Hello,
We had been seeing SIP-guessing attacks on our Asterisk server here.
While it wasn't that hard to write a once-a-minute cron job to spank
the lusers, that runs once a minute and creates little spikes in the
usage and I/O graphs, and is slower to respond than I'd really prefer.
I felt that
On 12/08/2010 11:46 PM, mayamatakeshi wrote:
Hello,
does anyone know a way to load test the SkypeForAsterisk module without
actually generating calls to Skype Network? (only inside test
environment). I mean, is there any way to simulate the endpoint SFA
talks to?
No, there is not. It talks
I have now logged issue number 0018447 relating to this query.
Thanks all for your responses.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: 03 December 2010 22:53
To: Asterisk Users
I do not have log examples to provide but do have info about other issues.
There is a nocolor option in asterisk.conf that can turn off color.
logger.conf has a provision to use syslog directly.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 9, 2010, at 5:57 AM,
sorry i am not familiar with sshguard, but you can also try ossec by
trend micro http://www.ossec.net/ it can auto-block an IP address using
iptables. you can also follow this howto for asterisk:
http://sysbrain.wordpress.com/2010/05/24/asterisk-ossec-part-ii/
hope this helps.
regards
Ron
On
On Dec 9, 2010, at 5:57 AM, Joe Greco wrote:
Hello,
We had been seeing SIP-guessing attacks on our Asterisk server here.
While it wasn't that hard to write a once-a-minute cron job to spank
the lusers, that runs once a minute and creates little spikes in the
usage and I/O graphs,
Hi RR,
I've not tried compiling 1.8.1-rc1 on Solaris yet and I've not come across this
issue as of yet. I did build 1.8.0-rc5 on Solaris 10 without any build error's
though. I'm not sure if the code has changed that much between 1.8.0-rc5 and
1.8.1-rc1.
I'm no coding guru by anyone's
I'm not sure if this is the log entry you are looking for. I had many of these
last
night.
[Dec 9 06:47:51] NOTICE[5630]: chan_sip.c:15593 handle_request_register:
Registration from '106 sip:1...@mywanaddress' failed for '121.11.158.174' -
Wrong password
If you need more information from
can someone give more eduaction to me about what the asterisk exchange is
all about?
thanks
On Thu, Dec 9, 2010 at 5:43 AM, Sevana Oy sa...@sevana.fi wrote:
Hi,
A couple of months ago we registered our product AQuA at Asterisk Exchange.
We were told that it collects like 14K visitors per
On Thu, Dec 9, 2010 at 1:29 AM, Daniel Knoll dan...@danielknoll.de wrote:
Hey Guys,
for debugging i need to read the Events from AMI. But i have a lot of
unwanted RTCPSent Events.
How can i filter this Events in Asterisk 1.6.2.x Version of Asterisk?
You can control to some extent on what
On 12/09/2010 08:15 AM, Bruce McAlister wrote:
I have now logged issue number 0018447 relating to this query.
The real question here is how you define 'not responding to INVITEs'.
According to the RFC, Asterisk must wait 64*T1 for a response to an
outbound INVITE, which is 32 seconds. If
Hi,
This is it: http://www.asteriskexchange.com/
It's also good to know that people from such respectful community may not know
it at all. Besides, the ones from Digium who read and moderate also don't reply
to my post - good to know that too :)
- Original Message -
From: Goke M
On Tue, Dec 07, 2010 at 12:48:43AM +0100, Giuseppe D'alessio wrote:
Hi, i have context in a dialplan, I want to execute this context
without insert the Answer Application (s? ..without call any ext).
[snipped]
I'm puzzeled:
What is the question? Is something going wrong or unexpected? (I'm to
The AsteriskExchange.com is a directory of products and services that
complement, extend, or enhance Asterisk. As announced at AstriCon 2009, Digium
answered the demand for a marketplace that would guide Asterisk users to
available solutions. Free listings are available for free products and
On Thu, Dec 09, 2010 at 07:57:37AM -0600, Joe Greco wrote:
Specifically looking for examples of (or how to generate)
1).*No registration for peer '.*' (from HOST)
2).*Host HOST failed MD5 authentication for '.*' (.*)
3).*Failed to authenticate user .*@HOST.*
If anyone who is
On Thu, Dec 9, 2010 at 10:31 AM, Daniel Tryba dan...@tryba.nl wrote:
You could use SIPVicious to run attacks on your own servers:
http://code.google.com/p/sipvicious/
Yeah, why not? All the criminals on the internet are using it, too! ;^)
I'm seeing 1-4 scans per day on the average. And
BTW, the issue was created yesterday, but I didn't think there was a need to
post it here but nevertheless for posterity, the Issue ID is: 18442
Thanks
\RR
On Wed, Dec 8, 2010 at 6:57 PM, RR ranjt...@gmail.com wrote:
On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher tles...@digium.comwrote:
On Thu, Dec 9, 2010 at 10:02 AM, Bruce McAlister
bruce.mcalis...@blueface.ie wrote:
Hi RR,
I’ve not tried compiling 1.8.1-rc1 on Solaris yet and I’ve not come across
this issue as of yet. I did build 1.8.0-rc5 on Solaris 10 without any build
error’s though. I’m not sure if the code has
Thank you for the reply.
On 8 Dec 2010 at 13:38, Danny (Danny Nicholas da...@debsinc.com) commented
about RE: [asterisk-users] Configuring Softphone:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary
http://john8802.ru.gg/Hcthealtsor.htm
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
On 9 Dec 2010 at 13:31, Gary (Gary Kuznitz docf...@theoffice.la) commented
about
[asterisk-users] (Fwd) Re: Configuring Softphone:
Thank you for the reply.
On 8 Dec 2010 at 13:38, Danny (Danny Nicholas da...@debsinc.com) commented
about RE: [asterisk-users] Configuring Softphone:
On Thu, 9 Dec 2010, Gary Kuznitz wrote:
I'm getting closer. Express Talk is now making the call.
I'm getting an error on the cmd line:
-- Executing [91myareacodepho...@dlpn_dialplan1:1] Macro(SIP/120-
b6003810, trunkdial-failover-0.3|Dahdi/g1/1MyAreaCodePhone#||trunk_1|) in
new
stack
Thanks for the reply.
On 9 Dec 2010 at 20:56, Steve (Steve Edwards asterisk@sedwards.com)
commented about Re: [asterisk-users] (Fwd) Re: Configuring Softp:
On Thu, 9 Dec 2010, Gary Kuznitz wrote:
I'm getting closer. Express Talk is now making the call.
I'm getting an error on the
I see in sip debug it says Audio is at port 10342
Express Talk suggests Audio at UDP 8000-8020
I've tried changing Express Talk to 1 and forwarded 1-10400.
Is there a possibility Express Talk won't work in the 1 range?
Is it possible to limit Asterisk to 8000-8020?
Thank you,
Gary
On 9 Dec 2010 at 22:32, Gary (Gary Kuznitz docf...@theoffice.la) commented
about [asterisk-users] Audio ports:
I see in sip debug it says Audio is at port 10342
Express Talk suggests Audio at UDP 8000-8020
I've tried changing Express Talk to 1 and forwarded 1-10400.
Is there a
On 12/08/2010 02:48 PM, Alex Saavedra wrote:
Jonas,
I've been using H.264 and H.263+ with a few Grandstream GVX3140. When
using H.264 the image quality was better, and required bandwidth
appeared lower compared with H.263+. In fact H.264 is expected to
consume less bandwidth for as much as
Hi Gary,
I not using anything to create my dialplan. I'm trying to add a softphone to
a dialplan
that was created a couple years ago by someone that knew what they were doing.
Everything else in the dialplan works. As you can see I don't understand how
to
create a dialplan and I'm
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