Very strange...
I ran the Asterisk CLI for other tasks, and suddenly I got this message:
== Using SIP RTP CoS mark 5
-- Executing [000972592603325@default:1] Verbose(SIP/192.168.
20.120-002a, 2,PROXY Call from 0123456 to 000972592603325) innew
stack
== PROXY Call from 0123456
Hi list!
Very strange...
I ran the Asterisk CLI for other tasks, and suddenly I got this message:
== Using SIP RTP CoS mark 5
-- Executing [000972592603325@default:1]
Verbose(SIP/192.168.20.120-002a, 2,PROXY Call from 0123456 to
000972592603325) in new stack
== PROXY Call from
On Mon, 8 Jun 2015 13:19:53 -0700 (PDT)
Steve Edwards asterisk@sedwards.com wrote:
Look for address blocks (class A, B, C) that are allocated to
geographic regions you do not have any providers. If you limit your
'attack surface' you make your security problem manageable.
Get this file:
On Mon, 8 Jun 2015 22:24:33 +0200
Luca Bertoncello lucab...@lucabert.de wrote:
Kevin Larsen kevin.lar...@pioneerballoon.com schrieb:
Basically, they are hoping that you are running the equivalent of a
mail server open relay. They are trying to use you to dial out to
another number. You
On Mon, 8 Jun 2015, Kevin Larsen wrote:
Better to fail and fix than to permit and pay for it later.
That would make a great T-shirt:
Deny and Fix
vs
Permit and Pay
--
Thanks in advance,
-
Kevin Larsen kevin.lar...@pioneerballoon.com schrieb:
Based on SIP packets coming in from IP addresses you don't recognize,
while you may not be hacked, you would seem to have people probing your
I think, too, it's someone probing my IP...
system. One thing you can do at the firewall level
On Mon, 8 Jun 2015, Luca Bertoncello wrote:
This is not really possible, since I'll login on my Asterisk from many
Providers...
many all
So make a list of the 100 or so providers you have active accounts with.
It's still way less than 'all.'
Also, I'm willing to bet you won't be
OK, I set alwaysauthreject = yes and I discovered a allowguest, which I
set
to no, too.
The PBX is behind a Firewall and I just allow UDP 5060 and 1-10100.
Now I log the SIP-pakets coming from Internet, too...
Hopefully I solved my problem...
Make sure you have solved the problem. You
I'm guessing this is a small/home system? I suggest you install SecAst from
this site: www.telium.ca It's free for small office / home office and will
deal with these types of attacks and more. It can also block users based on
their Geographic location (based on the phone number it
The Asterisk Development Team has announced the releases of:
DAHDI-Linux-v2.10.2-rc1
DAHDI-Tools-v2.10.2-rc1
dahdi-linux-complete-2.10.2-rc1+2.10.2-rc1
This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
On Mon, 8 Jun 2015, Michelle Dupuis wrote:
You're definitely under attack (based on the 0123456 ID) so be sure to
take preventative steps to avoid a $50k phone bill..
Don't enable 'auto-replenish' in your provider account and don't keep a
balance you can't afford to lose.
--
Thanks in
Hi Asterisk-user.
I'm starting in a soft-phone project with lots of requirements and some of then
caused me some doubts about Asterisk. Could someone tell me if Asterisk can
help me with some requirements? See below:
1 - My SIP server (Asterisk) will have some SIP clients registered in its
Make sure you have solved the problem. You don't want to get hit with
a
phone bill for calls from your location to Israel. Basically, they are
hoping that you are running the equivalent of a mail server open
relay.
They are trying to use you to dial out to another number. You don't
Kevin Larsen kevin.lar...@pioneerballoon.com schrieb:
Make sure you have solved the problem. You don't want to get hit with a
phone bill for calls from your location to Israel. Basically, they are
hoping that you are running the equivalent of a mail server open relay.
They are trying to
As a practice, by default all the extensions you expose on the allowguest
mode should lead inbound to your asterisk and should never pick any
outbound trunk and dial out.
Your best option is to remove all outbound extensions from the default
context, move them to default2 and set default
Hi,
Sorry if off topic, but is anyone here on this list using it?
I am currently searching for a good router for my home network wich supports
SIP.
Many thanks!
--
_
-- Bandwidth and Colocation Provided by
Hi again, list!
I know, I'm really annoying the list... :)
Well, maybe I got my Asterisk at home (wrt on the previous E-Mails)
to accept my mobile phone from Internet.
It was a problem with the network and the firewall.
Now I can log my mobile phone in my Asterisk in and the phone is
Zitat von A J Stiles asterisk_l...@earthshod.co.uk:
If I call a phone at home using my cellphone it works and the quality
is perfect!
If a phone at home call my cellphone, however, the quality on my
cellphone is very poor, but on the other phone is perfect...
I think, it is something by the
On Monday 08 Jun 2015, Luca Bertoncello wrote:
Hi again, list!
I know, I'm really annoying the list... :)
Everyone has to start somewhere; and at least you aren't asking hundreds of
questions in one go, including some which come under the heading of Don't
even think about trying to set this
Hello!
I've got a little problem with Asterisk (11.14.1), the voicemessages are
kinda limited to 40 seconds (average) aproximately; because when a message
reach this long I got a cut in the file (*.wav) after I got this message:
WARNING[15035][C-21ef]: format_wav_gsm.c:418 wav_read: Short
Hi,
I have configured a certified asterisk 13 server with chan_mobile and
res_pjsip. I have a Cisco 7940 hardphone and I use ekiga as softphone
client.
Now the problem is, using the hardphone I'm able to call the softphone
and hear everything properly. But when I call from the hardphone to some
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