Am 30.12.2012 04:24, schrieb Jeff LaCoursiere:
On 12/29/2012 05:20 PM, Mr. James W. Laferriere wrote:
2003, 24471
2004, 48608
2005, 59116
2006, 41215
2007, 26414
2008, 20746
2009, 18304
2010, 14948
2011, 11588
2012, 7542
If you remove the top-posting thread, it may cut it in half again.
j
Hello everybody!
Lately I've had experiences that I'd like to share with you:
I did a some faxing over VOIP during the last two years. Not that much,
lets say 1 fax per day on average. The setup is
Old analog fax machine - Linksys PAP2 ATA - Asterisk 1.2 - DSL -
VoIP Provider
I would
Am 08.08.2011 18:28, schrieb Kevin P. Fleming:
On 08/08/2011 07:51 AM, Norbert Zawodsky wrote:
Am 08.08.2011 13:38, schrieb Soeren Malchow (MCon):
Do you see the loaded modules when not using the conf file,
somethinglike this ?
cdr_mysql.so MySQL CDR Backend 0
res_config_mysql.so MySQL
Am 08.08.2011 13:38, schrieb Soeren Malchow (MCon):
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Norbert Zawodsky
Sent: Sunday, August 07, 2011 8:49 PM
To: Asterisk Users Mailing List - Non-Commercial
Hello everybody,
I've been using asterisk 1.2 for quite a long time now, but I thought
it's time to try a newer version of asterisk.
So I downloaded 1.8.5, extracted the tar, ran configure, make, make
install ...
Everything looks fine (no obvious compile/link errors).
But as soon as I
Maybe just a typo ? Misplaced dots between all those 1's and 0's ...
Maybe we should call it version 12 instead of 1100 ;-)
Am 22.07.2011 21:50, schrieb Danny Nicholas:
I thought it was going to be 1.10.0
*From:*asterisk-users-boun...@lists.digium.com
Am 11.07.2011 19:11, schrieb Jerry Geis:
Is there a method to lock asterisk into memory
such that once its loaded it does not get paged out?
lock into memory: disable swapping. (But this might have other
impacts ;-) )
increase probability to be in memory: Add RAM and stop other applications.
Am 21.06.2011 18:28, schrieb Mark Deneen:
On Tue, Jun 21, 2011 at 4:12 AM, randulorand...@randulo.com wrote:
On Tue, Jun 21, 2011 at 5:47 AM, Alex Balashov
abalas...@evaristesys.com wrote:
I nominate this for most imaginative use of Asterisk-users of 2011.
It's already qualified to win in
Am 28.10.2010 09:41, schrieb Per Jessen:
Over the last two weeks, we have had at least two incidents where our
asterisk server got flooded (a hundred or more per second) by SIP
packets. Once from 114.31.50.10, second time from 173.212.200.146. We
became aware of the problem when bandwidth
Am 28.10.2010 12:14, schrieb Per Jessen:
Ishfaq Malik wrote:
On Thu, 2010-10-28 at 09:41 +0200, Per Jessen wrote:
Over the last two weeks, we have had at least two incidents where
our asterisk server got flooded (a hundred or more per second) by SIP
packets. Once from 114.31.50.10, second
Am 27.07.2010 08:42, schrieb Motiejus Jakštys:
If all you need is block the SIP traffic from external sources, you
may do the following:
# iptables -A INPUT -s 192.168.1.0/24 -p udp --dport 5060 -j ACCEPT
# iptables -A INPUT -p udp --dport 5060 -j DROP
# iptables-save
Hello again!
after it being relatively quiet her for the last weeks, my Astrerisk
server was the target of 3 of that nasty REGISTER attacks during the
last days. While I can see not much danger coming from these attacks (I
use very long, complicated random generated passwords), they are still
Am 13.04.2010 10:47, schrieb Gordon Henderson:
I'd strongly disagree with this. (And I was the OP of this thread and had
my home/office network connection taken down due to it)
But then, I'm an old worldy Unix sysadmin and the philosophy of having a
program do one thing well is still etched
Hello to everyone!
Same here (Vienna, Austria).
I had this attack yesterday 6am (local time) from IP 216.105.128.63
whois 216.105.128.63 returns:
OrgName:Globalvision
OrgID: ACSIN-3
Address:78 Global Drive
Address:Suite 101
City: Greenville
StateProv: SC
PostalCode:
Am 11.04.2010 17:05, schrieb Mark Smith:
Same this end from 184.73.17.150.
Use this little piece of iptables magic to block the whole of Amazon's EC2 ip-
range.
iptables -F
iptables -A INPUT -m iprange --src-range 216.182.224.0-216.182.239.255 -j DROP
iptables -A INPUT -m iprange
Hi Philipp!
Philipp Kempgen schrieb:
Where exactly did you register your DNS server? Did your registrar
handle it for you? http://www.nic.at ? http://www.enum.at ?
Yes, my registrar http://www.my-enum.at handles it. (my-enum.at seems to
be a sub-company of nic.at)
First you have to register
Leif Neland schrieb:
Norbert Zawodsky skrev:
Sorry N. !
But - at least here in Austria - it is definitely *no* assumption that
my number with some extra digits can not be issued to someone else.
You probably have too many no/nots :-)
No! (Again, another no ;-) )
I meant
SIP schrieb:
By the time telephone operators began to be replaced by mechanical
switches, open numbering plans became impossible to design for. Once
Sorry, but there was definitely a time between retired human operators
and software switches.
And during that period, our open numbering plan
SIP schrieb:
ENUM is, quite literally, E164 Number Mapping (that's what it stands
for). If you're mapping numbers which are invalid E164 numbers (i.e. in
your scenario in which you're taking an E164 number and attaching digits
to it), you're violating the ENUM idea for the sake of
Leif Neland schrieb:
But if a pstn or cell call +431123456720 will it be connected to
+4311234567 ? Or will the call fail?
If so, +431123456720 is an invalid number.
Leif
That depends on the Dialplan coding.
A non-sip call comes in from the VoIP provider into the associated
context. The
SIP schrieb:
Yes... you would have to register (and possibly pay for, dependent on
the ENUM registrar) each individual number. The idea behind ENUM is that
it's an E164 number that is already yours that maps to whatever you want
it to map to (email, SIP, etc). The key point here is that you
But then you create phonenumbers in enum, which doesn't exist as
pstn-numbers.
Not the idea behind enum.
On the other hand, if you owned 10 or 100 pstn-numbers in series, you
could get the last one or two digits delegated to your dns-server.
Leif
Hello all you Gurus out there!
Please could you explain something to me:
Currently I try to get ENUMLOOKUP() working. Naturally I do all the
testing with my own number.
I registered my number at e164.org
I paid for registration of my number at a registration agent for e164.arpa
(I know, I don't
Leif Neland schrieb:
- Original Message -
*From:* Norbert Zawodsky mailto:norb...@zawodsky.at
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
*Sent:* Monday, November 23, 2009 3:15 PM
*Subject
Hi everybody,
i've been googling for quite some time now but can't find an answer to
my problem...
I'm using Asterisk 1.2.12.1 with mysql as the cdr backend.
In the dialplan i've written
exten = 1234,n,Set(CDR(userfield)=blah)
exten = 1234,n,Answer()
exten = 1234,n,Queue(.)
exten =
Philipp Kempgen schrieb:
Norbert Zawodsky schrieb:
I'm using Asterisk 1.2.12.1 with mysql as the cdr backend.
In the dialplan i've written
exten = 1234,n,Set(CDR(userfield)=blah)
exten = 1234,n,Answer()
exten = 1234,n,Queue(.)
exten = 1234,n,Hangup()
When I'm doing a call I can
The mortality rate on power supplies, diplays and the number or
broken
receiver hook swicthes on the lot of Snom 360's i bought 3 years
ago is
outright embarrassing.
That's odd. We've had Snom 190s, 320s, and 360s running day in day out
for years with not a single issue.
Hello everybody!
Please let me ask you a question:
Is it possible (and if yes, how) to configure 2 asterisk servers on two
machines so that the second one acts as a backup system if the first one
is unresponsive?
Clearly, the second should take over automacigally (but not necessarily
during an
Hi everybody!
I've setup my dialplan so that if an extension dials *21*, that
extension is added/removed as a queue member to a queue. (State toggled).
But it would be great to get an optical feedback of that phone's state
regarding the queue membership.
Does someone know if it is possible to
1.2.12.1
Lacy Moore - Aspendora wrote:
On 2/22/07, Norbert Zawodsky [EMAIL PROTECTED] wrote:
Does someone know if it is possible to light up a LED under this
szenario?
1.2 or 1.4?
___
--Bandwidth and Colocation provided by Easynews.com
Sune Kloppenborg Jeppesen wrote:
On Thursday 22 February 2007 22:24, Norbert Zawodsky wrote:
Hi everybody!
I've setup my dialplan so that if an extension dials *21*, that
extension is added/removed as a queue member to a queue. (State toggled).
But it would be great to get an optical
Hello everybody,
I have a analog Fax connected to a Linksys PAP2 Adapter but got some
problems with the Hook-Flash timing.
I played around with the Hook Flash Timer min and Hook Flash Timer
max settings at the PAP2 regional section, but the best I can get is:
If I call my fax (for example from
Hello everybody,
currently I'm implementing redirection into my dialplan.
What I want to do is: If a call comes in to my extension I want to dial
back out to my cell phone.
So far it works very well, but I've got a problem with the displayed
number on the cell phone.
What I want is that the
Hi Carla,
Carla Schroder wrote:
Debian is my fave, but for Asterisk I use CentOS. It's a free-of-cost clone
of
Red Hat Enterprise Linux, so it's very stable and reliable, and Asterisk runs
great on it. Debian is good too. They have Asterisk packages, but they're
generally a little bit
RR wrote:
Norbert, mate, I don't know why you're having so much problems. Do you
wanna post your extconfig.conf here? just to humour us? I have it
running with MSSQLServer a more complicated prospect than mySQL which
has a dedicated driver for it, and it still works.
RR, mate, I don't think
Derek Whitten wrote:
Norbert Zawodsky wrote:
RR wrote:
snip
Mate, I can't say it with authority but I'm almost certain that the
only DB that a specific driver was written for is MySQL. I think if
you use res_mysql.o you should be able to talk to mySql directly
without needing ODBC
Noah Miller wrote:
Hi Peder -
Is the storage of actual voicemail messages in a database still limited
to ODBC? If so, why?
Yes. Why? Nobody has developed a voicemail solution that directly
connects to a *SQL database for message storage.
A clear answer :-) Although a sad one :-(
Because
Hi Noah,
Noah Miller wrote:
Hi Norbert -
Just a thought: You could go the other way - share a volume on a
separate webserver, and have the asterisk box connect to the webserver
via NFS as a client, and store the voicemail on the NFS share. While
I don't have any exact numbers, it seems
RR wrote:
snip
Mate, I can't say it with authority but I'm almost certain that the
only DB that a specific driver was written for is MySQL. I think if
you use res_mysql.o you should be able to talk to mySql directly
without needing ODBC.
/snip
O.k., Nice to hear. But I'm not sure *how* to
Hi Peder,
I asked the same question some time ago.
Never got any answer... :-(
Norbert
Peder @ NetworkOblivion schrieb:
Is the storage of actual voicemail messages in a database still limited
to ODBC? If so, why?
And is the use of mySQL and ODBC at the same time still a bad idea? If
Hi everybody,
just to confirm that I understood it right (and that the info isn't
obsolete):
I have to store the voicemail audio data in an external mysql DB. In
http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage I
read that this is only possible via ODBC and *NOT* via native
Hi Brian,
many thanks to you for your answers in the past! The always gave me the
little bit of mising information...
My Asterisk box is running fine now so I want to try the next step...
And now to all of you
What I want to implement is to use 1 button of my snom-360 phone for
following
Hi everybody,
I've got a very strange problem:
As far as I remember I didn't change anything on my Asterisk side. I
have 2 SIP providers to which I can place outbound calls.
Today I noticed that outbound calls to provider inode fail (and
inbound from this provider too). On the CLI I get every
Hi Brian, hi list,
Brian Candler wrote:
On Fri, Oct 13, 2006 at 01:35:04AM +0200, Norbert Zawodsky wrote:
I've set canreinvite=no on the channel to the SIP provider and it
immediately worked. O.k., I'm happy about that but I want to
*understand* what's going on here.
.
My setup
Hello everybody,
I have a problem and already browsed the mailing list archives but
didn't find any help. So I ask here
My new * Box ist up runnig. Got access to the SIP server of my
Internet provider (Userid, password, phone number, ...). And yesterday I
tried my first calls to the outside
Brian Candler wrote:
On Thu, Oct 12, 2006 at 02:26:16PM +0200, Norbert Zawodsky wrote:
As soon as the connection is up and the receiver is lifted on both
sides, the leds of the DSL Modem between Asterisk and my ISP, and the
leds of the switch between Asterisk and the SNOM phone start
Hi again,
as I wrote before, I'm new to Asterisk. And so, many many new questions
pop up .
For example:
I have here a very small telephony system. We have only 5 (or so)
extensions. (4 phones, 1 fax).
So I wonder if there is disadvantage if we use only 1 digit extensions
(1 for boss, 2 for
And Hi again,
I wrote:
I think I can remember something from the Asterisk-TFOT book saying
that one must not use 1-digit extension numbers. But I can't remember
that very well and can't find it in the book any more
I found it! On page 90 the book says:
quote
... (Well, almost.
Michel Vaillancourt wrote:
Norbert Zawodsky wrote:
Hi everybody!
I have some Linux experience but I'm completely new to asterisk.
I bought a small VoIP-PBX which has Linux (Kernel 2.6.13) Asterisk
(1.2.12) preinstalled and some basic configuration (Wiht a few
extensions). Now I want
Hi everybody!
I have some Linux experience but I'm completely new to asterisk.
I bought a small VoIP-PBX which has Linux (Kernel 2.6.13) Asterisk
(1.2.12) preinstalled and some basic configuration (Wiht a few
extensions). Now I want to implement something more, fox example
voicemail (storing
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