similar to: Syntax for 2 ISDN Cards

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, i'm in debian sid 3.1 with kernel 2.6.7, * last cvs & chan_capi 0.3.4b; nt1+ with 2 bri in ptmp (http://www.voip-info.org/tiki-index.php?page=DDI) i tried to install avm c4 following step by step http://www.voip-info.org/tiki-index.php?page=Asterisk%20How%20to%20connect%20with%20CAPI step 1. i compiled capi 2.0 support in kernel

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 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 ==

Anyone here running asterisk on openvz, if so what are your experiences? Right now we are trying to tune out the resources for the difference VEs, but not with a whole lot of luck. Just wondering if someone watching could shed some like on what has worked for them, and how many exts/simultaneous calls etc are happening. Thanks Miles -------------- next part -------------- An HTML attachment was

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 all, I compiled/installed chan_capi.so without problems. When I launch Asterisk, I get the following error: --- [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338 mkif: ast_capi_pvt(91xxxxxx,*,pstn,0x2,2) (1,2,64) (0)(0.800000/0.800000) 0 Mar 30 19:47:52 NOTICE[16384]: chan_capi.c:2338

Hi, Just started to get this error after updating to the latest CVS. Asterisk dies if it can't create a channel - not so good. -- Executing SetCallerID("SIP/750-2550", "39660426") in new stack -- Executing Dial("SIP/750-2550", "CAPI/39660426:22179808") in new stack Jun 22 13:52:05 NOTICE[262161]: chan_capi.c:1172 capi_request: didn't find

Hello list, I need to record all calls. So I'm using application Monitor. Works good until someone transfers a callee to another internal extension. Example: A calls B A set B on hold A calls C A transfers B to C with SIP transfer (SIP REFER - with phone funktions and not Asterisk attended transfer). I found http://bugs.digium.com/view.php?id=0013538 . "corruptor" asked about this

Hello, I have a problem with MoH at attended transfers. - Mobile A dials into Asterisk - Asterisk dials another Mobile B - Mobile B presses *1 for attended transfer and for example 20 to dial extension 20 - Asterisk sends "Remote hold" message to Mobile A, so the carrier of Mobile A starts playing it's own music-on-hold - Mobile B hang up, so Mobile A should be connected to

Hello list, Using Asterisk 1.2.29 I use the Monitor() application. In extensions.conf I have set MONITOR_EXEC to my script (for mixing files together and convert to mp3) and I set TOUCH_MONITOR on every new channel which has to be recorded. But sometimes I'm missing the recording files. I had a look to the Asterisk-Log and saw those lines: Feb 10 15:18:57 NOTICE[16772] res_monitor.c: monitor

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

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

Hello list, I have a problem with called ZAP channels making an attended-transfer or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk is wrong. At the moment there is a bristuffed Asterisk 1.2.18 running with bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit: [default] exten => 0123456789,1,Macro(dialpstn,${EXTEN}) [macro-dialpstn] exten =>

Hello, I want to enable call forwarding at my telco. In Germany you can press *21*destination# and all calls will be redirected to the destination without interaction with any equipment on my side. How to dial this with Asterisk and Zap-Channels? It can not be send as "called number", it has to be send as "keypad facility". Anyone here with some hints? The application

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 can't make outgoing calls with CAPI (passive ISDN Fritz card). See Asterisk error below. Incoming calls and SIP to SIP calls do work. It looks like a msn mismatch in extensions.conf and capi.conf, but I can't find it. Can anyone help me find the problem? Thanks, Rob *CLI> -- Executing Dial("SIP/8112-1be9", "CAPI/356666666:BYEXTENSION") in new stack

Hello, I have a little problem with signalling. An E100p is connected to an Alcatel PBX, wich has an E1 to the outside. Located in Germany. zapata.conf: switchtype=euroisdn pridialplan=local prilocaldialplan=local overlapdial=yes signalling=pri_cpe .... With asterisk 1.0.2 I can call from a SIP phone to a phone connected to the Alcatel and the SIP number is correctly displayed at the caller.

Hi, Just wanted to let you know that we have just made our GPL toolkit "Gemeinschaft" available to the public. (Finally.) Mostly German for now - about half of the strings in the language strings file have been translated to English. I'm a software developer, not a marketing guy, so ... svn co https://svn.amooma.de/gemeinschaft/trunk gemeinschaft-trunk German readers: see

Hi, I am having some trouble with ISDN Dialout. Using a Netjet-s PCI Card. When in Minicom, the only way I can dialout is if i issue ATS18=1 First. Otherwise I get a BUSY message. So thats fine. But when I dialout from asterisk, I get an immediate hangup, so my guess is that asterisk is not issuing ATS18=1 to the ttyI device. Here are my configs, any input would be greatly appriciated.

I just downloaded chan_capi.0.3.4a.tar.gz but it will not compile on my system (Suse 9.1). I compiled and installed Asterisk and it is running, did I miss a configuration or dependency for Chan_Capi somewhere ? cheers, Mike linux:/usr/src/chan_capi-0.3.4a # make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES