Block calls based on NAME display

[xts431]

Hi all,

I'm attempting to block an extremely annoying telemarketer, who calls from various area codes, every day. Same company. There's no point in blocking the numbers as they're never the same and they're spoofed/invalid.

One thing which all of the calls have in common is the name display part of the caller ID uses the following strings:

V22400435500020
V13120413700045
etc.

Capital V, followed by 14 numerical digits.

I've tried experimenting with adding the following blocks to my InBoundCallRoute (I already have other blocks in place), but obviously I'm doing something wrong, as it's not working:

'[V]xxxxxxxxxxxxxx'
[V]xxxxxxxxxxxxxx
'V13120413700045'
V13120413700045

To avoid confusion, I've got |  (pipe character) on both sides of these rulesets, followed by the other pertinent parts of the ruleset. Like I said, many other blocks in place already, unfortunately. Telemarketers are hell.

What string should I be using? Perhaps I'm mistaken, and I can't do call blocking based on the incoming name from the caller ID.

I've also go my own SIP line and can generate/spoof calls to test rules (which I did to test above), but no luck

ATA is an oldschool OBi110.

Thanks!

[GPz1100]

Are you using google voice or something else?  If google voice, then forward to callcentric's free DID, use a call treatment with an IVR to forward such calls to nowhere.  Is it an actual live person calling or robocall/automated?

[xts431]

Are you using google voice or something else?  If google voice, then forward to callcentric's free DID, use a call treatment with an IVR to forward such calls to nowhere.  Is it an actual live person calling or robocall/automated?

I'm actually using the bridge functionality on the OBi110 to connect a landline from the telco (on LINE port), to the rest of the phones on the "phone" port. These calls are coming in via that line.  The rules affect forwarding between the LINE and the PHONE port. If a matching blocking rule is found, the phones connected to the phone port just don't ring.

Not my line/service, otherwise I'd port-out the number and use a real provider with way better filtering options.

[xts431]

Does anyone else have input on this issue and how it may be resolved from a ruleset within the ATA?

[drgeoff]

Perhaps I'm mistaken, and I can't do call blocking based on the incoming name from the caller ID.

You are not mistaken.

[azrobert]

You need a PBX to block calls by caller name. Yate is a free PBX for Windows that can access caller name. It's very easy to setup. Here is a sample:
http://www.obitalk.com/forum/index.php?topic=13253.msg85279#msg85279

That link is to setup GV. You will only need 1 or 2 config files.

Yate will need to answer a call to block it. You would fork the call in your OBi110 to the Phone Port and to Yate. Yate would answer any call with a caller name equal to "Vxxxxxxxxxxxxxx" and reject all other calls. When Yate rejects a call, the other leg of the fork will continue to ring the OBi110 phone port.  If Yate wasn't active, all calls will ring the OBi110 phone port.

This setup requires a trunk on the OBi110 defined as SIP or a free SP trunk. It's been a long time since I played with Yate, so I will need to test this setup. Let me know if you want to try it.

[azrobert]

Below is the solution, using Yate PBX for blocking calls by caller name. I tested by calling my OBi110's SP2 with a softphone. This is the only way I knew how to set the caller name. I included 2 rules in the regexroute. "^V\([0-9]\{14\}\)$" will check for exactly a V followed by 14 digits. "^V\([0-9]\{14\}\)" will check for a string beginning with a V followed by 14 digits. I did this in case there are blanks or unprintable characters at the end of the name. Pick which one you want or use both. The OBi110 will fork Line calls to the phone port and Yate. If there is a match, Yate will answer the call with a SIT tone ending the ringing on the OBi110. If there isn't a match, Yate will reject the call and the OBi110 phone port will continue to ring. If Yate isn't active, all calls will ring the OBi110 phone port.


Download Yate and run setup:
http://voip.null.ro/tarballs/yate6/yate-6.0.0-1-setup.exe

Take all the default settings.

Don't check any boxes for desktop ICONs.
Those are for the Yate Client, not Yate Server.

Using File Explorer go to:  c:\Program Files (x86)\Yate
Right click on yate-console and create a desktop shortcut.

You only need to create 1 conf file.
Place this file in: c:\Program Files (x86)\Yate\conf.d
Folder conf.d also contains sample configuration files.
Copy the code into Notepad and save as the file name above code.
Do NOT save as txt file
Select file type "All Files".

The Yate folder is restricted.
Assuming you are the administrator on your computer, you will be prompted when saving a file.
You can delete files.
You won't be able to modify an existing file.
You need to change the permissions on the Yate folder to make mods.
You can maintain the files in another location then delete old and copy the modified file.

Here is the config file:

regexroute.conf

Code Select Expand


[default]
${callername}^V\([0-9]\{14\}\)$=tone/info
${callername}^V\([0-9]\{14\}\)=tone/info
^.*$=error=noauth


Start Yate by double clicking on shortcut you created above.
Yate is also installed as a service "Yet Another Telephone Engine". Alternately you can start Yate by starting the service. Change the service to "Auto" to automatically start Yate when computer is powered on. Start testing with the Yate console shortcut. The console will show activity including any errors.

OBi110 Setup:

Service Providers -> ITSP Profile B -> SIP -> ProxyServer: 127.0.0.1
Service Providers -> ITSP Profile B -> SIP -> X_SpoofCallerID: Checked
Voice Services -> SP2 Service -> AuthUserName: anything
Voice Services -> SP2 Service -> X_RegisterEnable: unchecked
Voice Services -> SP2 Service -> X_ServProvProfile: B
Physical Interfaces -> Line Port -> InboundCallRoute: ph,sp2(0@xx.xx.xx.xx)

Replace xx.xx.xx.xx with the IP address of the computer running Yate.

If you already have an SIP trunk defined on SP2, you can use it. Just enable X_SpoofCallerID.