LOL :) that really made me chuckle this morning; and very apt for the fact I
did not post any fundamental details about the issue. All points duly noted!
--
Thanks, Phil
- Original Message -
Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like,
one of those who rocket
Hi,
I am attempting to make a SIP call between an Asterisk 10 server and an
Asterisk 1.8 system but when it goes to VM and the first prompt plays the line
drops and I see on the V10 console:
[Jan 23 15:47:04] WARNING[7859]: chan_sip.c:8944 process_sdp: Insufficient
information for SDP (m=
Hi Kevin,
will grab a sip trace today and post it up.
--
Thanks, Phil
- Original Message -
On 01/23/2012 09:48 AM, --[ UxBoD ]-- wrote:
Hi,
I am attempting to make a SIP call between an Asterisk 10 server
and an
Asterisk 1.8 system but when it goes to VM and the first prompt
If in a multi-tenant environment be aware of
https://issues.asterisk.org/jira/browse/ASTERISK-17198 as VMs cannot be
forwarded :(
--
Thanks, Phil
- Original Message -
Paul Schenkeveld wrote:
exten = 5551234,n,Voicemail(1234,su)
I'm still running 1.4 (slowly configuring a 10
Hello all,
I attempted to make a couple of outbound calls this morning and always got the
busy tone. I checked the Asterisk console and was greeted with:
[Dec 29 11:29:22] WARNING[12039]: app_dial.c:2218 dial_exec_full: Unable to
create channel of type 'IAX2' (cause 20 - Unknown)
==
Hello all,
This may sound an odd question but if you initiate a call using AMI does it
adhere to what has been defined in the dial plan or do we have to write the
logic into the AMI call ?
--
Thanks, Phil
--
_
--
Please ignore as this was a user error!
--
Thanks, Phil
- Original Message -
Hello all,
This may sound an odd question but if you initiate a call using AMI
does it adhere to what has been defined in the dial plan or do we
have to write the logic into the AMI call ?
--
Thanks,
quotes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of --[
UxBoD ]--
Sent: Monday, December 12, 2011 11:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
Hello all,
I have recently upgraded to version 1.8.7.2 and have started to see the
following errors in the logs:
[ Dec 12 16:10:38] ERROR[10223] app_voicemail.c: IMAP Error: SELECT failed
[ Dec 12 16:10:38] ERROR[10223] app_voicemail.c: IMAP Error: must be in
SELECTED state
They are not
1.8.7.0 ... am using Zimbra as the backend IMAP storage.
--
Thanks, Phil
- Original Message -
On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote:
Hello all,
I have recently upgraded to version 1.8.7.2 and have started to see
the following errors in the logs:
From what version
as the backend IMAP storage.
--
Thanks, Phil
- Original Message -
On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote:
Hello all,
I have recently upgraded to version 1.8.7.2 and have started to
see
the following errors in the logs:
From what version?
--
Paul Belanger
-users-boun...@lists.digium.com] On Behalf Of --[
UxBoD ]--
Sent: Monday, December 12, 2011 11:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoiceMail and IMAP
Hmmm, just tried leaving a voicemail on a new mailbox where the
imapfolder
Could we have more hours in the day to play with all the goodiness ? cannot
keep up with everything at the moment :)
--
Thanks, Phil
- Original Message -
I know what you mean - I'd rather have a working x-beta1 that a
failing x.0
-Original Message-
From:
Hi,
Sorry for an OT post but striking out a bit at the moment trying to get a
response from Yealink RD. Has anybody successfully managed to get a Yealink
phone to work across Open-VPN when using tlsauth ? We really do hope that it is
possible due to the benefits tlsauth offers against DoS.
Ah, now this is interesting as one of our clients had the same problem the
other day; in our case when they performed the *8 they got an extension
unavailable from a completely different dialplan! This was on Asterisk 1.6
though with Snom phones.
--
Thanks, Phil
- Original Message -
cel_odbc.conf and then use adapative odbc I think.
--
Thanks, Phil
- Original Message -
Is anyone using CEL with a MySQL backed at all?
I've found a table schema but I'm guessing I need some sort of
cel_mysql.conf and don't even have a sample for that.
Can anyone give me any
Hi,
compiling up a new installation of Asterisk 1.8.5 on CentOS 6 X86_64 and am
seeing the following when running the make:
/usr/bin/ld: skipping incompatible /usr/lib/libpam.so when searching for -lpam
/usr/bin/ld: skipping incompatible /usr/lib/libssl.so when searching for -lssl
Thank you Dave.
--
Thanks, Phil
- Original Message -
On 07/29/2011 06:56 AM, --[ UxBoD ]-- wrote:
Hi,
compiling up a new installation of Asterisk 1.8.5 on CentOS 6
X86_64 and
am seeing the following when running the make:
/usr/bin/ld: skipping incompatible /usr/lib
Simple answer to all this is to install http://lync.microsoft.com/ ... good
luck ;)
--
Thanks, Phil
- Original Message -
Kevin P. Fleming wrote:
'alwaysauthreject' in not imcompliant with any RFCs; the RFCs
define
response codes that *can* be used to indicate (for example)
That is pretty interesting. I am writing a similar tool but using OSSEC to
identify the attacks and then share the data between nodes using Memcached and
AnyEvent. Both Asterisk and Apache, or any other server that can run OSSEC,
will be able to feed into the shared ban database.
--
Thanks,
, --[ UxBoD ]-- wrote:
Since upgrading to 1.8.5.0 I have had to add into modules.conf:
load = func_callerid.so
load = func_cdr.so
otherwise they do not get loaded even though I have set
autoload=yes.
Is this something you would expect as it is different behavior
If you are using OSSEC here are some rules:
rule id=1 level=5
decoded_aslocal-asterisk-denied/decoded_as
descriptionAsterisk Potentially Under Attack/description
/rule
rule id=10001 level=8 frequency=5 timeframe=10
if_matched_sid1/if_matched_sid
same_source_ip /
/asterisk/modules.
How could I help to debug this please ?
--
Thanks, Phil
- Original Message -
On 07/21/2011 04:31 AM, --[ UxBoD ]-- wrote:
Since upgrading to 1.8.5.0 I have had to add into modules.conf:
load = func_callerid.so
load = func_cdr.so
otherwise
Since upgrading to 1.8.5.0 I have had to add into modules.conf:
load = func_callerid.so
load = func_cdr.so
otherwise they do not get loaded even though I have set autoload=yes.
Is this something you would expect as it is different behavior to 1.8.3.0 and I
do not see any issues in
- Original Message -
On 07/21/2011 04:31 AM, --[ UxBoD ]-- wrote:
Since upgrading to 1.8.5.0 I have had to add into modules.conf:
load = func_callerid.so
load = func_cdr.so
otherwise they do not get loaded even though I have set
autoload=yes.
Is this something you would expect
Been looking at SwitchVox and how it handles mobility using virtual extensions.
Does somebody have any examples on how this can be achieved with Asterisk ? I
have Bria on my Android and it would be nice if I could get my office phone
and/or cell to ring.
--
Thanks, Phil
--
- Original Message -
On Sat, 16 Jul 2011 11:01:07 +0100 (BST)
--[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
On 11-07-15 02:18 PM, Doug Lytle wrote:
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine. What am I missing
- Original Message -
- Original Message -
On Sat, 16 Jul 2011 11:01:07 +0100 (BST)
--[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
On 11-07-15 02:18 PM, Doug Lytle wrote:
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine
Hi,
When starting Asterisk (1.8.5.0) I see in messages:
[Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module
'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol:
ast_aji_get_client
[Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be
- Original Message -
- Original Message -
From: --[ UxBoD ]-- ux...@splatnix.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, July 18, 2011 11:42:25 AM
Subject: [asterisk-users] chan_gtalk load error
Hi
- Original Message -
On 11-07-15 02:18 PM, Doug Lytle wrote:
--[ UxBoD ]-- wrote:
I back leveled to 1.8.3 and that works fine. What am I missing as
app_macro has been installed okay?
Macro was depreciated in 1.6 and most likely removed in 1.8.5
Removed, no. However in future
Have just tried to test an upgrade to 1.8.5 and when making an outbound call I
get:
[Jul 15 18:48:52] WARNING[21038]: pbx.c:4071 pbx_extension_helper: No
application 'Macro' for extension (context, XX, 1)
I back leveled to 1.8.3 and that works fine. What am I missing as app_macro has
Hi, When is the next release planned for as very keen to get it into Production
but require the call pickup fix.
--
Thanks, Phil
--
_
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New to Asterisk? Join
Hi,
Needed to test follow-me this evening on Asterisk 1.6.2.17 and received the
following message:
== Spawn extension (international-US, 0114407590XX, 5) exited non-zero on
'Local /0114407590XX@aXX-a62a;2'
-- no live channels left. exiting.
I have not seen that before. What does
- Original Message -
On Fri, 2011-05-06 at 20:21 +0100, --[ UxBoD ]-- wrote:
Are you not seeing issues with *8 call pick up then ?
Nope, I double checked it after seeing someone saying they had issues
with it and it is fine on the installation I have.
Which release are you
- Original Message -
Dear;
Where I can find a new documentation for Asterisk 1.8?
Where is the wrong in that line? I see it is as 1.8 version !
500 = 1234,Operator,opera...@gama.com
Regards
Bilal
---
You are using an old format for specifying the
I know a lot has changed over the past couple of years, and even monthly, and
that Asterisk running within a virtualised environment is very happy indeed. If
one would only be using SIP/IAX would Xen/KVM be the best solution ? / or
perhaps VServer/LXC maybe advantageous due to binary hashing.
- Original Message -
On Thu, 2011-05-05 at 14:13 +, satish patel wrote:
Hi All,
Just wondering is it safe to use asterisk 1.8 latest branch on
production ?
http://svn.asterisk.org/svn/asterisk/branches/1.8/ Revision
317100
-S
We've been running 1.8.3.2 with
- Original Message -
I've just started deploying these (well the T28P model) after years
of
Snom issues and they look pretty good (although the documentation is
execrable; if you thought the Snom stuff was obtuse Yealink have got
them knocked into a cocked hat!).
Anyway, for
- Original Message -
On 11-04-07 08:20 AM, Satish Patel wrote:
Is it ture 1.8.3 is unstable? We are planning to put this in
production.
Please suggest me what should I do?
This is a loaded question, since it really depends on what you plan
to
do. What does your migration plan
- Original Message -
My Asterisk 1.8 (with Dahdi/Wanrouter) is crashing every minute or 2.
It just keeps restarting.
Any pointers on log files to watch? I tried to debug it but i
couldn't find a good reason for the crashes.
Maby the box is just overloaded or something like that but
Hi,
With Asterisk 1.8 is it now possible to register the same SIP account at
multiple endpoints and for both to ring when the associated extension is dialed
?
--
Thanks, Phil
--
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- Original Message -
Hi,
could anybody suggest a usable doorphone and magnetic door opener
hardphone system for me, please? Of course should be connectable to
asterisk. I am in the EU, should be available here.
thank you,
Csaba
--
Hi,
Would someone know where I can download the CEL schema for (create commands)
for PostgreSQL please ?
--
Thanks, Phil
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- Original Message -
Pretty sure I saw those on wiki.asterisk.org .
Thanks,
--Warren Selby, dCAP
On Feb 28, 2011, at 10:39 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
Hi,
Would someone know where I can download the CEL schema for (create
commands) for PostgreSQL please
Apologies in advance if this has come up a thousand times before but is there
any way to stop this error in 1.8 ?
[ Feb 26 15:09:09] ERROR[6678] chan_sip.c: No SRTP module loaded, can't setup
SRTP session.
--
Thanks, Phil
--
Hello all,
After numerous issues with Snom phones (360/370/870) potentially looking to
migrate too Yealink as their product range looks very promising indeed.
Are any of you using them with Asterisk ? How do they perform ? Do you use mass
deployment at all ?
Would be very interested to
- Original Message -
I found some great pieces of script on the internet that I've
combined to allow Asterisk to send voicemails as an MP3 file, and
encode the sender name and number as well as message number as tags
into the MP3 file. I even include a cover art image which has our
Hi,
Over the weekend tried to setup a test using the new app_calendar code but
receiving the following error:
[Jan 17 09:23:35] WARNING[27663]: res_calendar_icalendar.c:146 fetch_icalendar:
Unable to retrieve iCalendar 'testcal' from
- Original Message -
Try to disable certificate verification on the app. I had never tried
it personally but check for that option.
Sent from my iPhone
On Jan 17, 2011, at 5:51 PM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
Hi,
Over the weekend tried to setup a test using
- Original Message -
--[ UxBoD ]-- wrote:
- Original Message -
Yes exactly that indeed. Though Asterisk appears to ignore which
context the user is in and selects default instead. Beginning to
think that it is a bug.
I got it figured out.
In your voicemail.conf
- Original Message -
Is that user trying to forward to xxx in the same context?
On Fri, Dec 17, 2010 at 5:57 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
Experiencing a problem when users attempt to forward a voicemail
from within VoiceMailMain(Option 8) I see the console message
Experiencing a problem when users attempt to forward a voicemail from within
VoiceMailMain(Option 8) I see the console message:
Couldn't not find mailbox XXX in context default
As why are running in a multi-tenant environment voicemail.conf has been
separated into individual contexts. The
- Original Message -
reply please
On 12/17/2010 10:03 AM, Nikhil wrote:
Hi
Does anyone knows how to find out a call in a asterisk is
external incoming ,external out going or internal
Thanks
Nikhil
Perhaps if you were clearer in the question you are asking ?
--
- Original Message -
HI,
My system been attacked from someone I guess, kindly check the link below
How can I stop the ircd attack
http://pastebin.com/tbjh5qzP
regards
*
No employee or agent is authorized to conclude any
Hi,
Has anybody had 1.8 crashing for no reason at all ? It has happened a couple of
times so far and when I check /var/log/asterisk/messages nothing is in there at
all :(
--
Thanks, Phil
--
_
-- Bandwidth and Colocation
- Original Message -
On Fri, Nov 26, 2010 at 8:00 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
Hi,
Has anybody had 1.8 crashing for no reason at all ? It has happened
a couple of times so far and when I check /var/log/asterisk/messages
nothing is in there at all :(
--
Thanks
- Original Message -
Hello,
I notice that the following proces is running :
astcanary /var/run/asterisk/alt.asterisk.canary.tweet.tweet.tweet 1527
What is this ??
Kind regards,
Jonas.
You are running Asterisk with priority set. Check /etc/asterisk/asterisk.conf
for the
I have read the wiki entry but unsure when we would likely see a 1.8.0.1 beta
or release candidate ?
--
Thanks, Phil
--
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New to Asterisk? Join us for a
- Original Message -
On 10-11-23 07:31 AM, --[ UxBoD ]-- wrote:
I have read the wiki entry but unsure when we would likely see a
1.8.0.1 beta or release candidate ?
It will be Asterisk 1.8.1-rc1 and that is now available (as of a few
minutes ago)
http://www.asterisk.org/node
- Original Message -
Hello,
We tried to upgrade our Asterisk from 1.6.2.13 to 1.6.2.14, after
trying
to install iksemel (jabber support) and spandsp, but now Asterisk
doesn't work anymore and we can't get it to run, althorugh we tried to
remove it completely and reinstall
- Original Message -
Hi all!
A few days I have problems connecting to the Internet on my house and
since then my local SIP extensions are no longer registered against
the
local Asterisk server.
I'm using Asterisk 1.4.24.1. I was researching on the Internet and I
found that it
- Original Message -
Hi, Phil.
A few days I have problems connecting to the Internet on my house
and since then my local SIP extensions are no longer registered
against the local Asterisk server.
I'm using Asterisk 1.4.24.1. I was researching on the Internet and
I
Has anyone managed to successfully connect Asterisk to Zimbra using the Jabber
service ? I have opened http://issues.asterisk.org/view.php?id=18198 as it
keeps failing for me. Am wondering whether it is due to using a self signed
cert.
--
Thanks, Phil
--
- 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
- Original Message -
When we designed our systems on asterisk we designed it to me multi-tenant. Se
we use customer prefixes on all extensions. This allows us to have multiple
customers using the same extension pools. It also reduces the hack foot print
as hackers must know the
Hi,
Running 1.6.2.11 and getting the odd occation that all phones will start
ringing with nobody on the other end. From the information we have received
from the client we can see that a call comes in, it is either answered or not
answered, but at the same time a second call comes in and it
- Original Message -
Rough area. Consider yourself lucky you haven't been ripped apart :P
Pete wrote:
I hope someone has helped poor Rob, I would as I am just over the
bridge
in Bristol, UK but some evil internet scammer has stolen all my
money! ;)
Cheers!
On 15/09/10
- Original Message -
Roger Burton West wrote:
I want to hook one of them to the PSTN. Given that I am in
the UK, what is a reasonably easily-available device to
provide an FXO interface from a Linux box, with a minimum of
faffing around with drivers? Just one line is needed,
- Original Message -
On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
Odd problem have just noticed in that when I call into the PBX DAHDI
detects the call and hands it off to the extension, if I then hang
up it still continues to process through
- Original Message -
On Tue, 24 Aug 2010, --[ UxBoD ]-- wrote:
- Original Message -
On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]--
ux...@splatnix.net
wrote:
Odd problem have just noticed in that when I call into the PBX
DAHDI
detects the call and hands it off
- Original Message -
On Mon, Aug 23, 2010 at 1:03 PM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
This is a real strange one and trying to phantom it out. One of our
clients is connected to our Asterisk installation, from two sites,
via VPN which works great. Every so often one
Hi,
Running Asterisk 1.6.2.11 and wondering what would be the best way to send an
email when a missed call has occurred ? I believe you can modify [stdexten] is
this still the case in V1.6 ?
--
Thanks, Phil
--
_
-- Bandwidth
Hi,
Odd problem have just noticed in that when I call into the PBX DAHDI detects
the call and hands it off to the extension, if I then hang up it still
continues to process through the dialplan.
This is what I have in chan_dahdi.conf:
[channels]
language=en
echocancel=yes
usecallerid=yes
Hi,
This is a real strange one and trying to phantom it out. One of our clients is
connected to our Asterisk installation, from two sites, via VPN which works
great. Every so often one of the sites VPN tunnel goes does for say a couple of
seconds. When that happens all the extensions,
Hi,
Do any of you have these phones ? How have you found it ? Are you using them
over WiFi or hard wired ? Does it play nicely with Asterisk ?
Need to replace my Snom M3s and this phone maybe a contender.
--
Thanks, Phil
--
- Original Message -
On Fri, 6 Aug 2010, --[ UxBoD ]-- wrote:
Hi,
Do any of you have these phones ? How have you found it ? Are you
using them over WiFi or hard wired ? Does it play nicely with
Asterisk ?
Need to replace my Snom M3s and this phone maybe a contender
- Original Message -
- Original Message -
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
that
we are unable to URI dial our clients
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are
unable to URI dial our clients. We run a multi-tenant server and have set
sip.conf to forward calls to a public context based on incoming domain name.
This was all working before but not it is complaining of a loop
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that
we are unable to URI dial our clients. We run a multi-tenant server
and have set sip.conf to forward calls to a public context based on
incoming domain name. This was all working before
- Original Message -
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
that
we are unable to URI dial our clients. We run a multi-tenant server
and have
Hi, all
Would any of you be able to suggest physical SIP phones that support inbuilt
VPN capabilities; akin to the Snom 370/870 ?
--
Thanks, Phil
--
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New
* Skype for Asterisk needs to run on this - so this means x86, right?
or x86_64 is fine
--
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- Original Message -
On Thu, 10 Jun 2010, Michelle Dupuis wrote:
I'm looking for a small formfactor mobo for an install that needs to
handle 25 phone sets (no transcoding). I found a new dual atom
1.66GHz
mobo - anyone know what kinds of call volume that will handle?
On Thu,
- Original Message -
The Asterisk Development Team has announced the release of Asterisk
1.6.2.8. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
Which release will http://issues.asterisk.org/view.php?id=17135 make it into;
was
- Original Message -
On Friday 28 May 2010 12:50:19 --[ UxBoD ]-- wrote:
- Original Message -
You're missing this in your chan_dahdi.conf:
#include dahdi-channels.conf
Hmm, I changed the signalling as per a previous post and now it is
okay. Why
Looking for some help from the UK please. I backed up all my Asterisk
configuration before re-installing the server from 32 - 64 bit. Unfortunately
I did not transfer the backup to another machine!
I now have a TDM400P that is not picking up the line. Can you see what I have
done wrong
- Original Message -
--[ UxBoD ]-- wrote:
Looking for some help from the UK please. I backed up all my
Asterisk configuration before re-installing the server from 32 - 64
bit. Unfortunately I did not transfer the backup to another
machine!
I now have a TDM400P
- Original Message -
On Fri, 28 May 2010, --[ UxBoD ]-- wrote:
[NON-Text Body part not included]
Er, my mailer's obviously struggled to interpret this, however did you
do
the include and I also have this in /etc/modprobe.d in a file:
options wctdm opermode=UK
Gordon
- Original Message -
- --[ UxBoD ]-- ux...@splatnix.net wrote:
/etc/asterisk/chan_dahdi.conf
- [channels]
language=en usecallerid=yes
cidsignalling=v23 sendcalleridafter = 2
rxgain=2.0 txgain=3.0
progzone=uk signalling=fxo_ks
callerid
- Original Message -
2010/5/14 --[ UxBoD ]-- ux...@splatnix.net:
- Original Message -
Hello,
i try to use soap in the phpagi.
i want to call a function from a web service
when a user call a telephne failed.
this is my phpagi script, Could you show me what's wrong
- Original Message -
Hello,
i try to use soap in the phpagi.
i want to call a function from a web service
when a user call a telephne failed.
this is my phpagi script, Could you show me what's wrong ? becasue i
can't excute it successfully.
please open the following url to
- Original Message -
mike mosier wrote:
Hey all.
My boss asked me to implement the following
When DID 713xxx is dialed send an email to mmos...@xxx.com
mailto:mmos...@xxx.com. with the time date and CID included in the
email. I know how to code some but am looking for
- Original Message -
Randy-
On Wed, Apr 21, 2010 at 5:33 PM, Steve Murphy m...@parsetree.com
wrote:
Assuming that every such spamming/hacking/attack site is funded on
a stolen identity/CC number, it will soon sink into Amazon that
they are
getting a bad rep, and losing money
- Original Message -
Think we need some solution WITHIN the Asterisk core. Roderick A.
suggested something that looks nice using iptables, some others have
pointed out using RBL or fail2ban, but the best would be to have some
generic solution not dependant on third party programs.
- Original Message -
On Tue, 13 Apr 2010, Alyed wrote:
Think we need some solution WITHIN the Asterisk core. Roderick A.
suggested something that looks nice using iptables, some others have
pointed out using
RBL or fail2ban, but the best would be to have some generic solution
- Original Message -
Speaking of all these attacks, are there any good web managed security
monitor tools for CentOS out there that can be installed on the system
so that it can give us a visual of let's multiple failed attempts
against SSH or HTTPd?
Something nice that is simple
, --[ UxBoD ]-- ux...@splatnix.net
wrote:
- Original Message -
Speaking of all these attacks, are there any good web managed
security monitor tools for CentOS out there that can be installed on
the system
so that it can give us a visual of let's multiple failed attempts
against
- Original Message -
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
- Original Message -
On 04/12/2010 12:05 PM, Randy R wrote:
On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman
dhart...@djhsolutions.com wrote:
I don't think anyone else brought up the Spamhaus DROP project.
It's a
blacklist of IP addresses and address ranges which are known to
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