[asterisk-users] getting invites to rtp ports ??

John Covici covici at ccs.covici.com
Wed Aug 29 20:33:57 CDT 2018


OK, Thanks.  I have a couple of questions -- the line numbers do not
match exactly, so can you tell me a couple of lines before and after
the line in question?  Also, when will this be logged, if its only
during sip debug, I need to change it to log when I can see it more
readily.

Thanks.

On Wed, 29 Aug 2018 20:31:15 -0400,
sean darcy wrote:
> 
> On 08/29/2018 08:07 PM, John Covici wrote:
> > I wonder if I could have that patch, maybe I could add it to my
> > fail2ban regexp and if you have the correct regexp, I would apperciate
> > that as well.
> > 
> > Thanks.
> > 
> > On Wed, 29 Aug 2018 19:18:29 -0400,
> > Telium Support Group 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
> >> 
> >> 
> >> -----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
> >>> 
> >>> 
> >>> --
> >>> _____________________________________________________________________
> >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>> 
> >>> Astricon is coming up October 9-11!  Signup is available at:
> >>> https://www.asterisk.org/community/astricon-user-conference
> >>> 
> >>> Check out the new Asterisk community forum at:
> >>> 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
> >> 
> >> 
> >> 
> >> --
> >> _____________________________________________________________________
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> 
> >> Astricon is coming up October 9-11!  Signup is available at: https://www.asterisk.org/community/astricon-user-conference
> >> 
> >> Check out the new Asterisk community forum at: https://community.asterisk.org/
> >> 
> >> New to Asterisk? Start here:
> >>        https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >> 
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>     http://lists.digium.com/mailman/listinfo/asterisk-users
> >> 
> >> 
> >> -- 
> >> _____________________________________________________________________
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >> 
> >> Astricon is coming up October 9-11!  Signup is available at: https://www.asterisk.org/community/astricon-user-conference
> >> 
> >> Check out the new Asterisk community forum at: https://community.asterisk.org/
> >> 
> >> New to Asterisk? Start here:
> >>        https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >> 
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>     http://lists.digium.com/mailman/listinfo/asterisk-users
> >> 
> > 
> The patch, more accurately a hack, is in my second post above.
> 
> chan_sip.c 4127 : 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)));
> 
> The added second %s shows the ip address of the pkt owner.
> 
> I wouldn't submit it in a coding class !
> 
> sean
> 
> 
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> Astricon is coming up October 9-11!  Signup is available at: https://www.asterisk.org/community/astricon-user-conference
> 
> Check out the new Asterisk community forum at: https://community.asterisk.org/
> 
> New to Asterisk? Start here:
>      https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 

-- 
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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