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

Telium Support Group support at telium.ca
Wed Aug 29 18:18:29 CDT 2018


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
> 
> 
> --
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> 
> 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



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