[SIP scanners] Spam calls

[al.mcne]

Hello
During the last 2 weeks, I started receiving spam calls to my freephoneline home line. Calls are coming every 2-3 minutes from "different" numbers. It can start from 000, call twice or 3 times, "change" the number to 001, etc. When the number reached 20+ digits, it started again from "low" numbers. From the beginning, I tried to put these calls on the blacklist in my phone, but it didn't help a lot: I still have incoming calls/phone ring continuously.
https://www.freephoneline.ca/doGetCallLogs doesn't show these calls. It shows the same other calls, but I have absolutely no way to divide real calls from spam.

What options do we have?
How can I stop these spam calls?

Thank you

[Liptonbrisk]

They're not calls. They're sip scanners (crackers/script kiddies) port scanning you. I'm pretty tired and trying to find posts I've written about this before that don't have dead links to forums that no longer exist (dslreports, for example) . . . I've posted so much on SIP scanners on these forums in the past.

Related links
https://support.intermedia.com/app/arti ... ious-calls
https://blog.kolmisoft.com/sip-attack-friendly-scanner/
etc.

[Liptonbrisk]

If calls or ringing correspond with different numbers (that clearly aren't normal phone numbers) on your call display, such as 100, 1000, 1001, 9999, etc., those aren't calls and won't ever be listed in FPL call logs after logging in at https://www.freephoneline.ca/callLogs. If you pick up, you won't hear anything. They're SIP scanners. SIP scanners are programs written by crackers (script kiddies). They look for ways to break into your home network by scanning for open ports. Typically, they'll scan for common service ports to see if they're open, such as UDP 5060, 5061 and a few others (some scan for a lot more than that). If a port is open, they can access your ATA (and, potentially, other devices on your LAN). Your phones will ring with caller IDs appearing as 1001, 999, something else that's clearly not a regular phone number, etc. These crackers will try to make free calls using your services (or much worse). That's why it's important to have a good NAT firewall in a router protecting your ATA, computers, and other devices on your LAN (local area network) and to never port forward or use DMZ. Port forwarding is a security risk. Enabling DMZ is even worse. Refer to point A below.


---
(Generic info)

Typically, for SIP services, especially for Freephoneline/Fongo, you want

A) a router that does not have a full cone NAT,

Visit https://www.ietf.org/rfc/rfc3489.txt (scroll down to "NAT Variations").
Mango from the Obitalk.com forums writes,
“Use a restricted cone NAT router, and do not use port forwarding or DMZ. Restricted cone NAT will only permit
inbound traffic from the service provider you're registered to. If you have a full cone NAT router, it will allow traffic
from any source. This is probably not what you intend.
If you have a Windows computer, you can test your router using the utility here:
https://web.archive.org/web/20241001215 ... k,22292023. To run it, use stun stun.ekiga.net from a command prompt.”
Essentially, you download the stun-test.zip file; extract the stun.exe file from within the zip file to an easily
accessible location; use an elevated command prompt (visit
http://www.thewindowsclub.com/how-to-ru ... inistrator); change directory (cd) to the
directory or location where you extracted stun.exe (visit
http://www.digitalcitizen.life/command- ... c-commands); and type “stun stun.ekiga.net” without
the quotation marks followed by the enter/return button on your keyboard.
Asus routers, at the time of this writing, produce port restricted cone NAT routers, for example and are fine,
provided you’re using one with Asuswrt-Merlin, third party firmware installed.

B) a router that lets you disable SIP ALG if it's buggy,

To understand why SIP ALG often causes horrible problems, please visit
https://www.voip-info.org/routers-sip-alg/ (scroll down to the section on SIP ALG problems).

If you're dealing with a modem/router combo issued by an ISP or a router with SIP ALG forced on, you may have
to use voip4.freephoneline.ca:6060 for the Proxy Server. The purpose of voip4.freephoneline.ca:6060 is to circumvent
faulty SIP ALG features in routers.

Also, scroll all the way down to the final post in this thread.



C) a router that allows you to set QoS or assign highest priority to your ATA or IP Phone over all other devices on your LAN (local area network),

For a very general description of what QoS can do for you, visit https://www.voipmechanic.com/qos-for-voip.htm.
The basic idea is if you're torrenting or have a bunch of other computers, smartphones, tablets, etc. downloading and uploading (hogging all your available bandwidth), you don't want
your ATA not to have access to enough bandwidth to make or receive calls properly. So QoS or a Bandwidth Monitor feature (which is just another form of QoS) is a really good idea for VoIP users.

I often get an occasional relative complaining to me, "Hey my calls sound choppy." And then when I go visit, some kids are playing MMOs on a computer, while another person is downloading a huge file,
and another person is backing up files to a cloud service all at the same time someone else is trying to talk on the phone. All those devices, without QoS enabled, are fighting over available bandwidth along with the ATA.

and D) A router that lets you adjust both Unreplied and Assured UDP timeouts.

Thanks to Mango, many of us now understand that in order for ATAs to remain registered and working properly with a SIP services like Freephoneline, in particular after power failures, the following conditions must be met:

UDP Unreplied Timeout (in your router) < NAT Keep-alive Interval (in your ATA; for Obihai ATAs this is X_KeepAliveExpires; for Grandstream, the setting is SIP Notify Keep Alive Interval) < UDP Assured Timeout or UDP Stream (in your router) < SIP Registration Failure Retry Wait Time (or RegisterRetryInterval in Obihai ATAs)

“<“ means less than.

When a modem leases a new IP address, a problem can arise where prior associations using the old IP address are maintained in the router. When the ATA attempts to communicate using the old IP address, the response is unreplied, and then if the UDP Unreplied timeout is greater than the Keep Alive Interval (and UDP Unreplied timeout is often set to 30 by default in consumer routers) a problem arises where the corrupted connection persists. If UDP Unreplied timeout is, for example, 15, and the NAT Keep Alive Interval is 20, then the corrupted connection will timeout or close. A new connection will be created, and everything will work fine.

Another problem can occur when the Keep-Alive interval is greater than UDP Assured Timeout (often 180 by default in consumer routers): the NAT hole will close due to the ATA not communicating frequently enough with the SIP server. In turn, incoming calls may, intermittently, not reach the ATA. Again, X_KeepAlivesExpires (SIP Notify Keep Alive Interval) is supposed to be 20 with FPL.



Getting access to both UDP Unreplied Timeout and UDP Assured Timeout settings in consumer routers may be difficult, if not impossible. Asuswrt-Merlin (I would avoid any model below/less powerful than an RT-AX86U at this time), third party firmware for Asus routers, does offer easy access to these two settings, which are found under Advanced Settings-->Administration-->Tweaks. My understanding is that third party Tomato firmware has these two settings as well. So if your router supports Tomato firmware, that may be another option. Note that I will not be held accountable any damage resulting from failed firmware updates. Apparently, Mikrotik routers also allow users to change both Assured and Unreplied UDP timeout settings as well: https://forums.redflagdeals.com/recomme ... #p28059363.

Router firmware that allows users to adjust Assured and Unreplied UDP timeouts include

Asuswrt-Merlin
Ubiquiti
Mikrotik
pfSense
Tomato
DD-WRT
OpenWRT

The keep alive interval for FPL is 20. The SIP Registration Failure Retry Wait Time is 120. I use 15 for UDP Unreplied Timeout and 115 for UDP Assured Timeout.



ISPs do not issue customers routers that can do all four things I just listed. Typically it's far better to have your own router with strong QoS functions and a restricted cone NAT firewall,
disable whatever SIP ALG feature is enabled in the router, and stick whatever modem/router combo your ISP gives you into bridge mode.

[Liptonbrisk]

(Link to the above modified post is located at viewtopic.php?p=80257#p80257.)


Based on your forum profile, you're using a PAP2 ATA.

1. Disable DMZ and all port forwarding in your router. Port forwarding is a security risk. Only port forward if you have no other choice.

2. a) Dial ****
b) Then dial 110#
c) Enter the IP address you hear into a web browser.
d) Login to your ATA.
e) Always choose the admin login and advanced view menus (select "advanced" in the upper right).


3. In your PAP2, navigate to Line 1 (or whatever you're using for FPL)-->SIP settings, and change SIP Port to a random number between 30000 and 60000 (UDP port 56256, for example). Do this for security reasons (to help avoid SIP Scanners/hackers). Also, this step may help to temporarily address a corrupted NAT association that's developed between a router and ATA (if you ever have registration issues, try selecting a new random port number in this range, and then reboot the ATA).

Never use UDP 5060 for local SIP Port, and don't use the same UDP port number for the SIP port on any other Line.



Anyway, the best solution is to use a good firewall.