[asterisk-users] getting invites to rtp ports ??
John Covici
covici at ccs.covici.com
Thu Aug 30 01:02:34 CDT 2018
I agree, but is it possible to try over and over with anything other
than the challenge warning in the security log as sean suggested and
put a patch for?
On Wed, 29 Aug 2018 22:52:05 -0400,
Matthew Jordan wrote:
>
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> On Wed, Aug 29, 2018 at 6:20 PM Telium Support Group <support at telium.ca> wrote:
>
> Depending on log trolling (Asterisk security log) misses a lot, and also depends on the SIP/PJSIP folks to not change message structure (which has already happened numerous time). If you are comfortable hacking chan_sip.c you may
> prefer to get the same messages from the AMI. It still misses a lot but that approach is better than nothing.
>
> Digium warns not to use fail2ban / log trolling as a security system: http://forums.asterisk.org/viewtopic.php?p=159984
>
> That's some pretty old advice.
>
> The rationale for *not* using general log messages with fail2ban still stands: the general WARNING/NOTICE/etc. log messages are subject to change between versions, and no one wants that to impact someone's security. So you should not use
> those messages as input into fail2ban.
>
> That rationale did lead to the 'security' event type in log messages. Security Event Logging - as it is called - got added into Asterisk quite some time ago. So long ago I'm really not sure which version. At a minimum, Asterisk 11, but
> I'm pretty sure it was in 10 as well.
>
> Documentation for it can be found here:
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger
>
> And here:
>
> https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration
>
> Note that this also fires off AMI events (and ARI events, IIRC).
>
> If, for whatever reason, you do not get a SECURITY log message or a corresponding event when something 'bad' happens, that would be worth some additional discussion. If anything, the events can be a bit chatty...
>
>
> -----Original Message-----
> From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of sean darcy
> Sent: Wednesday, August 29, 2018 6:33 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] getting invites to rtp ports ??
>
> On 08/29/2018 11:59 AM, Telium Support Group wrote:
> > Block a single IP is the wrong approach (whack-a-mole). You should consider a more comprehensive approach to securing your VoIP environment. Have a look at this wiki:
> >
> > https://www.voip-info.org/asterisk-security/
> >
> >
> >
> > -----Original Message-----
> > From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com]
> > On Behalf Of sean darcy
> > Sent: Wednesday, August 29, 2018 10:46 AM
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [asterisk-users] getting invites to rtp ports ??
> >
> > On 08/29/2018 09:42 AM, Carlos Rojas wrote:
> >> Hi
> >>
> >> Probably somebody is trying to hack your system, you should block
> >> that ip on your firewall.
> >>
> >> Regards
> >>
> >> On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <seandarcy2 at gmail.com
> >> <mailto:seandarcy2 at gmail.com>> wrote:
> >>
> >> I'm getting invites to very high ports every 30 seconds from a
> >> particular ip address:
> >>
> >> Retransmitting #10 (NAT) to 5.199.133.128:52734
> >> <http://5.199.133.128:52734>:
> >> SIP/2.0 401 Unauthorized
> >> Via: SIP/2.0/UDP
> >> 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734
> >> From: <sip:37120116780191250 at 67.80.191.250
> >> <mailto:sip%3A37120116780191250 at 67.80.191.250>>;tag=1872048972
> >> To: <sip:3712011972592181418 at 67.80.191.250
> >> <mailto:sip%3A3712011972592181418 at 67.80.191.250>>;tag=as3a52e748
> >> Call-ID: 1504207870-295758084-609228182
> >> CSeq: 1 INVITE
> >> .......
> >> WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on
> >> 1504207870-295758084-609228182...
> >>
> >> I thought invites had to go to port 5060 or so. I don't understand
> >> why somebody (let's assume a bad guy) is trying ports above 50000.
> >>
> >> sean
> >>
> >>
> >
> > Ok, so the high port is not the destination port but the source port.
> >
> > So I hacked the log warning in chan_sip.c on non-critical invites to show the source ip:
> >
> > ast_log(LOG_WARNING, "Timeout on %s non-critic invite trans from
> > %s.\n",
> > pkt->owner->callid,ast_sockaddr_stringify(sip_real_dst(pkt->owner)));
> >
> > With that in the log, I'm now blocking the ip addresses.
> >
> > Thanks,
> > sean
> >
> >
> > --
> > _____________________________________________________________________
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> > https://community.asterisk.org/
> >
>
> I agree. That's why I hacked chan_sip.c to get the addresses in the log.
>
> I'm surprised they're not in the log by default. I must be the only person who gets these "non-critical invites".
>
> sean
>
> --
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> --
> Matthew Jordan
> Digium, Inc. | CTO
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
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>
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>
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--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici wb2una
covici at ccs.covici.com
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