Someone has hacked into our system and is making calls overseas.
How can I:
1. Find out the where the calls are originating from?
2. Block all calls that are not authorized?
Our system is in the USA.
Only calls from inside our LAN are allowed.
Thank you,
Gary Kuznitz
how
to get in
to our box.
Thanks you,
Gary Kuznitz
On 22 Nov 2010 at 13:11, Danny (Danny Nicholas da...@debsinc.com) commented
about RE: [asterisk-users] Someone has hacked into our :
From: Gary Kuznitz [mailto:docf...@theoffice.la]
Sent: Monday, November 22, 2010 12:20 PM
To: Danny
Thank you for the reply...
Comments below...
On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher asterisk-
us...@lists.digium.com) commented about Re: [asterisk-users] Someone has
hacked
into our :
On Monday 22 November 2010 17:10:31 Gary Kuznitz wrote:
I have the log now. I'd like to know
On 23 Nov 2010 at 16:54, Joseph (Joseph syscon...@gmail.com) commented about
Re: [asterisk-users] Someone has hacked into our :
On 11/23/10 14:18, Gary Kuznitz wrote:
Thank you for the reply...
Comments below...
On 22 Nov 2010 at 17:23, Tilghman (Tilghman Lesher asterisk-
us
Thank you for the reply.
On 23 Nov 2010 at 18:51, John (John Novack jnov...@stromberg-carlson.org)
commented about Re: [asterisk-users] Someone has hacked into our :
Gary Kuznitz wrote:
Thank you for the reply...
Comments below...
On 22 Nov 2010 at 17:23, Tilghman (Tilghman
I have been told that my logic in extentions.conf is wrong in trying to
configure a SIP
software phone called Express Talk (remote) .
I'd like to make outgoing calls and calls to local extensions.
Could someone please look at my configuration files at
http://pastebin.com/ajp62wqF
and see what
Thank you for the reply.
Comments below...
On 30 Nov 2010 at 20:21, Warren (Warren Selby wcse...@selbytech.com)
commented about Re: [asterisk-users] Trying to configure a SIP so:
On Tue, Nov 30, 2010 at 8:04 PM, Gary Kuznitz docf...@theoffice.la wrote:
I have been told that my logic
Shouldn't Asterisk be listening on UDP port 5060?
I'm working with an Asterisk installation running in Ubuntu. Asterisk is
running but
non of the phone are connecting. I ran netstat -a and I didn't see 5060. Am I
supposed to see something listening?
Thank you,
Gary
--
On 2 Dec 2010 at 15:11, Danny (Danny Nicholas da...@debsinc.com) commented
about RE: [asterisk-users] Asterisk ports:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gary Kuznitz
Sent: Thursday
Thank you for the reply.
On 2 Dec 2010 at 16:23, Jeff (Jeff LaCoursiere j...@sunfone.com) commented
about Re: [asterisk-users] Asterisk ports:
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
Shouldn't Asterisk be listening on UDP port 5060?
I'm working with an Asterisk installation running
Thanks for the reply.
On 2 Dec 2010 at 14:11, Steve (Steve Edwards asterisk-users@lists.digium.com)
commented about Re: [asterisk-users] Asterisk ports:
On Behalf Of Gary Kuznitz
Shouldn't Asterisk be listening on UDP port 5060?
Yes. Unless configured otherwise, that's the SIP port
Port: 5060
UDP Bindaddress:0.0.0.0
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
In sip.conf bindport = 5060
'Sip show settings' doesn't work in 1.4.22
I don't have access to a '1.4' instance right now, but 'sip show settings'
works in 1.2 and 1.6 so I'm guessing
Thank you very much for the reply.
On 2 Dec 2010 at 17:06, Steve (Steve Edwards asterisk@sedwards.com)
commented about Re: [asterisk-users] Asterisk ports:
On Thu, 2 Dec 2010, Gary Kuznitz wrote:
You get extra points today. I think you found where the problem is. It
found /etc
Hi,
I'm trying to get a softphone configured. In Sip.conf [general] I found an
example
that said I need:
nat=yes
localnet=192.168.xxx.xxx
Is localnet supposed to be a LAN IP or a WAN IP?
Thank you,
Gary
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-- Bandwidth
I have no idea the correct way to configure this software phone.
It's called Express Talk
The Asterisk box is at IP = WanLocation
Software phone is at IP = WanSoftware
They are not on the same LAN.
What I have in Extensions.conf is:
[gary-incomming]
exten = 1001,1,Dial(IAX2/gogh)
exten =
The phone is finally registering. That's great.
I'm trying to understand what all these lines in Extensions.conf are defining.
I can't make heads or tails them. I have been reading the manual
AsteriskManualTheFutureOfTelephony2ndEdition.
I'm currently getting an error when placing a call on
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
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
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
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
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
Thank you for the reply.
On 10 Dec 2010 at 8:53, Jeroen (Jeroen Eeuwes jeroeneeu...@gmail.com)
commented about Re: [asterisk-users] Configuring Softphone:
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
On 10 Dec 2010 at 9:31, Jeroen (Jeroen Eeuwes jeroeneeu...@gmail.com)
commented about Re: [asterisk-users] Configuring Softphone:
Hi Gary,
That is a great suggestion. Yes I did try that. I might be having router
issues with a
SonicWall. I'm working with a port sniffer now to try
=0.0.0.0/0.0.0.0
permit=192.168.1.201/255.255.255.255
All the other local phones here
snip
One WanIP address
Thank you,
Gary Kuznitz
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New to Asterisk
I had lines 3 and 4 and added line 1 and 2 to extensions.conf
exten = 106,1,SetVar(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,2,Monitor(wav,${CALLFILENAME},m)
exten = 106,3,hint,SIP/106
exten = 106,4,Macro(stdexten,106,${HINT})
I received this warning:
WARNING[31463]: pbx.c:1832
I currently have in extensions.conf:
exten = 106,1,Set(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})
exten = 106,n,Monitor(wav,${CALLFILENAME},m)
exten = 106,hint,SIP/106
exten = 106,Macro(stdexten,106,${HINT})
When I called x106 this was logged:
-- Executing [106@voicemenu-custom-4:1]
?
Thanks,
Gary Kuznitz
WPM$LEX5.PM$
Description: Mail message body
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