (capi_request): no free b channel on controller 1!

Hi, after having read so much about Asterisk, I went on and tried out to create a little sample-setup. I'm using a Fritz Card USB with the AVM Capi Driver and two X-Lite Softphones. Dialing between the softphones makes no problem. Calling the MSN fron an external phone also works. I'm getting to the asterisk demo-voicebox which works flawlessly. Now may next step has been to enable

hi, klaus Console output: *CLI> -- Registered SIP '' at 192.168.16.114 port 12410 expires 1200 -- Executing Goto("SIP/ola-5a9c", "capiring|BYEXTENSION|1") in new stack -- Goto (capiring,81520400,1) -- Executing Dial("SIP/ola-5a9c", "CAPI/22545079:bBYEXTENSION|120|Ttr") in new stack == data = 22545079:b81520400 == capi request

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 all, I've recentrly starting to play around with *, when all I wanted is to configure an fritz ISDN card with A@H. Currently I'm stuck at the phase of what do I do with capi after everything is installed. I'm trying to understand how to setup incoming and outgoing calls at A@H since I'm getting a bit lost with the default dial plan. It seems that * answers but disconnect

Hi, I am having trouble configuring CAPI so that call transfers work. I make a SIP call to asterisk which goes out on ISDn via CAPI. Then I try to do a transfer from the SIP phone which doesn't work and results in the call being disconnected. The error message given by asterisk is that it chan_capi can't find an entry for the outgoing msn for the transfer however the outgoing msn is the

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

Kudos to the CAPI developers. I would like to have multiple MSN's on my ISDN Bri lines. I see all the cool features but cannot find any examples or guides to build from. Currently running Diva Eicon Cards with CAPI from http://www.junghanns.net <http://www.junghanns.net/> I would like to route calls to sip phones via msn. Set up callgroups etc. Can anyone share

Hi, I'm new to asterisk and try to get it work with capi.so. When I try to start asterisk with "asterisk -vvvvc" I get the following errors. I couldn't find any hint on the net what may be wrong in my configs. Has anybody got a hint? Here is the error output: [capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Jun 23

Hello! How can one select outgoing MSN when dialing out from ttyI-interfaces? I have successfully done this with CAPI e.g... exten => _XXXXX.,2,Dial,CAPI/60:bBYEXTENSION ...in extensions.conf. Currently correponding for my ISDN modem interface is... exten => _XXXXX.,2,Dial(Modem/g1:${EXTEN}) ...but this selects only MSN of outgoing group g1 for dialout MSN number. I also tried to

I've the following setup : sip phone -> ser (auth and routing) -> asterisk with capi isdn when I call a pstn number everything works fine, but I can't hear anything till the called answer. this is the output from a test call : -- Executing Playtones("SIP/2.7.184.61-08152880", "dial") in new stack -- Executing Dial("SIP/2.7.184.61-08152880",

Hello. If I receive a Phone call by ISDN or from SIP Provider, the Asterisk make some errors and the SIP Client don't react. The messages from Asterisk in verbose mode: er will net. Asterisk messages in Terminalmode: parse_srv: SRV mapped to host sip-ha.web.de, port 5060 Mar 10 00:02:17 NOTICE[5776]: chan_sip.c:7191 handle_request: Failed to authenticate user "unknown"