similar to: Zaptel error: Unable to create channel op type 'Zap'

I got the same problem yesterday and i just ch'ange the kernel from 2.6.12-1.1372_FC3 to 2.6.9 kernel of fedora core 3, and now it work perfectly. I think there is some problems with the new kernel of fedora. -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Altus Snyman Inviato: mercoled? 27 luglio 2005

I tend to make it pause for 10 secs when loading the module as I have had a few occurances of loading before /dev/zap has been populated. Wouldn't trust the 2.6.12 kernel as far as I could virutally throw it. Has anyone had any problems with PCI-X systems? In particular call dropping? Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Hi Everyone, I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS modules. The card works and ztcfg reports that it finds the two modules. Howevery when I try and place a call through the gateway I get the following error message. I have tried to refer to the ZAP device as ZAP/g2 etc Any suggestions? Anything that's different about Zaptel 1.4? -- Executing

Hi, I'm having a problem with the queue behaviour in my place: I have two ISDN channels to the outside (Zap/1) and two channels two a Siemens Gigaset (Zap/4). I also use a SIP gateway to call outside and have a couple of IP phones around as well (SIP). The Gigaset has about 5 phones connected to it (+base station). Whenever two people are using those, I always am blocking two internal

Hi list, any one can let me his config files for interconecting a Meridian Op 11 and Asterisk via a E1 PRI CARD. Actually i need the nortell config part, becouse my client nortell provider doesn't know how to config the PRI card at his part. Thanks all. -------------- next part -------------- An HTML attachment was scrubbed... URL:

when placing calls to the system through SIP, I got these messages, Dec 19 00:26:55 WARNING[5570]: channel.c:2571 ast_request: No translator path exists for channel type Zap (native 68) to 256 Dec 19 00:26:55 NOTICE[5570]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) any explanation for this? Thanks, -------------- next part -------------- An

Before you jump ahead, yes I do have chan_zap.so loaded.. Call Flow: Asterisk 1 --IAX2--> Asterisk 2 ---> PRI -- Accepting AUTHENTICATED call from 22.22.22.22: > requested format = ulaw, > requested prefs = (), > actual format = ulaw, > host prefs = (ulaw|alaw|gsm), > priority = mine -- Executing

Well, I'm experiencing a similar problem with my setup... debian etch, asterisk 1.4.2, zaptel 1.4.1, ... I cannot find the chan_zap.so module file anywhere, tried recompiling with zaptel 1.4.0... no change... I tried 'make menuselect', and going to the channels-part, chan_zap is marked XXX -> dependencies missing: and this is the message for it, as an explanation. Zapata Telephony

Hi, I just compiled and installed Asterisk-1.4.2 along with zaptel-1.4.1 and libpri-1.4.0 on a Debian machine with a TDM400P card. Everything goes ok but when I try to make a call through the ZAP channel get an error message about NO ZAP CHANNEL AVAILABLE. Ztcfg and zttool show the card correctly installed. When I tried to use the debug command ZAP SHOW, it was not present in the CLI. My

Asterisk Users, I have 3 FXO modules with the TDM400P Digium Card. I can dial into the Asterisk rings my Sip phone, but dialing out with my SPA941 phone through the zap channel is a problem. I keep getting this message on the Asterisk CLI. What am I doing wrong? Thanks in advance. -- Executing [103 at default:1] Dial("SIP/200-006fa300", "{Zap/g0/{EXTEN:1}") in new

(this time with subject....) Hello, I?m trying to get Asterisk to accept incoming calls from budgetphone.nl. When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy tone. I tried X-lite, which worked perfect, so my modem (with nat) probably is not the problem. I did a sip debug and got the following output. Because I?m new to Asterisk I can?t get the error why this is not

Hi Group, I have a problem with my FXO interface. It seems that asterisk cannot see any configured channels, only pseudo. Pls see the error and my config below. Rgrds Markus ----- May 2 15:09:00 NOTICE[17327]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel

Hello, I'm just starting out with asterisk and I'm playing around with the system. Currently I have a Digium TE210P connected to a PRI on the Asterisk server. I have a SIP soft phone on my laptop for testing that is working fine. When I try to place a call from my soft phone I get this from Asterisk: May 1 09:11:41 NOTICE[20098]: app_dial.c:1029 dial_exec_full: Unable to create

Updated the kernel on a fc3 box from 2.6.9-1.667 to 2.6.12-1.1372_FC3 today. Now the cvs-head for zaptel won't compile (libpri and asterisk does). The problem seems to be a symlink issue with the zaptel/Makefile looking for: KSRC:=/lib/modules/$(KVERS)/build and it doesn't exist. The kernel source for 2.6.12 was applied succesfully (supposedly) but a find on "build" does not

Here is a log from a recent call made out on a ZAP channel from a SIP phone inside my network. For some reason, CDR is billing time even though the "busy tone" was detected. It's also logging the call as ANSWERED. Is this normal behavior? Seems a little odd to me. I have this as the first 3 lines of my zapata.conf [channels] busydetect=1 busycount=3 CVS HEAD updated late

I'm having problems getting an Xorcom USB Bri 4 dialling out in Australia. I can receive calls into the system without an issue, but I can not for the life of me dial out of the system. Below are my configs, I'm hoping its something simple that I just can't see as I've been looking at it for to long. Can any one point me in the right direction. P.S. Yes it is meant to be in TE

Hi My asterisk output is: chan_sip.so => (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 201 -- Executing [907768385144 at default:1] Dial("SIP/201-083e75c0", "ZAP/g1/907768385144|60") in new stack [Oct 4 11:54:27]

Hi list, After upgrading from Asterisk v1.2 to v1.4.14, all kinds Zaptel error messages related to my HFC-S PCI card disappeared, but now I can't access the card's resources because it always seems to be busy. Any idea why? Thanks, Jaap PS -- Below is some info regarding my configuration. =========================== Zaptel version: 1.4.7 (incl. firmware and modules). OS: Debian

Hi, I'm using the macro below in extensions.conf for most of my outbound calls. One issue with my current configuration is that when I make an outbound call it doesn't properly detect that my PSTN line (Zap/1) is busy with another call and then overflow to my outbound IAX connections. I think the root cause is that DIALSTATUS gets reported as BUSY. The debug output is below. My desired

Hi, Asterisk Version : 1.2.15 Card : TDM11B (1 x FXO , 1 x FXS) I have internal dialling working okay SIP->ZAP (analogue phone) and ZAP (analogue phone) -> SIP. The problem comes when I try and make a outbound call. Here is my extensions.conf :- Code: [incoming] exten => s,1,GoToIfTime(17:00-09:00\mon-fri\*\*?outofhours|s,1) exten => s,2,GoToIfTime(*\sat-sun\*\*?outofhours|s,1)