similar to: Decoding SIP register hack

Hi list , someone on the list has seen this type of connection attempts in asterisk, fail2ban does not stop 2015-01-08 14:59:47] SECURITY[21515] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="1420750787-386840",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:100 at

Hello, fail2ban does not ban offending IP. NOTICE[29784] chan_sip.c: Registration from '"user3"<sip:1005 at asterisk-ip:5060>' failed for 'offending-IP:53417' - Wrong password NOTICE[29784] chan_sip.c: Registration from '"user3"<sip:1005 at asterisk-ip:5060>' failed for ?offending-IP:53911' - Wrong password systemctl status

On 05/17/2018 11:38 AM, Frank Vanoni wrote: > On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote: > >> 3. How do I set up the server to block these ? >> >> 4. Can I stop the retransmitting of the 401 Unauthorized packets ? > > I'm happy with Fail2Ban protecting my Asterisk 13. Here is my > configuration: > > in /etc/asterisk/logger.conf: > >

Just a note that I did a little work to extend FreePBX distro with some extra Fail2Ban which deals with some drive-by SIP registration attempts. My regex is poor to middling, but the steps detailed here: http://www.coochey.net/?p=61 manage to stop IPs which try to authenticate against Asterisk which FreePBX were not able to stop before. I would welcome any improvements anyone would care to

If this is a small site, I recommend you download the free version of SecAst (www.telium.ca <http://www.telium.ca> ) and replace fail2ban. SecAst does NOT use the log file, or regexes, to match etc.instead it talks to Asterisk through the AMI to extract security information. Messing with regexes is a losing battle, and the lag in reading logs can allow an attacker 100+ registration

Hello I'm using the Fail2ban. I configuration below. I want to try to prevent the continuous password. Fail2ban password that does not prevent this form. (Asterisk 1.8 / Elastix interface) What could be the problem ? Asterisk log; "Registration from '<sip:3060 at sip.x.eu;transport=UDP>' failed for 'x.x.x.x:32956' - Wrong password" Fail2ban asterisk

Hello Anyone have a working copy of Fail2ban asterisk filter asterisk.conf for Asterisk 16 running PJSIP. I have tried 10 different filters but none of them show any matches when testing with fail2ban-regex I see date template hits but no matches.... My log [2019-06-06 15:37:20] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" <sip:2405 at

Hi list, I would like to now what is the sense of such type of entry in security.log [2019-09-27 15:12:24] SECURITY[26964] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2019-09-27T15:12:24.181+0200",Severity="Informational",Servic e="PJSIP",EventVersion="1",AccountID="<unknown>",

Hello, How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough variables in (within) my custom Asterisk application? I can't use chan_sip.c internal structures (such as sip_pvt) in my custom application, because there's no chan_sip.h and I can't include it into my application (maybe there's other way?). I can do like this: exten =>

I solved the problem. "action.d/iptables-custom.conf" include only udp. service fail2ban restart Thank you. On Sun, Sep 13, 2015 at 9:17 PM, Andres <andres at telesip.net> wrote: > On 9/13/15 11:16 AM, Gokan Atmaca wrote: >> >> Hello >> >> I'm using the Fail2ban. I configuration below. I want to try to >> prevent the continuous password.

Le 30/09/2019 à 11:45, Joshua C. Colp a écrit : > On Fri, Sep 27, 2019, at 11:31 AM, Administrator TOOTAI wrote: >> Hi list, >> >> I would like to now what is the sense of such type of entry in security.log >> >> [2019-09-27 15:12:24] SECURITY[26964] res_security_log.c: >>

Hello. Voice quality when calling - this is one of the most important in the PBX. You need to record the quality parameters for each call to improve. Because the overall quality of a call can only be determined upon completion, I did it in the HangUp handler and wrote in custom fields of CDR. This worked well in asterisk 11. In asterisk 13 I did not find a handler after the call, but before

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay

I recently upgraded from Asterisk 13.19 to 16.6.1. Everything is working fine with a few minor tweaks except outgoinf fax. Incoming works fine. I do outgoing faxing through an AMI call. Here is the output from the security log: [Nov 27 06:16:05] SECURITY[101222] res_security_log.c:

Hi List, Is there easy way to get klips ipsec stack into centos 6? As it makes firewalling ipsec traffic much easier.. Eero

Hello; Did you remember to uncomment the dateformat in /etc/asterisk/logger.conf? That's necessary for fail2ban to work. Logger.conf [general] dateformat=%F %T Regards; John -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of ricky gutierrez Sent: Thursday, January 08, 2015 4:38 PM To: Asterisk

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I can't read a environment variable in a asterisk dialplan. When logged in as user root on the system an 'echo $HOSTNAME' gives the hostame of the machine. Asterisk (1.4) is started from the same console. I try to read it like this: exten => s,n,NoOp(host=${ENV(HOSTNAME)}) Does anyone know what i am missing? Ipv een saaie e-mail een leuk videobericht? Ga naar

You can use the following: switch3*CLI> show function SIPCHANINFO switch3*CLI> -= Info about function 'SIPCHANINFO' =- [Syntax] SIPCHANINFO(item) [Synopsis] Gets the specified SIP parameter from the current channel [Description] Valid items are: - peerip The IP address of the peer. - recvip The source IP address of the peer. - from