similar to: Problem with Capi Channel

-----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

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 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).

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 ==

I tried to setup an PBX System with the following parties: 1. SIP Phones 2. One AVM ISDN Card with CHAN_CAPI for Outbound dialing and receiving external calls 3. One HFC-S PCI Card in NT-Mode with ZapHfc for physical ISDN Phones. 1+2 work perfect I have problems with part 3 Card is working, drivers loading fine, asterisk initializes chan_zap without problems. Demo did work. My problem is

How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten => 492,1,Answer exten => 492,2,eagi,InfMsg -s 1 exten => 492,3,Hangup() It doesn?t work my * report... -- Executing [92@telpin-112:2] EAGI("Zap/4-1", "InfMsg -s 1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg

dear list if I use EXEC in an agi script I get the following doing EXEC VoiceMailMain -- AGI Script Executing Application: (VoiceMailMain) Options: ((null)) May 21 04:25:10 WARNING[1209214400]: chan_phone.c:422 phone_read: Error reading: Resource temporarily unavailable May 21 04:25:10 WARNING[1209214400]: res_adsi.c:205 __adsi_transmit_messages: Un able to send CAS May 21 04:25:10

Hi there, im having some troubles with my asterisk service, sometimes when im trying to make an outbound call, to any of the phones configured on the asterisk box, it enters inmediatly to voicemail and then hungs up. After that its necessary to stop the service and putting up again manually. Here is a piece of my log file when a call is trying to incoming: "Jun 9 06:30:16

Hey, I'm having some errors whenever I dial out and I can't dial in at all. I'm using NuFone as my provider just so you know. Jan 11 17:39:46 WARNING[1771]: chan_oss.c:413 soundcard_setinput: Unable to re-open DSP device: No such device Jan 11 17:39:46 WARNING[1771]: chan_oss.c:572 oss_write: Unable to set device to input mode Jan 11 17:39:46 WARNING[1771]: app_dial.c:359

Hi, I've got an important question: I use an E100P directly connected to PSTN, but it does not *really* work as it should be: exten => 1000,1,Dial(Zap/1/1234) BUT: It does NOT dial "1234" but it says in debug mode: -- Called 1/72976451 Apr 26 00:53:00 WARNING[10251]: chan_zap.c:5979 zt_pri_error: PRI: !! Facility message shorter than 14 bytes -- Channel 1, span 1 got

ser forwards a sip message with extension 99999996 to asterisk which plays my 'userisoffline' message and hangs up and should stop here but instead asterisk continues to process the match everything extension ._ and dials out which is not what I want... if I change the starting priority of the Dial app to a higher level than 3 asterisk stops after the hangup but then doesn't accept

I am trying to put a call on a E1 ISDN : The configuration are simple: zapata.conf : [channels] context=inbound switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ;echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;immediate=yes immediate=no callerid => asreceived amaflags

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)

I am having a strange problem with an asterisk servier using R2 Unicall in Mexico. Most calls go through fine but some of them give me an error like this: -- Executing Dial("SIP/86-db41", "Unicall/g2/014448343600") in new stack -- Called g2/014448343600 Feb 9 21:44:39 WARNING[23069]: chan_unicall.c:2644 handle_uc_event: Unicall/2 event Dialing Feb 9 21:44:45

We're using Cisco phones running skinny protocol. When I call other extensions I don't get a ringtone, although the remote end does ring and when answered we get clear two way audio. When I call a queue from a skinny phone then I don't hear the announcements. Likewise we don't hear music on hold on these phones, although we can see mpg123 in the process list and ls -l the fd

Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a on a Debian 3.1 The system is connected with an HFC card directly to the telco line card is in TE mode and signalling used is bri_cpe_ptmp I am able to dial out some "numbers" and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... If i use a normal

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

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

Dear all I have asterisk 1.4.11 on centos 4.x i have installed 2 PRI on is asterisk and it is working fine but i got this DTMF error on asterisk CLI what is it ?? -- Zap/36-1 is ringing -- Zap/36-1 answered SIP/5406-9fa59770 -- Channel 0/1, span 2 got hangup request, cause 31 [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: Unable to forward voice or