How to screen out robo-calls & solicitors on home VOIP

Asterisk Asterisk General
1

Hi - I’m new to this community and only moderately technical

  • so apologies offered in advance for my 1st post.

I would like to find a solution to screen out robo-calls and solicitors on my home telephone.

My home phone number is registered to Flowroute.com, caller ID is enabled.
That VOIP provider sends a signal to my Linksys SipuraSPA-1001 which rings on my home phone.

    • Note: Linksys specs: Software version 3.1.19(SE); Hardware version 3.0.0(90b3)
      I use a simple VTech 5-handset answering machine in the house
      Both the linksys and VTech are maybe 5 years old [if that matters?]

I want to see if you experts can please help me develop a solution to screen out SPAM and Robo-Calls and calls I want to BLOCK. I envision the system to work like this:

10 Incoming VOIP call to SIP to the NEW asterisk equipment
20 NEW hardware answers "Hi, Press 1 if you are a human … "
30 If nothing after 3 seconds - disconnect
40 If 1 then allow call to ring through to home

Now some crank callers are human … so I’d envision some sort of software having a BLOCK list that I could update manually. If that is possible, then the logic statements would be renumbered as …

15 * If Caller-ID is on BLOCK list then disconnect
20 * If not on BLOCK list then NEW Hardware answers "Press 1 if you are a human …*

50 * If home handset presses, for example, the # tone, the add that called ID to the BLOCK list.
( or if not automatic, then allow me to manually update the BLOCK list from my PC)

Is that something Asterisk and some hardware can perform?
If not, can anyone direct me to the right product forum?

Hey thanks for taking the time to read my first post.
With anticipation … Jim [near St. Louis MO]

2

Yes you can easily do this with asterisk.

You will need to create two PJSIP endpoints.

https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip

You will configure your ATA to connect to your asterisk server using one endpoint.

You will configure your asterisk server to register with flowroute using the other endpoint

You will need to write dialplan for both endpoints

You can use the BLACKLIST function to check if incoming callers are blacklisted

https://wiki.asterisk.org/wiki/display/AST/Function_BLACKLIST

3

First John - a huge THANKS for the quick reply in the affirmative.
Sorry but I’m unfamiliar with the terminology.
I’ve done some searching, please tell me if I’m on the right track.

PJSIP endpoints … found this description of PJSIP
… this appears to be a CONFIG file in the Asterisk program - right?
… I’d need one to replace my Linksys, right ?
… and a second PJSIP for my dial-out capability, right?

ATA … [ok after search, that’s a PC card right? … which are recommended ?]
… i could not find any posting on wiki.asterisk.org that recommends the ATA

Asterisk Server … [according to AsteriskWin32 is the Asterisk software running on the ATA / PC card right ?

Write a dialplan … I’m sure there is a posting for that

So what is my next step John?
I’d welcome a chance to chat with you or skype. Is that permitted by this community?

Thanks, Jim

4

Your Sipura is an ATA and hopefully you would be able to use it to connect to your asterisk box assuming the Linksys hasn’t been locked to only work with Fonality.

Asterisk runs under Linux and other Unix like operating systems, There are no supported versions that run on windows.

5

Whist Asterisk has some features specifically designed for fighting off telesales people, there is a steep learning curve for using Asterisk, and you really need a good understanding of how to make Asterisk handle normal calls before it is really possible to help you with this application on anything but a paid consultancy basis.

If you were using Asterisk in a home environment, for this purpose, you would most likely want to use a Raspberry Pi as your hardware.

However, if time is more important than money, even pricing time at the domestic cost of time, you should probably look into products like TrueCall (although I can’t say for certain that it will work behind your ATA.

Incidentally, in the UK, it may be unwise to reject calls from “solicitors” as they are the branch of the legal profession that makes first contact with the general public.

6

John: Thanks for the reply.

According to www.AsteriskWin32.com the Asterisk software runs on a Win32 PC. ?? Per your 2nd sentence - Do you disagree?

My Setup: Cable Modem to a Switch to my [RJ45] SIPURA / ATA out [RJ11] to the VTech base unit. To implement Asterisk would the SIPURA output RJ11 then go into a PC/RaspPi running Asterisk?
If No - please explain
If Yes - what card type do I search for ? The only RJ11 Phone-In / Phone-Out is an internal Fax Modem PC card.

I’d appreciate your input - Thanks, Jim

7

What he said was:

The Windows port is based on Asterisk 1.2, support for which ceased almost a decade ago (9 and a half years). The Windows port was probably never officially supported.

8

Thanks David for the quick reply. I have 3 more questions:
A) can the PC based Asterisk 1.2 provide the functionality I spec in my initial post?
I have a spare PC for zero cost to try.

B) TrueCall’s functionality is PERFECT for me. However on their HELP pages they say
"We do not recommend trueCall for VoIP lines" … well nuts ! I sent them an email.
I wonder if there is a USA version of that terrific product?

My Home Phone Setup:
Cable Modem Cat6 to a Switch Cat6 to my [RJ45] SIPURA / ATA out [RJ11] to the VTech base unit.

C) To implement Asterisk would the SIPURA output RJ11 then go into a RaspPi running Asterisk?
If No - please explain what I’m missing
If Yes - what hardware do I search for ?
–I couldn’t find any RJ11 expansion boards on -
https://www.raspberrypi.org/products/ or https://www.adafruit.com/

Thanks for being so patient with me John & David.
Regards, Jim

9

I haven’t used a version of Asterisk earlier than 1.4.

All versions of Asterisk run on PCs.