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

Telium Support Group support at telium.ca
Wed Aug 29 10:59:21 CDT 2018

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:


-----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
>     <>:
>     SIP/2.0 401 Unauthorized
>     Via: SIP/2.0/UDP
>     From: <sip:37120116780191250 at
>     <mailto:sip%3A37120116780191250 at>>;tag=1872048972
>     To: <sip:3712011972592181418 at
>     <mailto:sip%3A3712011972592181418 at>>;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", 

With that in the log, I'm now blocking the ip addresses.


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