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Securing Asterisk SIP PBX by simple iptables rule checking if the domain is correct

Securing Asterisk SIP PBX by simple iptables rule checking if the domain is correct published on 14 Comments on Securing Asterisk SIP PBX by simple iptables rule checking if the domain is correct


For some time I’ve been looking for a simple way to protect my Asterisk SIP pbx against attacks from bots, scanners which scans and trying dialing to premium numbers. Opening SIP port to the internet causes that there was no one minute without suspect requests hitting my Asterisk. The log was full of that attempts.

While analyzing this problem I noticed that bots, scanners, attackers using everywhere IP address of my server to trying break it. While my proper clients using domain of my Asterisk server. If user is using domain name in his SIP client/phone,this domain is used in further communication on SIP protocol.

Below I will show example of INVITE (INVITE is using to establish VoIP call) SIP request from user using domain name (, and from user using IP address ( of Asterisk server.

With domain:

With IP address:

Knowing that, I want to block requests to Asterisk server which are NOT contains my domain name. In this point I want to clarify that I have special subdomain for telephones. Bots, scanners, attackers are not knowing about this domain.
Blocking unwanted requests can be done by iptables rules with string matching.

Differences between rules result from different approach between TCP and UDP protocols when establishing a connection. TCP need to do three way handshake to establish connection, UDP not doing this.

After applying these rules, I did not see even one attack 🙂
At the and we can check increasing iptables counters:


Hi Jan,
I implemented the rules for SIP over UDP. Almost everything works, but I have problems receiving INVITEs from DID providers (i.e., from servers on which my Asterisk is REGISTER’d). Did you assume that the rule for state “ESTABLISHED,RELATED” would catch them? In my case, it does not.

Hi Enzo,
Could you record INVITE packets from your SIP trunks (DID providers)? You can do it with tcpdump, example:
tcpdump -vvv -i eth0 -p udp and port 5060 -s 0 -A

Probably INVITES packets from SIP trunks, not contains yours domain name. But as you mentioned it should be catch by “ESTABLISHED,RELATED -j ACCEPT” rule.

I suspect that the connections entries in the conntrack table are expiring too fast in your case.
UDP connections are kept for short time in the conntrack table. On my machine with default kernel settings it only 180 seconds:

router:~ # cat /proc/sys/net/netfilter/nf_conntrack_udp_timeout_stream

I think you can solve this problem by configure asterisk to sending keepalives packets in your sip trunks. Option qualify=yes in trunk configuration do that. This should refreshing udp timeout for conntrack entry, connection to yours sip trunk should not expiring from conntrack table and then rule “ESTABLISHED,RELATED” should work properly 🙂

Hi Jan,

Yes, you are right: the “qualify=yes” does the trick. And the nf_conntrack_udp_timeout_stream is 180 also in my system. Pity Asterisk doesn’t allow to send simple keep-alive packets (CRLF/CRLF) as per RFC 6223 section 4.1, so I have to bother the peers with heavier Options requests.

I had provisionally solved the problem by adding ACCEPT rules for the domains referenced in the “From:” header of the INVITEs, but that’s obviously an inferior solution because those domains are relatively well-known, also by hackers who might use them to mount scanning attacks.



I know this is 4 yrs old; does this still works?

If you are using iptables then yes, this still works and and it worked very well on my Asterisk server for years.

A few months ago I move my firewall from iptables to nftables for NAT performance reasons (my server also works as router for my local network, nftables has implemented flow offload mechanism)
and unfortunately on nftables it is impossible to implement rules described in this post.

Thanks Jan,
I see a lot of fail ‘REGISTERS’ in the LOG, but nothing showing up in iptables as DROP, why is this you think? Also these guy have gotten the server IP address, both local and ISP IPs how can this be? Is there anything I can do to prevent this?

Are you sure that the attackers are not using your dns domain to trying register to your server?

If your Asterisk is behind NAT I think you should use externaddr configuration parameter in sip.conf. I suspose that without configured externaddr Asterisk may sending information about its local IP addresses.

Regarding to the not increasing DROP counter – maybe other firewall rule is passing traffic to 5060 port without restrictions?

thousand and thousands of register requests from thousands of IP addresses aware overloading fail2ban, and this put a stop to it. Thank you this simple but ingenious filter helped. I guess I should thing about using a domain name a long time ago.
As to calls from DID providers, I simply added the IPs as ACCEPT.
Thank you again

First, thanks a lot to Jan and all the other contributors for this valuable blog post!

Du you think this block also make sense for SIP over TLS (port 5061)?

If so, how would you configure it for SIP over TLS?

First, thanks a lot to Jan and all the other contributors!

Du you think this block also makes sense for SIP over TLS (port 5061)?

If so, how would you configure it for SIP over TLS?

Hi Martin,
Rules posted above will not work, because iptables will se encrypted traffic.
I think there is two options to try in case of sip over tls.
1) Maybe module will be usefully to check if client is using your sip domain. In case of TLS traffic you can check if client is using your domain by check SSL SNI (Server Name Indication) or rather block connection attempt which contains your server IP addr instead of your SIP domain.
I would try something like that:
iptables -A INPUT -p tcp –dport 5061 -m tls ! –tls-host “your.secret.sip.domainl” -j DROP
iptables -A INPUT -p tcp –dport 5061 -j ACCEPT
similar case is described here:

2) Another option is to use loadbalancer like HAproxy in front of your pbx. SSL in this case will be terminated on that loadbalancer. Traffic between loadbalancer and pbx will not encrypted and rules from my post should work 🙂

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