similar to: Disallow CALLS without registry

Got a strange situation [ext-queues] ... exten => h,2,ExecIf($[${CALLERID(num)} = ' ']?Set(var29=${SHELL(curl -X POST --header "Content-Type: application/json" --header "Accept: application/json" -d "{\"Phone\": ${FROMEXTEN}, \"Source\": \"asterisk\"}" "

> On 11/02/2017, at 3:40 am, Frank Vanoni <mailinglist at linuxista.com> wrote: > > On Thu, 2017-02-09 at 14:58 +0200, ????? ?????? wrote: > > >> so the main question is -- how to Disallow CALLS without registering >> on PBX > > sip.conf configuration > In the [general] section, define: > > > [general] > ... > allowguest=no >

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

Hi. I was studying about how to use databases in Asterisk, accessing it from the dial plan. In my project, my dial plan will have to store simple data (ex: IP number, port number, device name, etc) in a persistent way, so that it will be possible to retrieve such information in future moments, still via dial plan. For this case, I would like to know? 1. What is the best choice for storing and

Hello all. I was recently the victim of a SIP flood attack. I'm wondering what is the best method to prevent such things in the future. Many thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101003/2e254523/attachment.htm

... using it as a tool and understanding what it does... So one part of it's toolset identifys valid SIP accounts - and I was under the impression that alwaysauthreject=yes was supposed to stop this... However, it sends a request for a highly probably non-existent account, then sends requests for probably existing accounts and I guess compares the results - account not found vs. bad

Hello, I am trying to make my first call on Asterisk to succeed. I have Asterisk 1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for my first test, Trying to have a call between two X-lite sipphone. The subscribers succeeded to register and the call is established, but still no voice can be heard, and lead the call to be disconnected after! By checking the logs, I can see

Hey, I'm going thru logs, and I see some very common and interesting things that the hackers are looking for. In a whole bunch of scans, I've noticed that the first guess or two for sip accounts is usually a 10-digit number. I'm asking myself, why these numbers? Are they looking for a voip trunk? Or is it just like a serial number for the scan? What? Here's some examples:

Is there a way to drop a ip connection to asterisk after a number of register attempts. I have been having issues with hackers doing registration scanning against our server. We block their address at the fire wall but since asterisk does not force a drop of the connect after so many bad reg attempts I can't enforce the block until they drop and try again. This allows them to run the box

We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were

Hi Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer get the 'no matching peer' error when we get a dictionary SIP attack. Now the logs always show a 'wrong password' when there actually isn't a matching peer. We even have alwaysauthreject = yes in our sip.conf. Has anyone else noticed this phenomenon? Thanks in Advance Ish -- Ishfaq Malik

So when someone's brute forcing your server is there a way to identify the originating IPs without using a tcpdump? When I get a failed auth on the console it shows 'account at asteriskserver' then tag=as25ca5023 (or some random string, though it's a bit odd as alwaysauthreject = yes is on in sip.conf). Anyway, the logs don't show anything more useful either. Is there

Hi List, i've been receiving several sip registration probes in the last month, and as this server is a testing site (no external lines, no nothing) i have no fail2ban and still not planning to install. Whenever i have nagios telling me that there is another 'guest', i go and edit iptables manually and that's it. Recently i discovered that these attacks start with some kind

HI LIST CAN U HELP ME If there are multiple sip trunks with the same ITSP then an incoming call is arbitarily matched to the last peer with the same host IP address. This is not a serious problem because the DID is still correct but it does have many insidious effects due to the incorrect channel name Example register=myaccount1 at sip.myitsp.com/line1 register=myaccount2 at

hi list can U help me caller id in USTM if now working -- Starting switch on '4211 at 4211-1' to 4203 -- Executing [4203 at office:1] DumpChan("USTM/4211 at 4211-0x7f7ba4228fd0", "") in new stack Dumping Info For Channel: USTM/4211 at 4211-0x7f7ba4228fd0: ================================================================================ Info: Name=

What I have is: * Asterisk 1.8.10.1~dfsg-1ubuntu1, * SPA112 ATA with analog fax in 1-st FXS port connected, * SIP trunk with provider supporting T.38. My network looks like this: * spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in neighbouring LANs, * Asterisk connects to the provider (80.75.130.136) via router (82.200.7.184). Router has full DNAT to Asterisk server. What happens?

*friends help me * *cant get incoming calls in asterisk* *(when i connect **80081 in softphone ---every thing is ok**)* *<--- SIP read from UDP:200.152.125.221:5060 <http://200.152.125.221:5060> --->* *INVITE sip:80081 at 10.47.10.10:5060 <http://sip:80081 at 10.47.10.10:5060> SIP/2.0* *Record-Route: <sip:200.152.125.221;lr;ftag=as6872d065>* *Via: SIP/2.0/UDP

Hi all, I want to two sip clients connect through Asterisk in local network for testing. My sip.conf file looks like this [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse localnet=192.168.1.0/255.255.255.0 [7001] type=friend host=dynamic

-- Luca Pradovera luca.pradovera at gmail.com Hello, sorry, I managed to lose the reply amidst the traffic. What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer. Leg B then decides that user C knows more about the subject, and initiates a blind transfer to C?s phone

Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615 at sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1' --