2010/10/27 Benoit maver...@maverick.eu.org:
On 27/10/2010 12:59, Krzysztof Urbaniak wrote:
Hi!
We've experienced asterisk has gone without any message, it wasn't any
segfault, anything in asterisk messages log that says about shutting
down.
How do you launch asterisk ? did you try without
Hello,
Is there any reason why an IP-phone would pounder on port 5060 ? My
firewall blocks the public IP because it thinks the remote IP is port
scanning on port 5060.
I think the phone is just registering but for some reason it does this
repeatedly in a very short time.
Oct 28 09:01:48
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 started suffering because
asterisk got very
Le 26/10/2010 14:49, Shaun Ruffell a écrit :
On 10/26/2010 06:38 AM, Administrator TOOTAI wrote:
I installed 2 HB8 cards each of them with a Quad Bri modules in a HP 360
G6 running Debian Squeeze. Here is an output of dmesg wafter server has
booted:
[...]
before asking RMA for
I assume that you checked and the remote IP is a legitimate IP phone? If not,
it could be an attempt to break into your system.
If it is a legitimate IP phone, make sure that the SIP configuration is correct
- if the SIP authentication fails, you can see this happening.
From:
Paul,
Thanks, I'll try this patch later tonight.
Nic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: 28 October 2010 03:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Wed, Oct 27, 2010 at 05:37:15PM -0500, Don Kelly wrote:
I have recently updated from Centos/*1.2 to Ubuntu Server and FreePBX
2.8.0.2.
We have an Astribank with 4 T1 ports and 16 FXS ports. After updating, we
had it working for a while with one NT PRI and one TE PRI and, in the
On 10/28/2010 10:44 AM, Kevin Keane wrote:
I assume that you checked and the remote IP is a legitimate IP phone?
If not, it could be an attempt to break into your system.
If it is a legitimate IP phone, make sure that the SIP configuration
is correct -- if the SIP authentication fails, you
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 time from 173.212.200.146. We
became aware of the problem when
Le 28/10/2010 08:41, Krzysztof Urbaniak a écrit :
2010/10/27 Benoitmaver...@maverick.eu.org:
On 27/10/2010 12:59, Krzysztof Urbaniak wrote:
Hi!
We've experienced asterisk has gone without any message, it wasn't any
segfault, anything in asterisk messages log that says about shutting
down.
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
Hi,
On 10/28/2010 01:06 AM, GBR Icasiano, Ryan A. wrote:
Hi,
Thanks for your reply. I'm calling a normal phone using the DIAL cmd. Here is
my sample dial command:
exten =s,4,Dial(SIP/xxx${extensi...@media_gateway,10,t)
but when I use:
exten =s,5,NoOp(SIP/xxx${extensi...@media_gateway
Norbert Zawodsky wrote:
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
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 time from 173.212.200.146.
We became
Hi,
I can actually place a successful call using that configuration. The telco i'm
currently working requires the prefix.
What I'm trying to do is to capture the status of the mobile phone, if it is
currently engaged in a call or not. I achieved this successfully by emulating
it via a
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
On Thu, 28 Oct 2010, Jonas Kellens wrote:
On 10/28/2010 10:44 AM, Kevin Keane wrote:
I assume that you checked and the remote IP is a legitimate IP phone? If
not, it could be an attempt to break into your system.
If it is a legitimate IP phone, make sure that the SIP configuration is
On Thu, 28 Oct 2010, Norbert Zawodsky wrote:
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
Norbert Zawodsky wrote:
Per,
(didn't want to be unfriendly to you !)
Not at all.
As you say, you don't like anything to modify your firewal. My
words!
Someone (don't remember who when) on this list showed me a very
clever trick (=iptables rule) to drop the packets if too many
On 10/28/2010 12:52 PM, Gordon Henderson wrote:
On Thu, 28 Oct 2010, Jonas Kellens wrote
On 10/28/2010 10:44 AM, Kevin Keane wrote:
I assume that you checked and the remote IP is a legitimate IP phone? If
not, it could be an attempt to break into your system.
If it is a legitimate IP
Gordon Henderson wrote:
On Thu, 28 Oct 2010, Norbert Zawodsky wrote:
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
I keep the AGI in Git as a version control system. But, you can view the AGI
source here:
http://messinet.com/trac/browser/gv/gv.agi
And at the very bottom of that page is a link to download it as an individual
file here:
On Thu, Oct 28, 2010 at 10:11 AM, Stephen Reese rsre...@gmail.com wrote:
Is there a way to prevent Google Chat from staying logged in but still
be able to dial outbound? People think I'm logged in persistently and
send me messages that I miss. Even if I set a status message in
asterisk most
Friday we'll be hearing about SIP Communicator Java VoIP and Instant
Messaging client.
SIP Communicator is an audio/video Internet phone and instant
messenger that supports some of the most popular VoIP and instant
messaging protocols such as SIP, Jabber, AIM/ICQ, MSN, Yahoo!
Messenger, Bonjour,
I have a very simple setup with two SIP routes to my carrier. I need to have
every other phone call placed to that carrier go to a different address.
This is what I need the call flow to look like. I have spent many hours
searching and have not found a working example.
Call1 exten =
- Original Message -
I have a very simple setup with two SIP routes to my carrier. I need to have
every other phone call placed to that carrier go to a different address.
This is what I need the call flow to look like. I have spent many hours
searching and have not found a working
On Thu, Oct 28, 2010 at 11:08:12AM -0400, Tim King wrote:
I have a very simple setup with two SIP routes to my carrier. I need to have
every other phone call placed to that carrier go to a different address.
I think what you need to do here is check/set a variable in the astdb.
(If the variable
Thanks to the hard work of many people in the Adhearsion community, I am
pleased to be able to announce the immediate availability of Adhearsion version
1.0. Since Jay Phillips first began work on the project in 2006 Adhearsion has
changed the way developers think about telephony applications.
Fail2Ban
Regards
- Original Message -
From: Per Jessen p...@computer.org
To: asterisk-users@lists.digium.com
Sent: Thursday, October 28, 2010 2:41 AM
Subject: [asterisk-users] being bombarded with SIP packets
Over the last two weeks, we have had at least two incidents where our
Two incidents in two weeks is not bad. I get 2-4 a day. There must be many
here with even more than that. You should start considering some safety
practices like disabling long distance and international calls by default,
put a cap on long distance and international calls even for genuine users,
Sorry for the confusion, but the last sentence throws me off. Translation
of this to dialplan logic is left as an exercise for the
student. Is this example from some sort of book or is this a way of saying
I am left to figure the rest out??
I was hoping to find a simple example of how this works.
Hi,
are you installed unixodbc-dev?
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Thanks For The replies. I have tried piecing the samples together. Just for
testing purposes i have created the following.
[test]
exten =
_X.,1,GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:route2)
exten = _X.,n(route1),Set(DB(avoics/route)=1)
exten = _X.,n,SayNumber(1)
exten =
It seems that the
GotoIf(${SET(DB(avoics/route)=$[!0${DB(avoics/route)}])}?route1:route2) is
always returning false as if the SET command is not returning a value nor is
it changing the value in the DB.
Will this not work because I am running Asterisk 1.4.25.1??
On Thu, Oct 28, 2010 at 3:15 PM,
On 10/28/2010 3:41 AM, Per Jessen wrote:
2) if you've got some iptables rules for limiting inbound SIP by rate?
exactly what i was going through; here's how i reacted (throttles both
SSH and SIP Register:
First, I completely blocked all non-North American Amazon EC2 networks
- I won't be
Hi,
On 10/28/2010 11:20 AM, GBR Icasiano, Ryan A. wrote:
Hi,
I can actually place a successful call using that configuration. The telco
i'm currently working requires the prefix.
What I'm trying to do is to capture the status of the mobile phone, if it is
currently engaged in a call or
Consider this RESOLVED thanks to the help of [David
Vossel](http://www.davidvossel.com/?p=162) (*HIGH FIVE*) and the new
wiki entry from [Malcolm
Davenport](https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google).
The trick was the following in extensions.conf:
exten = s,1,Answer()
exten
On Thu, Oct 28, 2010 at 7:30 PM, Vinh Nguyen vinhdi...@gmail.com wrote:
Consider this RESOLVED thanks to the help of [David
Vossel](http://www.davidvossel.com/?p=162) (*HIGH FIVE*) and the new
wiki entry from [Malcolm
Davenport](https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google).
Hi
I have asterisk 1.4
I want to make a MGCP trunk as a client to connect to a provider who is
using MGCP protocol, he provided me with user password,
I tried a custom trunk:
MGCP/$outn...@user:passw...@66.152.163.106:4000
Not seems to help,
Any suggestions plz?
--
On Fri, Oct 29, 2010 at 4:21 AM, Baha @ SH i...@saudihome.com wrote:
Hi
I have asterisk 1.4
I want to make a MGCP trunk as a client to connect to a provider who is
using MGCP protocol, he provided me with user password,
I tried a custom trunk:
On Thu, Oct 28, 2010 at 9:54 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Fri, Oct 29, 2010 at 4:21 AM, Baha @ SH i...@saudihome.com wrote:
Hi
I have asterisk 1.4
I want to make a MGCP trunk as a client to connect to a provider who is
using MGCP protocol, he provided me with
Hi,
Thanks for your very informative response. This is really helpful. I wouldn't
be pushing it though since it isn't possible as of now.
Kudos!
RYAN ICASIANO
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On
On 10/28/2010 11:18 PM, GBR Icasiano, Ryan A. wrote:
Here is what I do today and it works fine:
- asterisk/trixbox
- Dext/android phone
- Bell Canada cell provider
- call comes in, to an extension with voicemail
- rings a bunch of sip devices (real phones, and the android via
linphone if it
On Mon, Oct 25, 2010 at 11:02 AM, Danny Nicholas da...@debsinc.com wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *ALAEDDINE abbech
*Sent:* Monday, October 25, 2010 10:52 AM
*To:*
On Mon, Oct 25, 2010 at 11:11 AM, Steve Edwards
asterisk@sedwards.com wrote:
Un-self-top-posting...
--- En date de : Jeu 21.10.10, ALAEDDINE abbech alasup...@yahoo.fr a
écrit :
De: ALAEDDINE abbech alasup...@yahoo.fr
Objet: thousands Hangup per second /saturation of bandwidth
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