Asterisk 14 Constant chan_sip handle request rejection

Hello,

I am very new to using Asterisk. I recently installed Asterisk 14 on Centos7. I have configured my sip.conf and my extensions.conf to work with my the prodiver I’ve gotten for a SIP Trunk. So far everything is working fine. I can make calls and receive calls. I don’t have FreePBX or any other GUI installed. I started watching videos on Asterisk Security, because I wanted to educate myself on anything I may need to do to keep my system from being compromised, as well as good practices.

My question is. While I am logged into the Asterisk CLI. using asterisk -rv I keep seeing some random traffic I am not sure is suppose to be there or not. It is not flooding, but it’s enough to make me worried that maybe I am doing something wrong.

Here is the paste from Asterisks cli

[Mar 29 11:42:49] NOTICE[18702][C-000001d2]: chan_sip.c:26422 handle_request_invite: Call from '' (92.114.32.24:5070) to extension '011972595144330' rejected because extension not found in context 'default'. [Mar 29 11:43:25] NOTICE[18702][C-000001d3]: chan_sip.c:26422 handle_request_invite: Call from '' (92.114.32.24:5070) to extension '9011972595144330' rejected because extension not found in context 'default'. [Mar 29 11:44:03] NOTICE[18702][C-000001d4]: chan_sip.c:26422 handle_request_invite: Call from '' (92.114.32.24:5075) to extension '011972595144330' rejected because extension not found in context 'default'. [Mar 29 11:44:30] NOTICE[18702][C-000001d5]: chan_sip.c:26422 handle_request_invite: Call from '' (107.150.62.162:5070) to extension '0810046322648764' rejected because extension not found in context 'default'. [Mar 29 11:44:40] NOTICE[18702][C-000001d6]: chan_sip.c:26422 handle_request_invite: Call from '' (92.114.32.24:5071) to extension '9011972595144330' rejected because extension not found in context 'default'. [Mar 29 11:45:02] WARNING[18702]: chan_sip.c:4077 retrans_pkt: Retransmission timeout reached on transmission cdcddae5d3ebf1dce977eb90a7d61508 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 31998ms with no response [Mar 29 11:45:18] NOTICE[18702][C-000001d7]: chan_sip.c:26422 handle_request_invite: Call from '' (92.114.32.24:5078) to extension '011972595144330' rejected because extension not found in context 'default'. [Mar 29 11:45:56] NOTICE[18702][C-000001d8]: chan_sip.c:26422 handle_request_invite: Call from '' (92.114.32.24:5078) to extension '9011972595144330' rejected because extension not found in context 'default'. [Mar 29 11:46:33] NOTICE[18702][C-000001d9]: chan_sip.c:26422 handle_request_invite: Call from '' (92.114.32.24:5071) to extension '011972595144330' rejected because extension not found in context 'default'. [Mar 29 11:46:52] NOTICE[18702][C-000001da]: chan_sip.c:26422 handle_request_invite: Call from '' (107.150.62.162:5070) to extension '820046322648764' rejected because extension not found in context 'default'. [Mar 29 11:47:15] NOTICE[18702][C-000001db]: chan_sip.c:26422 handle_request_invite: Call from '' (92.114.32.24:5071) to extension '9011972595144330' rejected because extension not found in context 'default'.

I am not sure if there is more I should post to help figure out this issue. Any advice would be wonderful

Thank you

Those are hack attempts if the IP 92.114.32.24 the connections come from is not yours.

I suggest to configure fail2ban to block such attempts. Ideally you should have iptables setup on your system to white-list/black-list traffic.

You should also go through this, http://blogs.digium.com/2009/03/28/sip-security/

–Satish Barot

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Start by disabling unauthenticated calls allowguest=no

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Thank you for the replies, I configured fail2ban and the traffic has died down a lot.

Isn’t regardless to what kind of security have in your Asterisk at all, it seems on your router you have open ports.