Hey, I'm going thru logs, and I see some very common and interesting things
that the hackers are looking for.
In a whole bunch of scans, I've noticed that the first guess or two for sip
accounts
is usually a 10-digit number. I'm asking myself, why these numbers? Are they
looking
for a voip trunk?
Here's some examples:
2648061411
3190339404
I'm getting exactly the same. Odds of getting a working number, are like the
odds of winning the lottery.
My guess is they are either trying to find a voip trunk, or they are trying to
make cold calls to the extensions on my system. Sales or
My guess is they are looking for 10 digit phone numbers as extensions.
Are they all from 1 IP address or from many? If from many, they are likely
many serial scan or from a list of suspected VOIP numbers. If from one, and
that random, then from a list of suspected VOIP numbers.
Since
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Sunday, November 07, 2010 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Why are the hackers scanning for
asterisk 1.4.35
dahdi 2.3.0.1+2.3.0
one span on a 4port T1 card
Got complaints this morning that outbound and inbound calls were
scratchy and I made a few test calls. It kind of sounds like the gain
is too high somewhere, and the audio is overdriven. Is this a problem at
the carrier? I'm
On Sun, Nov 7, 2010 at 10:00 AM, Cary Fitch ca...@usawide.net wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Sunday, November 07, 2010 8:33 AM
To: Asterisk Users Mailing
On Sun, Nov 07, 2010 at 07:11:43AM -0700, Steve Murphy wrote:
Hey, I'm going thru logs, and I see some very common and interesting things
that the hackers are looking for.
In a whole bunch of scans, I've noticed that the first guess or two for sip
accounts
is usually a 10-digit number. I'm
I've just switched my outbound ip address a week ago. Not static, but
dhcp on TimeWarner cable. I've registered only with another of our
offices. The outbound calls are all pstn bound through Teliax.
But somehow my log is filling up with registration requests over this
new ip address from a
On 11/7/10 9:26 AM, Jeff LaCoursiere wrote:
asterisk 1.4.35
dahdi 2.3.0.1+2.3.0
one span on a 4port T1 card
Got complaints this morning that outbound and inbound calls were
scratchy and I made a few test calls. It kind of sounds like the gain
is too high somewhere, and the audio is
Adding on more thoughts:
Think what Google has done in Mapping the Earth, Mapping the Web, and now
working on Google Voice and Google Mail.
Every one of those makes money either directly and/or synergistically with
other components.
Now consider someone with telephone interests or spam
Hi Everyone,
Knowing that running Asterisk on an embedded board like the Alix2d3 requires
some fine tuning. Do you know of any good guides out there that does this
from beginning to end? Looking to run this in a small office environment.
Thanks
--
Check out the AstLinux site
There is a version there for the Alix boards, though I am not impressed
with Alix. IMO overpriced.
John Novack
Bruce B wrote:
Hi Everyone,
Knowing that running Asterisk on an embedded board like the Alix2d3
requires some fine tuning. Do you know of any good
On Sun, Nov 7, 2010 at 11:23 AM, Bruce B bruceb...@gmail.com wrote:
Knowing that running Asterisk on an embedded board like the Alix2d3 requires
some fine tuning. Do you know of any good guides out there that does this
from beginning to end? Looking to run this in a small office environment.
John,
AstLinux seems promising. Have you used this flavor in
production environment?
Paul,
So, don't use the Yum repositoy?!
And, are you sure that is the only thing needs to be done. I am thinking
there is more tweaking need to be done. I am not looking to just install
Asterisk but it should be
On Sun, Nov 7, 2010 at 1:01 PM, Bruce B bruceb...@gmail.com wrote:
So, don't use the Yum repositoy?!
Usually not. If you don't want to get your hand dirty managing the OS
layer, try Askozia[1]. Most embedded solutions will use a modified
Busybox installation, allowing for lightweight binaries.
I don't want to start the How many calls can Asterisk handle? discussion
or How many angels can stand on the point of a pin? discussion either.
But can anyone contribute some practical knowledge of systems that take in
channel bank T1s or DS3s from far away, and process the calls?
I am looking
On Sun, Nov 7, 2010 at 11:03 AM, Cary Fitch ca...@usawide.net wrote:
Adding on more thoughts:
Think what Google has done in Mapping the Earth, Mapping the Web, and now
working on Google Voice and Google Mail.
Every one of those makes money either directly and/or synergistically with
other
On 07/11/2010 19:29, Cary Fitch wrote:
I don't want to start the How many calls can Asterisk handle? discussion
or How many angels can stand on the point of a pin? discussion either.
But can anyone contribute some practical knowledge of systems that take in
channel bank T1s or DS3s from far
inline
I don't want to start the How many calls can Asterisk handle? discussion
or How many angels can stand on the point of a pin? discussion either.
You just did
But can anyone contribute some practical knowledge of systems that take in
channel bank T1s or DS3s from far away, and process
Yes, Astlinux and Askozia are the leading candidates for use on such
small platforms. I have used Astlinux on Soekris boards which are
similar. I wrote it up here:
http://www.mgraves.org/?p=1092
That was some time ago but the basics of it are still sound.
Here's some further thoughts on small
I believe this looks like a standard channel bank. Asterisk generates all
audio. Ring and hook status are sent out of band. Dial tones are in-band.
Ringback, busy, congestion are in-band audio. I would think a standard T1 card
would be fine.
That said, I would verify this with the LEC.
On 11/06/2010 09:18 PM, Sherwood McGowan wrote:
On Sat, Nov 6, 2010 at 2:45 PM, Jonas Kellensjonas.kell...@telenet.be
wrote:
On 11/06/2010 07:18 PM, Tilghman Lesher wrote:
On Saturday 06 November 2010 11:22:06 Jonas Kellens wrote:
Hello,
I just experienced a
I have installed Asterisk before w/ no issues but while trying today
(1.6.2.13 and centors 5.4) I receive the following at the CLI:
The configure script must be executed before running 'make'.
Please run ./configure.
Any tricks on getting through this?
I did not select to
On 7 Nov 2010, at 20:59, Thomas Perron wrote:
I have installed Asterisk before w/ no issues but while trying today
(1.6.2.13 and centors 5.4) I receive the following at the CLI:
The configure script must be executed before running 'make'.
Please run ./configure.
Any
There is a network of telephone switch collectors, worldwide that uses
Asterisk to interface the network with their switches, as well as
members who have an interest in these old switches but don't yet have
one working.
I have personally set up about 20 nodes with AstLinux on HP thin
clients,
When all else fails, do what the program tells you to do!
The requirement to run ./configure has been around since sometime in 1.4
And CentOS 5.5 is current. You might wan to update it first?
John Novack
Thomas Perron wrote:
I have installed Asterisk before w/ no issues but while trying today
On Sun, Nov 7, 2010 at 3:58 PM, Jonas Kellens jonas.kell...@telenet.be wrote:
On 11/06/2010 09:18 PM, Sherwood McGowan wrote:
On Sat, Nov 6, 2010 at 2:45 PM, Jonas Kellensjonas.kell...@telenet.be
wrote:
On 11/06/2010 07:18 PM, Tilghman Lesher wrote:
On Saturday 06 November 2010 11:22:06
On Sun, Nov 7, 2010 at 1:29 PM, Cary Fitch ca...@usawide.net wrote:
But can anyone contribute some practical knowledge of systems that take in
channel bank T1s or DS3s from far away, and process the calls?
Yes. Adtran makes excellent gear. The MX 2800 is good for breaking a
channelized DS3 into
Hello,
I am running Asterisk 1.6 with Realtime enabled for my SIP users and peers. The
backend is a MySQL database running through the ODBC backend in Asterisk. At
this point everything works in terms of phones registering, placing calls
between them, etc. However, I am having a problem
Yes. Adtran makes excellent gear. The MX 2800 is good for breaking a
channelized DS3 into PRIs.
Thanks, will look at that. Ah, a DS3/T1 mux. I was looking for a DS3
PC Card... it would have 672 channels but the system doesn't need to
handle but 20% of them at one time.
If you're just
Has anyone used HUDlite recently and got it operating with Open Source
Asterisk 1.6 or 1.8? I have read the instructions on HUDLite but it appears
that it is only suited to Fonality versions like Trixbox. I would like to
test HUDLite as a presence panel. If there are other options we are open to
Forget about HUDlite you want iSymphony, http://www.getisymphony.com/
On Sun, Nov 7, 2010 at 5:02 PM, Rupert Utteridge rupe...@dtasia.com.auwrote:
Has anyone used HUDlite recently and got it operating with Open Source
Asterisk 1.6 or 1.8? I have read the instructions on HUDLite but it appears
Hello All,
i have one simple Question regarding integration of asterisk into sugar crm
whether using trixbox or normal asterisk,
can anyone have any link , forum or tutorial where i can find some
information and some starting point .
any help appreciated
regards
Dhaval
--
hey,
I want to ask whether VAD is the asterisk functionality or softphones's
functionality. Because I am using speex and zoiper but configuring VAD=true in
codecs.conf does not suppress silence ..
Thank in advance for help :)
--
Hello list members,
We're trying to get MWI notifications on our ATA device and we set it to
send SUBSCRIBE messages to Asterisk, but it gets UNAUTHORIZED messages,
despite the fact that we set the following lines in its settings in
sip.conf:
subscribemwi=yes
mailbox...@from-extensions
We
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