similar to: capiECT problem

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to get callers forwarded to by mobile phone when they dial a certain digit. In my extensions.conf I have defined the following: [279xxxx] exten => s,1,SetLanguage(de) exten => s,2,Wait,5 exten => s,3,BackGround(demo-congrats) exten => s,4,Goto(boksa,#,1) exten => 3,1,VoiceMail,u1 exten => 4,1,VoicemailMain exten =>

Hi all, does anyone have capiECT working? I can't get it to work - please see my logfiles below. While using an * CVS version of late September and chan_capi-0.2.5 (I guess), it worked!!! (I know, never change a running system ... should've backuped ... etc.) Here's the setup: NT----AgfeoISDNPBX----AgfeoISDNPBX----AnalogPhones in between the two PBX I attached my * server with a

Hi, I?ve strange problem when I?m making a call from SIP (Cisco 7960) to capi (Fritz PCI). When I call a national number, I?m hearing the ringtone when the called party is ringing but when I call an international number, I don?t hear the ringtone and I?ve a silence until the called party answers. Both call are going through the same extension. Here?s 2 log files, one with a national number and

Hello, I am trying to configure asterisk 1.0.7pre to get incoming calls from an ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is that the context is not recognised in the /etc/asterisk/capi.conf I have in /etc/asterisk/capi.conf 's section "[interfaces]" the following directive context=isdn and the following directive in /etc/asterisk/extensions.conf in

Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI> -- data = @89930:0107901723168212 -- capi

Dear list, I tried to install the CAPI support RPM from the SUSE 9.2 disks. YaST tells me it is happy with the process. The Asterisk release I am using is the one that comes packaged in RPM format, also included in the distribution. Still starting asterisk with the usual asterisk -vvvc I see that something goes wrong. [app_capiCD.so]Dec 23 19:21:45 WARNING[1076850816]: loader.c:242

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",

Hi, I installed Asterisk with CAPI support. Everything works fine while starting Asterisk, but when a call comes in Asterisk hangsup the call after two times of ringing. The output is like: Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg: CONNECT_IND ID=002 #0x011d LEN=0048 Controller/PLCI/NCCI = 0x101 CIPValue = 0x10

Hi all ! We have 3 NTBAs which are all going to our existing PBX. Our areacode is 06003 and our DDI enabled number 9141. I want to exchange that PBX with Asterisk, but still struggle to get it working. My CAPI.CONF is currently like this: [general] nationalprefix=0 internationalprefix=00 rxgain=0.800000 txgain=0.800000 [interfaces] context=in-capi msn=60039141 incomingmsn=* controller=1

Hi, whenever I include a "Ringing" Line in some Voicemail Extension I get an error when a call from the outside (via ISDN) comes in, but it works when an internal (SIP-phone) calls the extension. Here is my configuration for testing: ------------extensions.conf------------ [isdnext] ; strep external "101", our number and leave only extension exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 i fixed wrong capi.conf (there was [controller1] after [interfaces]) now capi.conf is: ; ; CAPI config ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=855285,859609 incomingmsn=* controller=1,2,3,4 softdtmf=0 accountcode= context=local ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1

Hi * Fans, I have some fritz cards now, followed instructions from stuart hirsts email of Jun 28: -> Thanks for your info but I think I have it working at last. Below are the steps I took which might help others. 1) Download the PCI AVM drivers from ftp://ftp.avm.de/cardware 2) Download the Chan_capi from http://www.junghanns.net/asterisk/ 3) tar -xvzf fcpci-suse8.0-03.09.10.tar.gz which

Hello List, because this is my first post to this list, i'd like to introduce myself. My name is Thorsten M?ller, 26 years old and live near Frankfurt/Main in germany. Okay, now to the reason for this posting: I just installed my first asterisk server (Debian 2.2 Kernel 2.4.18-686) with AVM ISDN Fritz PCI card (passive). I followed the configuration on http://voip-info.org and the

Hi, i use chan_capi 0.3.1 with asterisk (stable branch cvs) and 3 x c4 active ISDN card. From Controller 1 - 7 there are no problems making calls between asterisk and the pstn. But when i make calls from controller 8 - 12 i get on every controller (8 - 12) a segmentation fault in asterisk :( I tried different linux distributions (gentoo 2004.1, redhat 9.0 , suse 9.1) but same error.

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

Hi there, I just recently installed asterisk from cvs on SuSE 8.2 successfully. I was quite surprised since asterisk seems to be a quite complex and powerful application. I also installed chan_capi 0.3.5 and am using a Fritz!PCI card at an ISDN line (PtMP). I already got some basic extension schemes to work, e. g. the capi callback example from junghanns.net. One of my goals is to set up an

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"

Hi, Regarding capi debug, I don't know how to translate reasons like 0x3302 or infos like 0. I didn't find any 'translator' googleing capi debugging. Do you know about any 'translator' for this or should I be as clever as to know what a reason 0x3302 is? What is this debug for if I can't interpret it? Kind regards, Victor. >From capi debug: == CAPI Call

Hi, I'm trying to get a fritz working in our asterisk box, however I'm getting the following error and was wondering if anyone could be of any help: -- Executing Macro("SIP/2002-5501", "enum-call|call|0405152568") in new stack -- Executing Goto("SIP/2002-5501", "call|0405152568|1") in new stack -- Goto (call,0405152568,1) ==

Hi, i want forward a call witch is comming over isdn (avm b1 witch i have) out to isdn (same card 2. b channel). The call is comming (one b channel open one is free) the forwarding is processed (snom 200) all seems correctly. Then the message that the b channels all busy, but so is it not. Forwarding to a sip phone works. Can anyone help me with that ? nicolas SNIPS: ------ ==