Displaying 20 results from an estimated 30000 matches similar to: "Detecting DoS attacks via SIP"
2017 Aug 17
3
Detecting DoS attacks via SIP
Well, correct me if I'm wrong, but I would say this conversation you have
posted is a bit outdated, now fail2ban can be used with asterisk security
log
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger.
On Thu, Aug 17, 2017, 4:53 AM Telium Technical Support <support at telium.ca>
wrote:
> Keep in mind that the attacks you are seeing in the log are ONLY the
2012 Nov 16
1
Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Hello,
After Upgrade to Asterisk 11.1.0-rc1 I keep getting
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [603 at DLPN_AlDimnaDialPlan:601]
Dial("SIP/601-00000002", "SIP/603") in new stack
[Nov 16 06:42:33] WARNING[15547][C-00000004]: app_dial.c:2433
dial_exec_full: Unable to
2015 Mar 11
2
PJSIP some AMI events is absent?
Hello.
Asterisk 13.2, PJSIP.
Problem: I do not get any AMI events when changing the status of the
contact.
When using chan_sip I got "peerstatus" event.
When using res_pjsip and devices (endpoint configuration) I got
"peerstatus" event.
When using res_pjsip.so and OUTBOUND REGISTRATION WITH OUTBOUND
AUTHENTICATION i got "registry" event.
When using
2010 Jan 29
1
callerid not working over sip
Calling from my home using Asterisk 1.6.2.1 to an office extension
(Asterisk 1.6.1.13) the callerid is not honored:
Home:
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [170 at internal:1] Answer("DAHDI/1-1", "") in new stack
-- Executing [170 at internal:2] NoOp("DAHDI/1-1", "Context:
office-extensions") in new stack
2009 Oct 08
2
Server-side scripting when SIP phones register
Hi,
Some IP Phones (Aastra) are able to send a custom HTTP request just after
registration completion.
Using this, it is possible to update phone's screen with messages like "Do
Not Disturb" or "Forwarded To VM".
RFC 3680 (http://www.faqs.org/rfcs/rfc3680.html) provides a mecanism to
support these interactions.
To my knowledge, this RFC is not implemented yet in
2009 Nov 22
1
transferring SIP call: no voice
I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk
B. Both are behind NAT, but port forwarded. I get the connection, but no
voice - either in or out.
I can call on SIP from A to B (and from B to A). Do it all the time.
Asterisk A receives SIP calls from Junction and Teliax.
CLI on A looks right:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
==
2004 Feb 06
2
IPFIREWALL_DEFAULT_TO_ACCEPT becomes default to deny
Hey Guys,
today I upgraded to 4.8-RELEASE-p15. As usual I set IPFIREWALL to default
accept in my kernel config file.
Config & make weren't complaining so, installed the kernel, reboot and there
it was:
>IP packet filtering initialized, divert disabled, rule-based forwarding
enabled, default to deny, logging disabled
Another rebuild didn't work out so... I reviewed
2011 Jun 20
1
Problems with pickupgroup/callgroup with Asterisk 1.8.4.2
I have problems using the call pickup under Asterisk 1.8.4.2. I have
another Asterisk with 1.6 - and it is working fine with the same settings.
I have setup the same callgroup and pickupgroup for all extensions in
sip.conf - just to make things simple for testing. The sequence *8 seems
to be completely ignored by Asterisk - the client shows "Call answered"
when dialing *8 while the
2008 Jan 30
5
One approach to dealing with SSH brute force attacks.
Message-ID: <479F2A63.2070408 at centos.org>
On: Tue, 29 Jan 2008 07:30:11 -0600, Johnny Hughes <johnny at centos.org>
Subject Was: [CentOS] Unknown rootkit causes compromised servers
>
> SOME of the script kiddies check higher ports for SSH *_BUT_* I only see
> 4% of the brute force attempts to login on ports other than 22.
>
> I would say that dropping brute force
2010 Jun 11
7
How to stop intruder from registering sip?
This is a small 12 line system, internal extensions 150 - 180. I didn't
have a phone on 151. Here's the sip.conf stanza:
;;[151]
;;type=friend
;;context=longdistance
;;callerid="Conf Room" <151>
;;secret=0000
;;host=dynamic
;;qualify=yes
;;dtmfmode=rfc2833
;;allow=all
;;defaultuser=151
;;nat=yes
;;canreinvite=no
There's no DISA. And then somehow (how???) ip address
2012 Mar 09
2
dreaded one-way audio with nat=yes
I'm trying to move the asterisk server to an Amazon Web instance. We
have teliax for our sip provider. I'd like for our DID lines to be
connected to a users cell phone.
Seems simple enough, but I'm getting the dreaded one-way audio, even
with nat=yes everyplace I can think of.
The dialplan is real easy:
[from-teliax-sip]
exten => _j.,1,NoOp("From teliax sip with exten
2011 Apr 18
2
Registrations stops after 403 FORBIDDEN
Hello list,
I have in sip.conf :
/maxexpiry=60 ; Maximum allowed time of incoming
registrations
; and subscriptions (seconds)
minexpiry=60 ; Minimum length of
registrations/subscriptions (default 60)
defaultexpiry=120 ; Default length of incoming/outgoing
registration
;-----------------------------------------
2009 Jul 01
2
Multi-tenant parking broken in 1.6.1.1?
Hello, all. With the assistance of very helpful folks, our brand new
multi-tenant setup seems to be working smoothly from start to finish
with just a bump or two. The biggest is parking. Now that we got most
kinks worked out, I'm a little more comfortable in trying to resolve
this.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
2015 Mar 26
1
CDR dst value null after attended transfer
I'm having an issue with CDR. Basically, I expect to have all "legs" of a
call having the same linkedid and differing only by the sequence value.
That does happen, but I'm getting null dst values after doing an attended
transfer.
I'm not sure if this is a bug or I'm doing something wrong. I'm running
Asterisk 13.2.0.
Here's the console log, step by step:
First,
2012 Jan 02
1
tcp version of toronto - osaka doesn't work
I'm trying to setup a simple tcp sip connection based on the toronto
osaka example in the Asterisk book.
On the remote box (osaka) (1.8.9.0-rc1):
[toronto]
type=friend
transport=tcp
secret=welcome
context=toronto_incoming
host=dynamic
disallow=all
allow=ulaw
sip show peer toronto
* Name : toronto
Secret : <Set>
MD5Secret : <Not set>
Remote Secret:
2013 Sep 28
1
iax: unable to transfer - one way audio
We have zoiper connected over iax to asterisk in Sydney. The call is to
asterisk in New York. The caller in NZ can hear clearly. Nothing in NY.
Here's the sydney server:
-- Accepting AUTHENTICATED call from <zoiperipaddr>:
> requested format = speex,
> requested prefs = (),
> actual format = ulaw,
> host prefs = (silk16|ulaw|gsm|g722),
2015 Mar 05
4
json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
Hello!
Just installed asterisk 13.2.0 and see many such messages in log, I see
them in console during calls, really something like this:
-- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6",
"SIP/6166 at asterisk") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/6166 at asterisk
> 0x7fa9d4007660 --
2008 Jul 21
20
Ideas for stopping ssh brute force attacks
just wanted to get some feedback from the community. Over the last few
days I have noticed my web server and email box have attempted to ssh'd to
using weird names like admin,appuser,nobody,etc.... None of these are
valid users. I know that I can block sshd all together with iptables but
that will not work for us. I did a little research on google and found
programs like sshguard and
2015 Mar 20
2
outbound calls
i noticed that when i active the voicemail in the IP-phone where the number
0033149xxxxxx is configured i can call this number without issue
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149xxxxxx == Begin MixMonitor Recording
SIP/101-0000010d
-- SIP/FD-0000010e is making progress passing it to SIP/101-0000010d
> 0x2b393cfc2610 -- Probation passed
2010 Aug 02
5
Asterisk and TV media server
Hello,
I would like to know whether there is a way to associate a TV media server
with Asterisk. Is it possible to access TV Chanels in the Telephone Sets.
Anybody have any tips or documents related to this please let me know.
Thanks
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