similar to: Am I cracked?

For such cases i created a dialplan in the default dialplan which blocks the ip of the hacker with iptables. On Monday, June 8, 2015, Luca Bertoncello <lucabert at lucabert.de> wrote: > Hi list! > > Very strange... > I ran the Asterisk CLI for other tasks, and suddenly I got this message: > > == Using SIP RTP CoS mark 5 > -- Executing [000972592603325 at

I'm guessing this is a small/home system? I suggest you install SecAst from this site: www.telium.ca It's free for small office / home office and will deal with these types of attacks and more. It can also block users based on their Geographic location (based on the phone number it attempted to dial I suspect this is middle east), look for suspicious dialing patterns, etc. If you

> Very strange... > I ran the Asterisk CLI for other tasks, and suddenly I got this message: > > == Using SIP RTP CoS mark 5 > -- Executing [000972592603325 at default:1] Verbose("SIP/192.168. > 20.120-0000002a", "2,PROXY Call from 0123456 to 000972592603325") innew stack > == PROXY Call from 0123456 to 000972592603325 > -- Executing

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > Make sure you have solved the problem. You don't want to get hit with a > phone bill for calls from your location to Israel. Basically, they are > hoping that you are running the equivalent of a mail server open relay. > They are trying to use you to dial out to another number. You don't want > to pay

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > Based on SIP packets coming in from IP addresses you don't recognize, > while you may not be hacked, you would seem to have people probing your I think, too, it's someone probing my IP... > system. One thing you can do at the firewall level is restrict inbound sip > communications to only those from your

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

Hi list! My Problem: all calls to international numbers will be dropped after exactly 15 minutes... I have a VoIP-account by Deutsche Telekom. This is what I see when I call someone (my parents) and the connection will be dropped: == Using SIP RTP CoS mark 5 -- Executing [+39015222222 at default:1] Set("SIP/00493511111111-00000125", "newNumber=0039015222222") in new

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

> OK, I set alwaysauthreject = yes and I discovered a allowguest, which I set > to "no", too. > The PBX is behind a Firewall and I just allow UDP 5060 and 10000-10100. > Now I log the SIP-pakets coming from Internet, too... > > Hopefully I solved my problem... Make sure you have solved the problem. You don't want to get hit with a phone bill for calls from your

> > Make sure you have solved the problem. You don't want to get hit with a > > phone bill for calls from your location to Israel. Basically, they are > > hoping that you are running the equivalent of a mail server open relay. > > They are trying to use you to dial out to another number. You don't want > > to pay for these calls. > > Of course,

Hi again, list! I know, I'm really annoying the list... :) Well, maybe I got my Asterisk at home ("wrt" on the previous E-Mails) to accept my mobile phone from Internet. It was a problem with the network and the firewall. Now I can log my mobile phone in my Asterisk in and the phone is REACHABLE. Wow! Got it! If I call a phone at home using my cellphone it works and the

2015-06-08 22:35 GMT+02:00 D'Arcy J.M. Cain <darcy at vex.net>: > On Mon, 8 Jun 2015 22:24:33 +0200 > Luca Bertoncello <lucabert at lucabert.de> wrote: > > Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > > > Basically, they are hoping that you are running the equivalent of a > > > mail server open relay. They are trying to use you

On Wednesday 10 Jun 2015, Luca Bertoncello wrote: > I'm very sorry to write that, but these answers are really NOT helpful... > I searched two days long how can I check it and didn't found anything > useful... > > Could someone suggest me a way to check if my Asterisk is an "Open > Relay" that accept connections from every peer? Someone on this list is bound

Am 23.06.2020 09:28, schrieb Marek Greško: Hi > if you need clampmss then it is highly probable there is a PMTU > discovery problem. The clampmss does not work for UDP. Is there a way to check if I have this problem? > I probably counted the size incorrectly. So you are able to ping with > size 1464 and not with 1466. How about trying same ping sizes from the > internet towards

Zitat von Remko Lodder <remko at freebsd.org>: Hi Remko, > Emails can only be read if they are authenticated / authorized in > someway to access the store. That means you might need to share the > info@ credentials with the other > people so that they can read it over imap or webmail etc. That is self-evident and it is not a problem. I can't understand what you

Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb: > What settings have you got for directmedia? > > Could you try > > nat=force_rport,comedia > directmedia=no Tried. Peer always unreachable, call not possible... :( Other idea? Thanks Luca Bertoncello (lucabert at lucabert.de)

Zitat von Israel Gottlieb <isrlgb at gmail.com>: > If you the c option in the dial command it will send answered > else where sip message to the phone and most ip phones understand that > The cell will always display a missed call? I'm very sorry, but I can't understand what you mean... Could you explain, maybe with an example? Thanks Luca Bertoncello (lucabert at

Am 23.06.2020 10:07, schrieb Marek Greško: Hi > this is a correct response: > > From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set > (mtu = 1492) > > So PMTU discovery is working. No problem here. You got correct message > to lower the packet size from 62.156.246.57. This is probably the last > hop before your site. No, the last hop is 62.156.246.65: