Giuseppe Parlato
2006-May-02 05:17 UTC
[Asterisk-Users] Zapata Telephony interface and torisa module error
Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000 May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian insmod: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: insmod char-major-196 failed Is the torisa module the problem? ..end Zapata Telephony Interface, is it the torisa module? Giuseppe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060502/c7265f73/attachment.htm
Hello, I have some problems with a new configuration: I always have on my asterisk console the message: chan_iax2.c:5886 update registry: restricting registration for peer '19' to 60 secondes I connect only two ip phone with iax protocol. And when i want to call 19 phone, it's hangup. No information in console view, or in file /var/log/asterisk/messages. Do you have any idea? My files a there: extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password TRUNK=Zap/g2 TRUNKMSD=1 [INTERNAL] exten => 19,1,Dial(SIP/19,20,tr) exten => 19,2,Voicemail(u19) exten => 19,hangup exten => 19,102, Voicemail (b19) exten => 19,103,Hangup exten => 20,1,Dial(SIP/20,20,tr) exten => 20,2,Voicemail(u20) exten => 20,hangup exten => 20,102, Voicemail (b20) exten => 20,103,Hangup iax.conf: [general] bandwidth=low disallow=lpc10 jitterbuffer=no forcejitterbuffer=no [19] type = friend username = 19 secret = 19 host=dynamic context = INTERNAL mailbox=19 [20] type = friend username = 20 secret = 20 host=dynamic context = INTERNAL mailbox=20 Best regards, -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montlu?on T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com
Kristian Kielhofner
2006-May-02 07:43 UTC
[Asterisk-Users] Zapata Telephony interface and torisa module error
Giuseppe Parlato wrote:> > Looking at my log file I found the following error: > > May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on > major 196 > May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000 > May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded > May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: > init_module: Input/output error > May 2 12:00:45 debian insmod: Hint: insmod errors can be caused by > incorrect module parameters, including invalid IO or IRQ > parameters. You may find more information in syslog or the output > from dmesg > May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: > insmod char-major-196 failed > > Is the torisa module the problem? ..end Zapata Telephony Interface, is > it the torisa module? > > Giuseppe >Giuseppe, You probably just don't have the original Tormenta ISA card. You can ignore this message. Better yet, configure your system to not load the torisa.o module. -- Kristian Kielhofner
yes, with ChanSpy. On 5/3/06, Olivier Saulnier <steganux@steganux.com> wrote:> Hello, > > is it possible to listen a conversation in real time, without recording it? > > Best regards, > > -- > Olivier Saulnier > STEGANUX > 35 Quai Louis Blanc > 03100 Montlu?on > T: 04.70.02.80.55 > F: 04.70.02.80.57 > http://www.steganux.com > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856
If its going over a zaptel interface then you certenly can. See http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ZapBarge On Wed, 2006-05-03 at 15:52, Olivier Saulnier wrote:> Hello, > > is it possible to listen a conversation in real time, without recording it? > > Best regards,
Look for ChanSpy Application in voip-info.org Regards On 5/3/06, Olivier Saulnier <steganux@steganux.com> wrote:> Hello, > > is it possible to listen a conversation in real time, without recording it? > > Best regards, > > -- > Olivier Saulnier > STEGANUX > 35 Quai Louis Blanc > 03100 Montlu?on > T: 04.70.02.80.55 > F: 04.70.02.80.57 > http://www.steganux.com > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"
Ok, thanks everybody :-) -- Olivier Saulnier STEGANUX 35 Quai Louis Blanc 03100 Montlu?on T: 04.70.02.80.55 F: 04.70.02.80.57 http://www.steganux.com
Anytime! Just one thing, do not hijack a thread again. :) http://en.wikipedia.org/wiki/Thread_Hijacking Olivier Saulnier wrote:> Ok, thanks everybody :-) >
Hello to all
Im trying to make SIP URI calls with my asterisk@home, and I followed this:
http://slacker.com/~nugget/projects/asterisk/page7
So I putted in extensions.conf:
MYDOMAIN => xxx.xxx.xxx.xxx
MYFQDN => xxx.xxx.xxx.xxx
[macro-uridial]
exten => s,1,NoOp(Outbound SIP URI call ${ARG1})
exten => s,2,SetCIDNum(5125380508)
exten => s,3,Dial(SIP/${ARG1})
exten => s,4,Congestion()
and in extensions_custom.conf :
[from-internal-custom]
exten => _.,1,NoOp(Incoming Call from house extension ${CALLERID} for
${EXTEN}@${SIPDOMAIN})
exten => _.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten => _.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten => _.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten => _.,5,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten => _.,6,NoOp(@${SIPDOMAIN} is remote, forwarding...)
exten => _.,7,Macro(uridial,${EXTEN}@${SIPDOMAIN})
exten => _.,8,HangUp()
exten => _.,10,Goto(noturi,${EXTEN},1)
exten => h,1,HangUp()
[noturi]
include => local
include => trunkld
include => trunkint
include => emergency
Then, I try to call 613@fwd.pulver.com and the call fails:
asterisk debug:
Looking for 613 in from-internal (domain fwd.pulver.com)
Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:5060:
SIP/2.0 404 Not Found
If I have "_." in [from-internal-custom] why do the call fails?
Thanks
Joao Pereira