...which, after upgrading to Dahdi 2.4.0 seem to be gone. Not sure if
2.3.0.1 was the culprit, or just the fact that sonething was stuck (stopping
dahdi took about 5 attempts because it was apparently in use, even if
Asterisk had been shutdown).
Hopefully this isn't just the calm before the
(sorry for the series of email, I realize I seem to be having a discussion
with myself).
I was wrong, the problem isn't fixed. Is having IRQ spikes every 10
minutes (under no load at all) the norm with Dahdi hardware?
Mike
-Original Message-
From: Mike [mailto:l...@net-wall.com]
Dear Leif;
Thanks a lot for ur kindly reply.
Well, based on what I have to decide to go for 1.6 or 1.8 version if I am at
1.4 version?
About the H323 gatekeeper functionality, I was hearing there is some work on
this direction at the 1.6 and 1.8 versions, but I was would to hear any
Hi All;
Anyone has tried to use A2Billing with Asterisk 1.6 and 1.8 to confirm that is
working fine and it is same as 1.4?
Appreciate ur kindly help.
Regards
Bilal
--
_
-- Bandwidth and Colocation Provided by
r...@sangoma-testing:/home# cat /lib/modules/2.6.26-2-amd64/build/.config
cat: /lib/modules/2.6.26-2-amd64/build/.config: No such file or directory
r...@sangoma-testing:/home# cat /usr/src/linux/.config
cat: /usr/src/linux/.config: No such file or directory
r...@sangoma-testing:/home# uname -a
Sascha,
Could you possibly share what you have done to get the 9971 working? I have one
of these and would dearly love to get it talking to my Elastix box with or
without annoying beep :-)
Thanks,
DT
From: Sascha Ferley sascha.fer...@infineon.net
To:
On Tue, Sep 28, 2010 at 09:28:24AM +0800, Ron wrote:
got this problem that IP phones could not re-register to my server. even
if device is power cycled it still would not register. the solution i
found was to change the SIP port settings on the phone and it will
register. but after
Hello,
Following my first mail about this issue [1], I think I know now what
the problem is.
When I have both lines being used and a third call comes in, the person
calling doesn't get a busy tone, he gets something like line unavailable.
I've been debugging mISDN and I think the reason is
hi daniel
thank you for your reply, i have enabled NAT keep-alive and NAT-Mapping
on the Linksys devices.
i have enabled qualify before but it seems it's worst as the linksys
devices keeps on rebooting and there was one issue that when i set
qualify to all users my server bogged down and
{\rtf1\ansi\ansicpg1252\fromhtml1 \fbidis \deff0{\fonttbl
{\f0\fswiss\fcharset0 Arial;}
{\f1\fmodern Courier New;}
{\f2\fnil\fcharset2 Symbol;}
{\f3\fmodern\fcharset0 Courier New;}}
{\colortbl\red0\green0\blue0;\red0\green0\blue255;}
\uc1\pard\plain\deftab360 \f0\fs24
{\*\htmltag19 html
You have to increase the time of expiration for the Register...on linksys
devices is located on Proxy and Registration section under the EXTN: (Where
N is the extension number)
Try putting this to: 3600
To check wheter or not is loosing Register, try with ngrep-sip and check it:
ngrep -p -q -W
Hi!
Looks like I still don't understand how SHARED works :-(
exten=6052,n,Dial(SIP/6052,,M(test))
exten=6052,n,Dial(SIP/6052,,M(test^${CHANNEL}))
Please check if in Asterisk 1.6 the Syntax for passing arguments to the M
option of Dial() has changed.
[macro-test]
exten =
Hello,
I'm checking this:
[Sep 28 13:32:46] NOTICE[30360] chan_zap.c: PRI got event: HDLC Abort (6) on
Primary D-channel of span 1
[Sep 28 13:32:46] NOTICE[30363] chan_zap.c: PRI got event: HDLC Abort (6) on
Primary D-channel of span 4
[Sep 28 13:32:46] NOTICE[30363] chan_zap.c: PRI got event:
Hi List.
It is possible to travel over the X.25 protocol on Asterisk SIP?
--
Atenciosamente
Daviramos Roussenq Fortunato
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
On Tue, Sep 28, 2010 at 07:08:36PM +0800, Ron wrote:
i have enabled qualify before but it seems it's worst as the linksys
devices keeps on rebooting and there was one issue that when i set
qualify to all users my server bogged down and asterisk crashed.
Never seen these problems before
Hi List.
It is possible to travel over the X.25 protocol on Asterisk SIP?
--
Atenciosamente
Daviramos Roussenq Fortunato
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
Hi Danny
On the pap2 by default it is set to 3600 and i have not change that.
by the way, is the NAT keep-alive same with the NOTIFY message? coz i am
seeing my asterisk respond to those as bad event could that be causing
it to loose the registration?
here's the registration from ngrep:
U
28.09.2010 15:35, Philipp von Klitzing ?:
Hi!
Looks like I still don't understand how SHARED works :-(
exten=6052,n,Dial(SIP/6052,,M(test))
exten=6052,n,Dial(SIP/6052,,M(test^${CHANNEL}))
Hello!
Thank you!
I can pass this constant , but I need RTCP stats
And this
On 09/28/2010 07:05 AM, Daviramos Roussenq Fortunato wrote:
Hi List.
It is possible to travel over the X.25 protocol on Asterisk SIP?
Your question doesn't make any sense, but we can try to answer anyway:
SIP is transported over UDP or TCP, which are transported over IP. It is
possible to
28.09.2010 16:19, Dmitry Melekhov ?:
btw, about bridged variables- they are really what I need.
Looks like there is bug in asterisk- if call is dropped from dahdi side-
there is no info in these variables.
I think I have to fill bug.
Thank you!
I got what I want :-)
Thank you again!
We need to monitorate the E1 with nagios, somebody did this? any ideia?
Thanks in advance!
--
Atenciosamente,
---
Dario Quiroz
(71) 9275-9080
gtalk: darioqui...@gmail.com
Danny Dias wrote:
PRI got event: HDLC Abort (6)
Google is your friend:
http://lists.digium.com/pipermail/asterisk-users/2005-June/107299.html
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
Thanks ,it solved by adding
insecure=very
regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Tuesday, September 28, 2010 2:16 PM
To: Asterisk; Asterisk List
Subject:
what do you want to monitor?
I ended up with MRTG graphing the Incoming/Ringing/Established calls.
On Tue, Sep 28, 2010 at 4:22 PM, Dario Quiroz darioqui...@gmail.com wrote:
We need to monitorate the E1 with nagios, somebody did this? any ideia?
Thanks in advance!
--
Atenciosamente,
Are you monitoring some dahdi hardware or a separate black box?
If dahdi, you could write a nagios plugin in shell with something like this:
ALARMS=`dahdi_scan | grep alarms | grep -v OK | wc -l`
and then set the appropriate exit code if ALARMS is not 0.
-M
On Tue, Sep 28, 2010 at 9:22 AM,
Enjoy...you can ignore certain T1/E1 ports if you pass in the name of the
port as an argument (I use this on ports that aren't yet connected to a
telco, but I don't want to get an alert on). I execute it via NRPE on the
Asterisk box. It will give you descriptions of which ports are bad, so you
Bro,
What OS u use ??
Maybe you can use SNMP .. snmpd its a good application
linux: apt-get install smtpd
freebsd: pkg_add -rv net-snmpd
ABS[]s
2010/9/28 Joel Maslak jmas...@antelope.net
Enjoy...you can ignore certain T1/E1 ports if you pass in the name of the
port as an argument (I use
On 09/28/2010 02:37 AM, Mike wrote:
I was wrong, the problem isn't fixed. Is having IRQ spikes every 10
minutes (under no load at all) the norm with Dahdi hardware?
Interrupt spikes on 10 minute intervals is not (should not be) the norm
with DAHDI hardware. What is the make/model of the PRI
I'd just like to thank everyone that helped me get this running. I thought I
had a FAS PRI but it turns out it was NFAS so there was no dchannel on the
second PRI. When getting the 2nd PRI changed, I received an email containing
the cheat sheet TELUS employees use which I thought I should
How do I go about finding which device is causing those interrupts? (to make
sure it`s actually Digium hardware and not something else?)
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Mike
Sent:
On 09/28/2010 11:40 AM, Mike wrote:
How do I go about finding which device is causing those interrupts? (to make
sure it`s actually Digium hardware and not something else?)
Can you let something like 'while true; do date interrupt_log.txt;
cat /proc/interrupts interrupt_log.txt; echo
Shaun,
Thanks for all the help. I tried but couldn't see a definite spike anyways.
I used mpstat to see the irq% for each CPU, and although it shows me spikes
to 100% for one CPU (not always the same one is spiking irq%) the intr/s
isn't actually spiking.
It`s like suddently there aren't more
On Tue, 2010-09-28 at 09:06 -0300, Daviramos Roussenq Fortunato wrote:
Hi List.
It is possible to travel over the X.25 protocol on Asterisk SIP?
--
Hi Daviramos,
You gotta be joking!
X.25 can only be found in telecom-musea's nowadays.
Latest development was the X.31 and X.32 interface
On Tue, Sep 28, 2010 at 11:23:16AM -0500, jeremy.hellst...@synovate.com wrote:
I'd just like to thank everyone that helped me get this running. I thought I
had a FAS PRI but it turns out it was NFAS so there was no dchannel on the
second PRI. When getting the 2nd PRI changed, I received an
I have one of my 4 PRI ports RED (that`s normal: there is a cable from my
DS3 MUX to it but there is no active PRI). I configured it for futur use,
thinking that it could just start running when that particular PRI was
activated without any asteisk modifications.
Could this be the cause, or am I
On 09/28/2010 12:53 PM, Mike wrote:
Thanks for all the help. I tried but couldn't see a definite spike anyways.
I used mpstat to see the irq% for each CPU, and although it shows me spikes
to 100% for one CPU (not always the same one is spiking irq%) the intr/s
isn't actually spiking.
It`s
Shaun,
Thanks, don`t think I don't appreciate all your time, I really do. So
you`re saying it's possibly just fake smoke from a non-existent fire, and
that nothing is actually wrong despite a high load one one of the CPUs?
That`s what it really is: I am seeing smoke and am looking for a fire,
On 09/28/2010 02:06 PM, Mike wrote:
Thanks, don`t think I don't appreciate all your time, I really do. So
you`re saying it's possibly just fake smoke from a non-existent fire, and
that nothing is actually wrong despite a high load one one of the CPUs?
That`s what it really is: I am seeing
Hi Mike,
I`m not sure, but I suggest the following tests.
The cards generate interrupts even with no load. And in real life they have
different clocks. A very small clock difference could accumulate and make
the interrupts happen at same time. USB , TCE400 or PRI card interrupt
routines could be
Thanks. Will try all that. Night time work again ;-)
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Tuesday, September 28, 2010 15:37
To: asterisk-users@lists.digium.com
Thanks. The problem is I only have two PCIe slots, and they are both taken.
I`ll definitely try disabling USB, I have no need for it right now.
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Antonio
Prata Barbosa
Sent:
Hello Ron..
The answer that i see here is only a trying to a Register...means the
REGISTRATION procedures are taking a significant amount of time.
You should get a 200 OK
Can you lease make a simple draw of your architecture? seems to be a NAT
problem, that's for sure
REgards!
2010/9/28 Ron
Does anyone know if I could use modprobe command to find out rather than
set the jumper on a Digium PRI card?
I want to find out the jumper settings on the card without opening the
box which will cause down time.
Thanks.
--
43 matches
Mail list logo
