On Sun, 31 Oct 2010, Matt Darnell wrote:
We have downloaded some royalty free music but it sounds 'fuzzy' when
we
test it with the system.
On Sun, Oct 31, 2010 at 5:34 PM, Steve Edwards asterisk@sedwards.com
wrote:
Can you post a link to the original?
On Sun,
On Sun, 2010-10-31 at 11:39 -0600, Joel Maslak wrote:
To guess an 8 character (which is short) password that consists of random
upper case, lower case, numbers, and 10 symbols (there are more you can use
if you want), the average number of passwords that you would have to try to
get in is:
Hello,
DNS SRV is not really working: when I stop the Asterisk proces on my
production server, then the SIP phone is not registering to my backup
SIP server.
I have the following DNS records :
sip2.domain.tld.AYY.YY.YY.YY
sip.domain.tld.AXX.XX.XX.XX
On Wed, Oct 27, 2010 at 3:43 PM, Jose P. Espinal j...@slackware-es.comwrote:
Hello List,
A few days ago I installed ViciDial on a server, and while looking to
the default 'extensions.conf' file, I saw this line:
exten = _010*010*010*015*.,1,Dial(${TRUNKTESTast}/${EXTEN:16},55,oT)
Can
Unsuccessful attempts are recorded, however SIP-s is not easily doable on
asteridk 1.4. I tried once without any success. Maybe somebody who has
successfully implemented it can write a little how-to on it.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-11-01 4:48 AM,
Hi All,
I would like to separate the media traffic from the signalling.
Can Asterisk send and receive media (rtp) traffic from a secondary network
interface?
Thanks,
Harel
This electronic message and any files transmitted with it are confidential and
intended
Hi All,
I would like to separate the media traffic from the signalling.
Can Asterisk send and receive media (rtp) traffic from a secondary network
interface?
Thanks,
Harel
This electronic message and any files transmitted with it are confidential and
intended
On 10/31/2010 11:26 AM, Joel Maslak wrote:
I suspect even munin would provide you such options. Not to mention any
more capable monitor.
I already have a monitor (tied into nagios, which pages me if my fraud
thresholds are exceeded), but I feel that is probably beyond
Hello again,
Here's the header as it appears in 1.6.2.11 CLI output:
INVITE sip:1...@192.168.10.169:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.10.3:5060;branch=z9hG4bK73713002;rport
Max-Forwards: 70
From: SIP ehf/Örn Arnarson sip:7712...@192.168.10.3;tag=as2813a8fe
To:
Only 100? We had a single server over 300.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria
Sent: Saturday, October 30, 2010 9:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I just wanted to add my voice to this attack. I saw the morning that I had
200+ distinct ips since the weekend. I used a small perl script that blocks
failed usernames and passwords at iptables level I found thei morning :
Its going on and on and on. Nothing like this has happened before. I have
several hundreds by now. Make me wish Internet was a more regulated place.
Its a place where bad people have the upper hand and good people cannot do
anything about it. I know incidences where spammers and attackers were
And obviously these attackers read our emails on lists like this and adjust
their sick strategies accordingly.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-11-01 12:02 PM, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
Only 100? We had a single
Please ignore this message (wrong subject by mistake). Please see message with
subject 2nd network interface for RTP/media
Thanks
Harel
--
Message: 2
Date: Mon, 1 Nov 2010 12:52:16 +0100
From: Harel Cohen ha...@easycall.gi
Subject: [asterisk-users] MoH and stuch
I was going to point out a failing of the attackers, but figured they read
the list and don't need any more tips.
Cary Fitch
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent: Monday, November 01,
Hi Cary,
Can you email me off the list to point it out?
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-11-01 1:37 PM, Cary Fitch ca...@usawide.net wrote:
I was going to point out a failing of the attackers, but figured they read
the list and don’t need any more
On Mon, 1 Nov 2010, Zeeshan Zakaria wrote:
Hi Cary,
Can you email me off the list to point it out?
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-11-01 1:37 PM, Cary Fitch ca...@usawide.net wrote:
I was going to point out a failing of the
LOL
On Mon, Nov 1, 2010 at 23:33, Cary Fitch ca...@usawide.net wrote:
I was going to point out a failing of the attackers, but figured they
read the list and don’t need any more tips.
Cary Fitch
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
On 11/01/2010 01:44 PM, Nyamul Hassan wrote:
I think the only real solution here is to make people take more
responsibility for their actions
- find and punish the actual abusers
- make users liable for damages caused by infected PC's - defaults from
an isp should be everything locked down
Too late, now switching to attack level: lethal :)
No, I am not one of these losers, and don't ever plan to be.
Zeeshan A Zakaria
--
www.ilovetovoip.com
www.pbxforall.com (beta)
On 2010-11-01 1:49 PM, Jeff LaCoursiere j...@sunfone.com wrote:
On Mon, 1 Nov 2010, Zeeshan Zakaria wrote:
Hi
Finding and punishing the abusers is the real problem, specially when in my
country (Canada) where we generally don't like punishing people (or they get
away finding loop holes in the law, or thanks to their lawyers), how would
we catch people in other parts of the world and punish them?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA256
Here's my take on the attack... Sigh...
http://www.stuartsheldon.org/blog/2010/11/sip-brute-force-attacks-escalate-over-halloween-weekend/
Stu
- --
Open up the window Let some air into this room I think I'm almost
chokin' From the smell of
Steve,
Did you use this syntax to convert:
sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql
-Matt
On Sun, Oct 31, 2010 at 9:43 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Sun, 31 Oct 2010, Matt Darnell wrote:
We have downloaded some royalty free music but
Chris Abel writes:
Hello everyone!
I've had this problem for a while and cant figure it out. When an outside
caller calls an extension on my asterisk system, they do not hear any
sort of ringing. Inside extensions calling other extensions do hear
ringing. We have 3 other asterisk systems that are
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA256
Here's my take on the attack... Sigh...
http://www.stuartsheldon.org/blog/2010/11/sip-brute-force-attacks-escalate-o
ver-halloween-weekend/
Stu
They were trolling for SIP account IDs, not really trying to register.
It was a coordinated bot or
I am trying to set up Hunt Groups and I am having some issues. Here is what
I am trying to do. All my users actually register with OpenSIPS. Asterisk
is using Realtime and I have set up a MySQL View Table so that Asterisk
see's all the SIP users info that OpenSIPS has. This is what I have
When I call into my Asterisk box via my VoIP line (using gsm codec) and then
try to make an outgoing DISA call over PSTN I get the following:
[Nov 1 15:12:54] WARNING[17694]: chan_dahdi.c:8930 dahdi_write: Cannot
handle frames in gsm format
[Nov 1 15:12:54] WARNING[17694]: app_dial.c:1401
Un-top-posting...
On Sun, 31 Oct 2010, Matt Darnell wrote:
We have downloaded some royalty free music but it sounds 'fuzzy' when we
test it with the system.
On Sun, Oct 31, 2010 at 5:34 PM, Steve Edwards asterisk@sedwards.com
wrote:
Can you post a link to the
Hey;
Anyone see this before:
[Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
6839, digest has 3169
G
--
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On Mon, 1 Nov 2010, Cary Fitch wrote:
Any small system should:
Use IPTABLES and block any parts of the world you don't need access
to/from. Start with any Class A address that is probing your system.
Make your SIP IDs 8-12 characters in length, and use at least alpha
numerical
On Mon, 1 Nov 2010, Silver Thorne wrote:
Anyone see this before:
[Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
6839, digest has 3169
You may have better luck with a more descriptive subject. Lots of users
have an issue or two with Asterisk.
Some details will also
On 1 November 2010 21:11, Silver Thorne zora...@gmail.com wrote:
Hey;
Anyone see this before:
[Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
6839, digest has 3169
G
`
Is it causing a problem for you?
--
On 1 November 2010 21:20, Steve Edwards asterisk@sedwards.com wrote:
On Mon, 1 Nov 2010, Cary Fitch wrote:
Any small system should:
Use IPTABLES and block any parts of the world you don't need access
to/from. Start with any Class A address that is probing your system.
Make your
I know there was talk on VUC recently about some kind of realtime RBL for
SIP. Has anything progressed?
It would be SO easy for asterisk users to contribute to a blacklist and also
do a lookup in realtime to see if an IP has been blacklisted.
A little bit of joined up thinking in the
Hi!
[Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
6839, digest has 3169
You most likely have two SIP UAs that use the same IP, of which the 6839
account is listed last in sip.conf while 3169 is trying to auth
(unsuccessfully).
Philipp
--
Be careful, telcos may make the users responsible if they have insecure
PBXes...right now they often write off much of the charges.
But I do agree that there would be a lot less garbage on the net if everyone
was liable for their insecurity. Heck, there would be no SIP attacks if
everyone's
On 11/01/2010 08:20 PM, Joel Maslak wrote:
Be careful, telcos may make the users responsible if they have
insecure PBXes...right now they often write off much of the charges.
you must have a great telco - around here even credits that are agreed
to seldom show up on the bill without investing
Yup, that's exactly what is happening. If there is only a way to override the
response(busy tone) by a ringing tone from asterisk, then the caller will not
hang up since after the busy status interpreted by asterisk as NOANSWER,
there will be a fallback which it will either transfer to another
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