similar to: ztdummy: rtc: lost some interrupts at 1024Hz.

Displaying 20 results from an estimated 1000 matches similar to: "ztdummy: rtc: lost some interrupts at 1024Hz."

2007 May 10
1
module zttranscode: what is it?
Hi, does anybody know what *zttranscode *module* *is for*?* Thanks!! Giorgio -- _________________________________________________ Giorgio Incantalupo, mailto:gincantalupo@fgasoftware.com FG&A srl - http://www.fgasoftware.com - Voice@Work - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172
2007 May 18
0
mISDN: long delay when making outbound calls
Hi, I have an Asterisk 1.2.9.1 box connected to an ISDN line via a beronet card (with ports in PTP mode). I noticed a long delay when making outbound calls, more precisely between (taken from Asterisk CLI) "Called 1/XXXXXXXXX/s" and "mISDN/1-u43 is proceeding passing it to SIP/8-5486" I searched on misdn.org but found nothing. I'd like to understand if this delay is
2008 Nov 11
1
ztdummy: rtc: lost some interrupts at 1024Hz
> I'm getting crazy about ztdummy. I have to replicate a PBX where ztdummy > is working fine but for some reason I cannot. > The two machines have the same kernel, motherboard, the same gcc version > and the same zaptel 1.4.8. On the second machine zaptel compiles without > errors and ztdummy.ko is generated but when I modprobe it I get the > following error in messages: >
2005 Oct 17
1
module loading error with Ubuntu: insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format
Hi, I am trying to use zaptel module on an Ubuntu 5.10 distro (2.6.x kernel) using gcc 4.0.2. Compilation does not give me errors so after a 'make install' I try to load zaptel module with insmod but the following error arise: *insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format* Is there anybody who can help me?? TIA Giorgio --
2007 Jul 12
0
No subject
... Activating "sip debug" shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named
2005 Aug 26
2
WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type
Hi, is there anybody who knows what this warning means?? WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com
2007 Jul 12
0
No subject
tnet.itand SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo < gincantalupo at
2005 Sep 02
1
Italy FastWeb problem: ISDN line crashes every time cisco router turns off
Hi, I live in Italy and I have an Asterisk 1.0.7 box with a HFC monoBRI card connected to my cisco router which is connected to FastWeb provider: does anybody knows why every time my cisco router turns off, my telephone connection to Fastweb drops (while internet connectior is ok)? Restarting Asterisk is worth nothing. TIA Giorgio --
2005 Aug 26
0
SV: Maximum retries error.
There is no static interval. But i found out that it was my IP-Phone Service Provider that was having serviceproblems today. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Giorgio Incantalupo Sendt: 26. august 2005 11:33 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users]
2005 Jul 26
0
include not working in bristuffed Asterisk 1.0.7 extensions.conf
Hi, I've upgraded my Asterisk to 1.0.7version patched with bristuff 0.2.0-RC8c. I'm using the same extensions.conf but it seems now include instruction doesn't want to work, here follows an extract: [inbound_menu] include => ins_exts exten => _X.,1,Answer exten => _X.,2,Wait(1) exten => _X.,3,Background(msg) exten => _X.,4,Background(3-sec-pause) exten =>
2005 Jul 28
0
Wrong cdr records
Hi Rosario, I have a problem about CDR: inbound calls are not correctly logged in CDR, it says they are always answered even if they are not. It is very strange since outbound calls and internal calls don't suffer this problem. I'll tell you more: I made Asterisk print the DIALSTATUS variable and it is ok, says BUSY when my internal hardphone SIP is busy. Or maybe it is allright and
2005 Aug 29
0
Conference and HFC card conflict: no solution??
Hi, I'm using a HFC card on my asterisk box. I tried to make a conference but it doesn't work. I read on internet to use ztdummy but my server has no uhci (only ohci but it doesn't work) so I cannot use it. I tried zaprtc but after loading the module (it appears when typing lsmod) nothing has changed. Should I buy a x100p to get the right timing? Or there is another solution? TIA
2007 Feb 27
0
rtc: lost some interrupts at 1024Hz
Hi im having this message in my console and dmesg. rtc: lost some interrupts at 1024Hz im not sure what this is. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070227/0862a1ba/attachment.htm
2004 Dec 21
2
SOHO PBX using asterisk
Hi, I'd like to build a personal PBX connecting 4 or 5 analogic phones with a ADSL line and I'd like to know what is the right card I need I visited digium site and I think TDM400 could be the right choice but I cannot understand how it works...I think it has 4 slots where 4 modules (FXS or FXO) can be inserted. How many cards do I need to connect my ADSL line to 5 phones? I think I
2008 Feb 13
2
Asterisk and fax
Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that Hylafax/Iaxmodem. And the Asterfax solution does dislike cause its licensing. The TE420B, is configured in E1
2008 Jul 09
2
cell phone hangup not getting recognised by system
Hi all, When I do a test call into the box (which is running latest version of Trixbox) it all goes fine. If i decide to hangup the cellphone (during the ivr playing options) the system does not recognize the hangup and the system continues through and ends up at the timeout option. What settings do I need to change to fix this. Is it the rxgain? If so is there something i can use to figure
2008 Apr 15
5
Conferencing..
I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey
2007 May 24
3
modprobe
Hello every boy again I have some problems with modprobe. When I type "modprobe zaphfc", this error happens "FATAL: Module zaphfc not found." And when I tyoe "ztcfg -vv" this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Someone can help me??? Thanks to all.
2009 Feb 06
2
asterisk and DNS
We've just had the problem where our DNS server went down, and * started to act "funny". Is the best solution to install a local DNS server on the * box, and have no other DNS servers ? - this is an internal app, no need for any external DNS resolution at all. Julian. ______________________________________________________________________ This email has been scanned by the
2008 Mar 11
3
E1 Card emulator?
Hello All, Does anyone know of a software emulator that can be used to simulate hardware such as an E1? I need to play with AstUnicall in a test environment and don't have access to these circuits from the US. If there is an alternate way to test/play with AstUnicall, please let me know! Thanks, Mark -------------- next part -------------- An HTML attachment was scrubbed...