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:[email protected]] On Behalf 
Of sean darcy
Sent: Wednesday, August 29, 2018 10:46 AM
To: [email protected]
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 <[email protected]
<mailto:[email protected]>> 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:[email protected]
     <mailto:sip%[email protected]>>;tag=1872048972
     To: <sip:[email protected]
     <mailto:sip%[email protected]>>;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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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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_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Astricon is coming up October 9-11!  Signup is available at: 
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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
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