On 26 June 2014 15:42, Anurag Rana anuragrana31...@gmail.com wrote:
Hi All,
In asterisk, default directory to store the call-recording files is
/var/spool/asterisk/monitor.
Can we change this directory? How?
--
Anurag Rana
http://newbie42.blogspot.in/
On the trampoline of life's
Hello,
Throwing calls from Asterisk to PSTN (via a VoIP gateway) some operators
sends an explaining audio, in situations as:
The phone number does is not assigned
The phone is powered off
etc.
The audio is sent before the call to be answered.
So, in an automatic dialling application I'd like to
Hi All.
Someone is attacking on my SIP server.
There are lot of requests coming in and I am not able to stop it because I
am unable to detect the IP address.
I used wireshark to capture the packets.
Although I am using very strong password for my SIP users but still is
there any way to drop
Hi,
Change the protocol from tcp to udp in iptables.
~Arun
On 27 Jun 2014 20:07, Anurag Rana anuragrana31...@gmail.com wrote:
Hi All.
Someone is attacking on my SIP server.
There are lot of requests coming in and I am not able to stop it because I
am unable to detect the IP address.
Hi,
Install fail2band and change sip listen port to avoid attack
With regards
N.Prakash
--
From: Anurag Rana anuragrana31...@gmail.com
Sent: 27-06-2014 08:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject:
I added bot rules TCP as well as UDP. Still not working.
How changing SIP listen port will prevent it. Please explain.
I will try fail2band.
On Fri, Jun 27, 2014 at 8:16 PM, Prakash N prakas...@tevatel.com wrote:
Hi,
Install fail2band and change sip listen port to avoid attack
With
Both Rules* (typo in last mail)
On Fri, Jun 27, 2014 at 8:19 PM, Anurag Rana anuragrana31...@gmail.com
wrote:
I added bot rules TCP as well as UDP. Still not working.
How changing SIP listen port will prevent it. Please explain.
I will try fail2band.
On Fri, Jun 27, 2014 at 8:16 PM,
On 27 Jun 2014, at 15:37, Anurag Rana anuragrana31...@gmail.com wrote:
There are lot of requests coming in and I am not able to stop it because I am
unable to detect the IP address.
I used wireshark to capture the packets.
If you can capture the packet, surely you have the IP? If they intend
very simple,
yet effective
http://www.palner.com/blog/171/asterisk-no-matching-peer-found-block/
Am 27.06.2014 16:58, schrieb Steven Howes:
On 27 Jun 2014, at 15:37, Anurag Rana anuragrana31...@gmail.com
mailto:anuragrana31...@gmail.com wrote:
There are lot of requests coming in and I am not
This is a common issue and is covered in the mailing list archives multiple
times.
Do a Google search for something like:
site:lists.digium.com fail2ban
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes
Sent: Friday,
I am working on an AGI script and all is going well except I can not get
any of my VERBOSE commands to display.
Is there any undocumented reason for this to occur? I am able to set
variables, call other commands ect..
I am sending my verbose command in the following format (VERBOSE
+1 fail2ban
Very easy and very effective.
On 27/06/2014 10:52 AM, Anurag Rana wrote:
Both Rules* (typo in last mail)
On Fri, Jun 27, 2014 at 8:19 PM, Anurag Rana
anuragrana31...@gmail.com mailto:anuragrana31...@gmail.com wrote:
I added bot rules TCP as well as UDP. Still not working.
Hi Brian,
This is the path that I've started down. It's a 8-Span Audiocodes Mediant
2000 running 6.4. Plan to look at the various syslog options today and
see what is available.
Thanks for your input...
On Thu, Jun 26, 2014 at 7:37 PM, Brian LaVallee b.laval...@globaltank.jp
wrote:
There
what if yoy change the verbose on the cli?
cli core set verbose 4
and then try again
i usually put on my perl agi something like
$verbose=5;
AGI-verbose(the number is $number, $verbose);
hope it helps.
rv
2014-06-27 11:24 GMT-04:00 Bryant Zimmerman brya...@zktech.com:
I am working on an
Hey all
Please disregard my question. I was looking for the word Verbose to show
up. I was just being dense.
There was no real issue it is working just different than what I was
expecting.
Thanks
Bryant
From: Bryant Zimmerman
In sip.conf change listen port 5060 to some other number like 7242 any
number ,then restart asterisk . Register sip phone with listen port (7242)
Example
Domain: 192.168.1.10:7242
With regards
N.Prakash
--
From: Anurag Rana anuragrana31...@gmail.com
Sent:
Is it possible to have the AMI Originate call a local extension, then configure
the local extension to do something like this
Set(CALLERID(num-pres)=allowed_passed_screen)
Dial some number passed in via the Originate
If so...
1) How would I pass a value from the Originate request to
Block the ip?
You should only enable sip for your specific clients in iptables.
Sent from Samsung Mobile
div Original message /divdivFrom: arun kumar
arunvsadni...@gmail.com /divdivDate:27/06/2014 4:42 PM (GMT+02:00)
/divdivTo: Asterisk Users Mailing List - Non-Commercial
Anurag,
Here is small script, that will check your logs and will block the IPs.
http://www.didforsale.com/blog/is-your-asterisk-system-under-heavy-attack
This is good if you dont expect any registration. If you do have some valid
registration, you might want to add some counter to see how time
I think your asterisk server is behind firewall or some sort of NAT where
the out to in packets are getting masqueraded with local or DMZ IP of your
firewall / gateway box.
Fix this first to get fail2ban detect the correct public IP.
Otherwise fail2ban will ban your local GW IP due to which you
Fail2band installation
http://striker24x7.blogspot.in/2011/07/fail2ban-in-asterisk.html?m=1
Iptables
http://striker24x7.blogspot.in/2014/03/simple-iptables-script.html?m=1
With regards
N.Prakash
--
From: Anurag Rana anuragrana31...@gmail.com
Sent: 27-06-2014
Right Mitul. System is behind some gateway.
On Fri, Jun 27, 2014 at 10:06 PM, Mitul Limbani mi...@enterux.in wrote:
I think your asterisk server is behind firewall or some sort of NAT where
the out to in packets are getting masqueraded with local or DMZ IP of your
firewall / gateway box.
Can't use anything which block IP addresses because my system is behind a
gateway and attacker gets the address of that gateway. In this way I will
end up blocking myself.
Please suggest something else.
On Fri, Jun 27, 2014 at 10:24 PM, Anurag Rana anuragrana31...@gmail.com
wrote:
Right
No way out. Fix ur gateway which is masquerading out to in traffic.
And do some research as others mentioned instead of expecting quick fix.
Mitul
On 27-Jun-2014 10:45 PM, Anurag Rana anuragrana31...@gmail.com wrote:
Can't use anything which block IP addresses because my system is behind a
Ok. Thanks. :)
On Fri, Jun 27, 2014 at 11:05 PM, Mitul Limbani mi...@enterux.in wrote:
No way out. Fix ur gateway which is masquerading out to in traffic.
And do some research as others mentioned instead of expecting quick fix.
Mitul
On 27-Jun-2014 10:45 PM, Anurag Rana
Please don't top-post.
Please trim posts to the specific post you are replying to.
On Fri, 27 Jun 2014, Anurag Rana wrote:
Can't use anything which block IP addresses because my system is behind
a gateway and attacker gets the address of that gateway. In this way I
will end up blocking
Is there something I can do regarding this issue?
On 16 June 2014 11:39, Tiago Geada tiago.ge...@gmail.com wrote:
Hi,
Thank you for your explanation about channel halds .. These .call files
are always different from other calls.
Well I would like some custom var to have a piece of
On Thu, Jun 26, 2014 at 12:30 PM, CDR vene...@gmail.com wrote:
I did what we use to dim that is add a line to pjsip.conf like
#include /etc/asterisk/pjpeers.conf
but the file is not loaded. Am I doing something wrong this
functionality is disabled?
Nope, this should work fine. You might have
On Fri, Jun 27, 2014 at 3:12 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
On 26 June 2014 15:42, Anurag Rana anuragrana31...@gmail.com wrote:
Hi All,
In asterisk, default directory to store the call-recording files is
/var/spool/asterisk/monitor.
Can we change this directory? How?
Hi
On Wed, Jun 25, 2014 at 9:30 AM, Rafael Visser visser.raf...@gmail.com wrote:
Hi gurus!!!
I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn
Every minute asterisk sends an OPTION Request, i beleived that it's related
to qualify functions.
The every minute annoyng answer of
On Sat, Jun 21, 2014 at 5:57 AM, Inventions resea...@businesstz.com wrote:
Can anyone tell me how to implement a popup URL native asterisk when
making outbound call?
For example, a user (A Part) dial from a softphone number 07112233, when a
call is received (or even before) by B-Part, a CRM
Hi All,
I am trying to execute some AGI script no matter what extension is called.
There is 'h' extension to call AGI script when any call hangs up no matter
what extension hangup.
for example -
[some-context]
/// something here which call AGI script no matter what extension receive
call.
On Fri, Jun 27, 2014 at 1:30 PM, Tiago Geada tiago.ge...@gmail.com wrote:
Is there something I can do regarding this issue?
Have you looked at these wiki pages?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files
The setvar parameter may help here.
The Asterisk Development Team has announced the releases of:
DAHDI-Linux-v2.9.2-rc1
DAHDI-Tools-v2.9.2-rc1
dahdi-linux-complete-2.9.2-rc1+2.9.2-rc1
This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
You can learn the source code of predictive dialer like vicidial, osdial, etc.
You have to discover how they work, then you can implement it on your asterisk.
But it's going to take a lot of time.
Sorry for my bad English
Regards,
Ikka Vertika
Jakarta - Indonesia
Rusty Newton
What CRM your going to use?
With regards
N.Prakash From: Rusty Newton
Sent: 28-06-2014 01:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Popup URL for outgoing calls.
On Sat, Jun 21, 2014 at 5:57 AM, Inventions resea...@businesstz.com wrote:
You can use vtiger or sugar
Both are working with asterisk.
On Fri, Jun 27, 2014 at 9:04 PM, Prakash N prakas...@tevatel.com wrote:
What CRM your going to use?
With regards
N.Prakash From: Rusty Newton
Sent: 28-06-2014 01:01 AM
To: Asterisk Users Mailing List - Non-Commercial
I could not find a way to set a max on the calls allowed through a
PJSIP endpoint.
In case we decide to add it, the we need another reason for the call
to fail in the Dial application, something like limit reached
Am I missing this capability?
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