VOIP Call Screening devices ?

I have recently ported my former PTSN landline as a VOIP line to AAISP

As I'm using a Gigaset N300A IP, my previous call screening device (Truecall) which is PTSN based no longer works and consequently I am being plagued by unwanted calls!

One solution I have considered to reinstate the Truecall device is to purchase a ATA Telephone Adapter - eg say a Granstream HT802 and use this (rather than the N300A) as the VOIP connection endpoint. Essentially I would be reverting to using the PTSN functionality of the N300A. However I note that the Truecall website does not recommend using ATA adapters.

Does anyone know of a comparable device that works with VOIP or any software (eg running on say a Raspberry Pi) that could provide such functionality ?

Looking around there appears to be a dearth of such devices, but I cannot but help think as telephony migrates to VOIP this will become a wider issue.

Cheers

freepbx would do the job, but it would be a learning curve http://www.raspberry-asterisk.org/

I've used an Asterisk-based VOIP system (like FreePBX) for some years and I've found that a very effective anti-spam technique is to program the system to answer a call and play a short announcement ("Please hold the line, we're connecting your call") before ringing any of the house phones.

Almost all nuisance callers (or the automatic dialling systems they use) interpret this as an answering machine and hang up, whereas genuine callers wait. I don't even know the nuisance callers have called until I check the logs.

Thanks for the suggestions - Having has a quick look I suspect that Asterisks is rather overkill for what I need (as well as having somewhat of a steep learning curve).

That said if the call spammers continue at the current rate I may have to go down this route.
Cheers

There is an option in the control panel to block withheld/ unknown callers which may help.

I wonder if turning on the announcement that the call is being recorded (even if you don't turn on the actual call recording) would make them think twice.

Many years ago I did this with FreePBX/Asterisk, but I don't use that anymore as I don't host it locally. I currently use Flextel and have two phone numbers. One which is an 03 number which is unlikely to get spam calls, and one which is a normal local area code which can and does occasionally receive spam calls.

The 03 number is a direct connection to my phone, while the local area code number points to an IVR system where it has a pre-recorded message telling them to press 1 to be connected pretty much. If they don't press 1 then it won't silently redirect them to my 03 number. Not the best setup I imagine but it works.

Thanks to all for yet further suggestions and observations.

As I continued to be plagued by unwanted SPAM callers (some international, some ostensibly local, but patently also international) somewhat in desperation I decided to go down the Grandstream HT802 route and purchased the same.

I utilised the Grandstream as my VOIP connection point to AAISP and then connected one of its 2 FXS (RJ11) ports to the input side of my Truecall device (required a RJ11 to BT Socket adapter). I then connected the RJ11 output on the Truecall device to the RJ11 PTSN input (RJ11) port on my Gigaset N300A IP.

The Gigaset Control Panel successfully showed a PTSN connection and test calls show that the Truecall device is operating totally normally and I am back in business!!

The average cost of an overnight call to AAISP to synchronise the Truecall unit is currently 2p (takes under a minute).

As an added bonus I can still use the remaining 5 VOIP accounts on the Gigaset, plus there is a spare FXS port on the Grandstream (obviously these will not NOT protected by Truecall device),

For the record the Truecall unit did require a firmware update (I rang their help desk and they imitated the same via the usual modem dial up call route). Mine is quite an old device >5 years old.

Current Truecall firmware I am using is

Runtime Version 34.11
Macro Version 6.28

Gigaset firmware is

42.250 (422500000000 / V42.00)

Delighted to be back to peace and quiet.

Hi farrinaf,

I'm in a similar position to yours, but my Truecall unit is failing its weblinks after switching to VoIP using a Grandstream HT01 (same as the 802, but only one port).

The Truecall unit has been updated (mine was old too) and works OK for incoming and outgoing calls, except for taking voicemail and making weblinks. I've tried it on another landline and it weblinks OK on that.

I was wondering whether there is some relevant Grandstream setting I've missed. My VoIP service is Sipgate and I used their settings. I've tried some UK dial tone and ring settings, etc, but still no joy.

Truecall support are on the case, but it's clearly a problem with my VoIP.

Can you say what changes you made to the Grandstream config settings?

Hi
I'm in a similar situation haing been moved to SOGEA/VOIP. I have a Fritzbox router with integrated VOIP/Anserphone.

I have been in touch with the makers and they have shown how to configure it to configure a 'whitelist'. IE For callers in your phone book it behaves as normal (inc answerphone). If the caller is not in my phone book they get sent to answerphone 2 where they get a message 'Your number is not recognised. Please leave a message and I will call back.' Its a bit messy to set up but they have put it forward to developers as a feature request to mke it easier to configure. I have not yet tested it but PM me if you want more details.

In reply to a post by DrBob:

Hi
I'm in a similar situation haing been moved to SOGEA/VOIP. I have a Fritzbox router with integrated VOIP/Anserphone.

I have been in touch with the makers and they have shown how to configure it to configure a 'whitelist'. IE For callers in your phone book it behaves as normal (inc answerphone). If the caller is not in my phone book they get sent to answerphone 2 where they get a message 'Your number is not recognised. Please leave a message and I will call back.' Its a bit messy to set up but they have put it forward to developers as a feature request to mke it easier to configure. I have not yet tested it but PM me if you want more details.

Dr Bob, when you refer to "the makers" above do you mean AVM the fritzbox makers or Truecall?
I am about to join Zen ISP from virgin so no DSL line installed yet. ZEN now only offer VOIP using what I assume is an ATA built into the Fritz box into which I will connect my existing phone pair after disconnecting it from virgin. I already have an analogue answering machine and not planning to use the fritzbox one.
I would like pick up a used truecall off ebay etc to cut down the junk calls. I'm not at all clued up on VOIP or much of the terminology used further up this thread. Any guidance you can offer will be most welcome. TIA