Lately I've been getting numerous telemarketing calls on both of my GV lines. While I can block them after the fact on GV, they keep changing the last 2-digits of their caller ID. GV offers no method to block these calls in advance, so theoretically I'd have to individually block 100 numbers times 2 lines to finally put a stop to this. :(
I have an OBi110 configured with 2 GV accounts setup on SP1 & SP2. SP1 is also forked to my cell phone. The line port is not used/connected. SoftwareVersion is 1.3.0 (Build: 2886).
I'm attempting to use this which I have found here in the forums:
SP1 & SP2 X_InboundCallRoute: {(MTelemarketers)|?:},{ph}
User Defined Digit Map
Label: Telemarketers
DigitMap: (1?41238714xx)
What I'm finding is that while my OBi connected phone does not ring, the caller gets 4 or so rings and then it goes to my GV voicemail. Calls to SP1 still ring my cell due to it being forked.
I though this configuration was supposed to give the caller either a busy signal or dead air. Is it the call-waiting which is causing this behavior?
While I'd really like to send the caller to a SIT disconnected tone, apparently that is not possible with the 110 (or even the newer 200 series). :(
I also tried this alternate call route to send it to my unused line port, but the result was no different than before.
SP1 & SP2 X_InboundCallRoute: {(MTelemarketers)|?:Li},{ph}
Any ideas or suggestions appreciated.
-gary
Using that method to block calls, the call is treated as unanswered and will continue to ring until sent to VM. To prevent this and ringing your cell the call must be answered. If you are not using the Auto Attendant, it can be used to answer the call. You can use a custom greeting with one of these announcements: http://www.beatriceco.com/bti/porticus/bell/recordedannouncements.html
If you don't like these you can search for others.
I'll show you how to modify the AA if you are interested.
There is a BOT called Lenny used to harass telemarketers, but I couldn't find a number that worked. Maybe you can find one. Do a Google search for "lenny bot".
Lenny can be reached as a SIP call to 2233435945@sip2sip.info
I've been using the SIT intercept tones from Wikipedia. It's just two consecutive SIT intercept tones without the additional message. Seems to be doing the job for me.
https://en.wikipedia.org/wiki/Special_information_tones (https://en.wikipedia.org/wiki/Special_information_tones)
It downloads in ogg format which I just converted to a wav file with an online converter.
Quote from: drgeoff on April 18, 2017, 11:03:39 PM
Lenny can be reached as a SIP call to 2233435945@sip2sip.info
The OP has GV defined on both SP1 and SP2, so SIP is not an option for him.
Quote from: azrobert on April 18, 2017, 09:57:25 PM
Using that method to block calls, the call is treated as unanswered and will continue to ring until sent to VM. To prevent this and ringing your cell the call must be answered. If you are not using the Auto Attendant, it can be used to answer the call. You can use a custom greeting with one of these announcements: http://www.beatriceco.com/bti/porticus/bell/recordedannouncements.html
If you don't like these you can search for others.
I'll show you how to modify the AA if you are interested.
There is a BOT called Lenny used to harass telemarketers, but I couldn't find a number that worked. Maybe you can find one. Do a Google search for "lenny bot".
I've listened to some of the Lenny calls on YouTube... very funny. But I've always preferred to not answer telemarketing calls as it just invites more such calls.
The auto attendant is something I rarely use. If it could be setup to play a SIT tone that might work.
Why does my attempt to route the call to the unused line port not work? Is it possible to route to the line port in such a manner as to deliberately cause a call failure and have the OBi110 generate a SIT tone?
-gary
Quote from: Taoman on April 18, 2017, 11:08:29 PM
I've been using the SIT intercept tones from Wikipedia. It's just two consecutive SIT intercept tones without the additional message. Seems to be doing the job for me.
https://en.wikipedia.org/wiki/Special_information_tones (https://en.wikipedia.org/wiki/Special_information_tones)
It downloads in ogg format which I just converted to a wav file with an online converter.
Are you somehow using the SIT tone with the auto attendant? Or...
-gary
Quote from: azrobert on April 18, 2017, 11:24:59 PM
Quote from: drgeoff on April 18, 2017, 11:03:39 PM
Lenny can be reached as a SIP call to 2233435945@sip2sip.info
The OP has GV defined on both SP1 and SP2, so SIP is not an option for him.
Is this because both my SP1 & SP2 slots are occupied, or because GV is not SIP?
-gary
Quote from: gary-gary on April 19, 2017, 06:57:16 AM
Are you somehow using the SIT tone with the auto attendant? Or...
-gary
No. I have a DID with VoIP.ms that I use strictly for SIP URI calls (although the calls originate from my GV number). I shouldn't be receiving any PSTN calls on that line. However, I started receiving a rash of telespammer calls to that number. I originally set it up to use Nomorobo which definitely works but you always get that first unwanted ring before the call is blocked. And the calls persisted. So I switched it to routing to the SIT intercept tone. Now it appears the robodialers will call twice and then give up judging by my call history.
Quote from: gary-gary on April 19, 2017, 06:58:09 AM
Quote from: azrobert on April 18, 2017, 11:24:59 PM
Quote from: drgeoff on April 18, 2017, 11:03:39 PM
Lenny can be reached as a SIP call to 2233435945@sip2sip.info
The OP has GV defined on both SP1 and SP2, so SIP is not an option for him.
Is this because both my SP1 & SP2 slots are occupied, or because GV is not SIP?
-gary
It's because you have no SIP lines. That limits your options.
In the olden days (before voip) I used to transfer calls to the bus station in neighboring DC. They NEVER answered the phone. The local rock radio station had a request line that was always busy.
Now I use recordings and Lenny.
Based upon caller-ID I now can tailor certain calls with Voip.Ms. When Sirius (the dog of a satellite music company) had my number listed for a deadbeat customer, I finally recorded a long message about how I wasn't a customer and was reporting this call each time. The calls stopped.
Although I have GV numbers, my main numbers are with a teal Voip company. Then U can use NOMOROBO, keep my own black and white lists, manage calls based upon partial phone numbers with wildcards, just as you are trying to do.
GoogleVoice is great as a free service, but for a few dollars a month, life becomes easier elsewhere.
QuoteWhy does my attempt to route the call to the unused line port not work? Is it possible to route to the line port in such a manner as to deliberately cause a call failure and have the OBi110 generate a SIT tone?
It's basically the same scenario as routing the call to nowhere. GV is forwarding the call to your OBi110. If the forward fails or not answered, it will be sent to VM. A forked call will continue to ring your cell if the other leg fails.
Auto Attendant Setup:
Voice Services -> Auto Attendant
Under Auto Attendant Prompts set the following.
Menu: %User1%
%User1% will point to custom prompt#1.
All Other Prompts set to: &pause()
&pause() is the equivalent of deleting the prompt.
OutboundCallRoute: {}
AnswerDelay: 0
Voice Services -> SP1 Service -> X_InboundCallRoute:
{(MTelemarketers)|?:aa},{ph}
Record a custom prompt:
Dial the following:
***0
1001# This will save the recording to %User1%.
1002# will save to %User2%
1 This is for a new value for the recording
1 Any digit will start the recording
Say your custom message
# Will end the recording. Leave a couple of seconds
of silence at beginning and end.
1 Confirm the recording and save
Instead of saying the custom message, hold the phone next to the computer speaker and play Toaman's suggestion from Wikipedia.
Quote from: Lavarock7 on April 19, 2017, 08:31:55 AM
Although I have GV numbers, my main numbers are with a teal Voip company. Then U can use NOMOROBO, keep my own black and white lists, manage calls based upon partial phone numbers with wildcards, just as you are trying to do.
GoogleVoice is great as a free service, but for a few dollars a month, life becomes easier elsewhere.
While I would like more configurability at times, the ability of GV to weed out the mass of telemarketing calls is something I really really like. I only need to deal with those few who get through. I'm not aware of other VOIP services which do as good a job of blocking these calls.
Nomorobo looks to be available for GV, as I see it in their list of carriers... but even the single ring of a telemarketer is a disruption which I don't appreciate.
-gary
Quote from: azrobert on April 19, 2017, 08:34:41 AM
QuoteWhy does my attempt to route the call to the unused line port not work? Is it possible to route to the line port in such a manner as to deliberately cause a call failure and have the OBi110 generate a SIT tone?
It's basically the same scenario as routing the call to nowhere. GV is forwarding the call to your OBi110. If the forward fails or not answered, it will be sent to VM. A forked call will continue to ring your cell if the other leg fails.
Auto Attendant Setup:
...
This is a rather neat trick! Thanks! I've implemented it with a few additions...
I found that the OBi announcements "Welcome to OBi attendant... Main menu" were still being played. I managed to disable them with these additional edits:
Auto Attendant 1 Prompts > Welcome: &pause()
Auto Attendant 1 Prompts > MenuTitle: &pause()
In testing, I found that it failed for some reason 4 out of 10 times, and the call went to GV voicemail. I'm puzzled by this, as a 40% failure rate is not good. Possibly I did not give the unit enough time to reestablish its connection to the GV mothership after rebooting.
Now is there some way terminate the call quicker? I find that it will play my recorded SIT intercept message 4-times, followed by "there is no service to complete your call" 3-times, before finally hanging up. Perhaps some edits to the current AA DigitMap?
([1-9]x?*(Mpli)|[1-9]|[1-9][0-9]|<00:$1>|0|**1(Msp1)|**2(Msp2)|**8(Mli)|**9(Mpp)|(Mpli))
-gary
Well, I just made an interesting observation...
My recorded SIT message consists of the intercept tones repeated twice. This message was initially played 4-times.
Then the OBi message "there is no service to complete your call" is played 3-times, each time followed by a single SIT intercept tone. This must be the internally generated SIT, as my recorded one is a double sequence!
-gary
Quote from: azrobert on April 19, 2017, 08:34:41 AM
Auto Attendant Setup:
Voice Services -> Auto Attendant
Under Auto Attendant Prompts set the following.
Menu: %User1%
%User1% will point to custom prompt#1.
All Other Prompts set to: &pause()
&pause() is the equivalent of deleting the prompt.
Oops!
I just realized I made a mistake... I misinterpreted this as "All Other
USER Prompts set to: &pause()"
-gary
Currently I have a "friends and family" number which only allows the call to proceed if the caller-id is in my AsteriDex database. Otherwise, the caller gets to enjoy a set of fax tones. (This requires more than an OBi to accomplish. In my case, I have an Asterisk server in the mix.) I don't answer my "public" number -- the one I give out -- unless I happen to be by the phone and recognize the caller. (I figure someone legitimate will leave a message, although it occasionally gets interesting when someone who similarly doesn't want to leave their direct number trys to reach me.)
To be honest, I've toyed with the idea of routing the friends and family spam callers to a recording which starts off "911 - What's your emergency?" and then pauses a few seconds and hangs up, but I really don't get enough spammers on that number yet to make it worthwhile.
Quote from: gary-gary on April 19, 2017, 11:34:02 AM
While I would like more configurability at times, the ability of GV to weed out the mass of telemarketing calls is something I really really like.
Speaking of which, I just saw this post in Google Voice support forum referenced by Android Police:
QuoteSpam callers beware!
To better protect you from unwanted calls, we've recently improved our spam filter for Google Voice. Using the same technology that powers spam protection in Pixel, Nexus, and Android One devices, we now catch 2x more spammers, receive 20% fewer spam reports from users, and identify 40% more calls correctly as spam than ever before.
https://productforums.google.com/forum/#!topic/voice/v5AzF8VpMes (https://productforums.google.com/forum/#!topic/voice/v5AzF8VpMes)
The AA wasn't designed to do this and is limited on the modifications you can make. Here is the normal function:
The Welcome, Menu Title and Menu prompts are played. If there is no response the Menu prompt is played twice more. If there is still no response a zero is sent to the AA DigitMap where it is validated and then the zero is passed to the outbound route. If the outbound route detects a zero, it will route the call to the phone port.
This basic function can't be changed. The example I gave you, all the routing was removed from the outbound route which produced the message "No routes available" plus the SIT tone. I just tried changing the outbound route to "pp($$$$$$$$$)" without the quotes. This will send the call to the OBiTalk network to an invalid destination. I got a similar result with a different message. Same when I changed the DigitMap to "(A)", so the zero will fail to validate.
You can try moving the Custom prompt to the Welcome prompt and delete the Menu prompt. The Welcome prompt is played only once compared to 3 for the Menu prompt and might reduce the time by a few seconds.
Restamp gave me an idea. You can route a robo calls to a fax machine. If you do this and the fax is busy, the call will go to VM. 1-408-487-4700 is Frys' fax number. You can pick another fax number.
Voice Services -> SP1 Service -> X_InboundCallRoute:
{(MTelemarketers)|?:sp2(14084874700)},{ph}
Edit:
You can fork the call to 2 fax machine to reduce the chance of a busy. The 1st fax to answer will get the call.
{(MTelemarketers)|?:sp2(14084874700),sp1(...........)},{ph}
Things seem to be working hit & miss...
I have configured both SP1 & SP2 per your example. My User Defined Digit Map contains the number of a telemarketer, along with my cell and several others temporarily to allow me to test things.
Using my cell, I call to SP2 and everything goes as expected... the OBi phone does not ring, I hear the SIT tone and then the "no service" message with its SIT tone. I can't test SP1 with my cell as it is forked with SP1 and would go immediately to the GV attendant.
So next I asked a friend to call me to test things. His calling number is placed in my DigitMap. From his perspective, both of the calls ring several times then go to GV voicemail. From my perspective, the OBi phone does not ring either time, but my cell rings with SP1.
I know both calls made it to my OBi110 as they show up in the call log, but neither one has the "Call Transferred" annotation in the log.
Call 6 04/19/2017 18:08:40
Terminal ID GoogleVoice2 AA1
Peer Name
Peer Number 1407.......
Direction Inbound Inbound
18:08:40 Ringing
18:08:40 Call Connected
18:08:55 End Call
Call 7 04/19/2017 18:07:06
Terminal ID GoogleVoice1 AA1
Peer Name
Peer Number 1407.......
Direction Inbound Inbound
18:07:06 Ringing
18:07:06 Call Connected
18:07:21 End Call
His calls were placed from a GV number. Suspecting something may be occurring as a result of it being a GV number, he tries again by calling from a magicjack number. This time SP1 operates as expected, but SP2 rings through to GV voicemail. I hear nothing from either my OBi phone or cell phone. Both calls show up in my log, and I can see where the SP1 call transferred.
Call 3 04/19/2017 18:24:53
Terminal ID GoogleVoice1 AA1
Peer Name
Peer Number 1407.......
Direction Inbound Inbound
18:24:53 Ringing
18:24:53 Call Connected
18:25:20 Call Transferred Transfer to Announcement1(1407.......), State=connected, Ref=60205
18:25:42 End Call
Call 4 04/19/2017 18:22:53
Terminal ID GoogleVoice2 AA1
Peer Name
Peer Number 1407.......
Direction Inbound Inbound
18:22:53 Ringing
18:22:53 Call Connected
18:23:08 End Call
This is getting curiouser and curiouser!
-gary
For testing you should be able to call GV#1 on SP1 using GV#2. The call won't be routed to the phone port, so there shouldn't be a conflict.
When I was testing the AA I had an audio problem and when I checked the Call History there wasn't a transfer. I made the call using an SIP softphone and called directly to SP2 on my OBi110, so this is totally different than your scenario. I fixed my problem by port forwarding the RTP ports to my OBi110's IP address. If this is an RTP problem, I don't know why it would work on SP2 and not on SP1, but I have nothing else to try.
Are you familiar with port forwarding in your Router? The OBi110 RTP ports are found here:
Service Providers -> ITSP Profile A/B RTP -> LocalPortMin thru LocalPortMax
SP1 RTP Ports are 16600 thru 16798
SP2 RTP Ports are 16800 thru 16998
You can use 1 entry in your router forwarding ports 16600 thru 16998 to the IP address of your OBi110.
Quote from: azrobert on April 19, 2017, 07:29:58 PM
Are you familiar with port forwarding in your Router? The OBi110 RTP ports are found here:
Service Providers -> ITSP Profile A/B RTP -> LocalPortMin thru LocalPortMax
SP1 RTP Ports are 16600 thru 16798
SP2 RTP Ports are 16800 thru 16998
You can use 1 entry in your router forwarding ports 16600 thru 16998 to the IP address of your OBi110.
I forwarded these ports in my router, and confirmed that they were indeed the correct ones as listed in the OBi's settings... but the issue continues.
It seems that a good percentage of the time (40%?) the call goes to voicemail rather than properly handled by the OBi110. It does not appear to be limited to either SP1 or SP2, or even the calling source... it occurs with any of the above.
My call history shows the call, shows the intended end point as AA1, but on the failures it does not complete the "Transfer to Announcement" for some reason. :(
-gary
I created a custom greeting containing 2 SIT tones. I tested a dozen or more times and heard the tones every time. You won't see the "Transferred" message in the call history unless the OBi110 tries to transfer the call. I had to enter "1" or wait long enough for my OBi110 to automatically try to transfer the call. I don't know why you don't hear the tones.
I created a Tropo app that plays the SIT tone 3 times. I thought you can test with my app and if works I would help you create your own app, but I forgot your OBi110 defined with 2 GV accounts can't make SIP calls. You could get a free phone number from IPComms or Callcentric and point the number at the Tropo app. Callcentric requires you to purchase e911 for $1.50/mo unless you lie and say you will only use the number for international calls. You also need the free Callcentric IP Freedom account.
Another method is to convert 1 of your GV numbers to SIP by purchasing Simon Telephonics GV Gateway for a one-time fee of $5.99. Now your OBi110 will be able to make SIP calls. The GV Gateway has an added feature of CallerID Name.
http://www.ipcomms.net/freedid
https://www.callcentric.com/products/
https://simonics.com/services/
OK, I think I may have figured out the issue...
I think the problem was that I did not allow for enough delay time in my "message" before recording the SIT tones. I suspect that GV was somehow still on the line during the hand-off to my OBi box... thought it was an out-of-service line and therefore did not complete the call.
I temporarily reverted to the original "Welcome to OBi..." message and all was well. I have since rerecorded my SIT tones with a bit longer pause at the beginning.
On a side note, I had configured the OBi110 to generate syslogs and was attempting to see what sort of clues I could find in there. I had placed several calls to the OBi to produce some data and while examining the logs, suddenly some additional call data appeared! Looking at the OBi's call history I saw it was the notorious telemarketer which had sent me down this path. Their call was successfully intercepted without my phones ringing and ended with their hangup 9-seconds later! ;D
-gary
I just wanted to follow up on how this has been working so far...
Perfectly!!!
My phones connected to the OBi110 no longer ring when the pesky telemarketer calls, plus the OBi unit grabs the call before it can be forked to my cell phone, so it too stays quiet.
My call log shows the telemarketer is still calling however, so the SIT tones have not stopped their attempts. They do seem to disconnect quickly as the call lasts only about 8-10 seconds.
I have also increased the range of my filter to cover the entire exchange, as examination of my call logs showed several older calls were outside of the original filter which blocked only part of the exchange.
I'm a happy camper! ;D
-gary
Quote from: drgeoff on April 18, 2017, 11:03:39 PM
Lenny can be reached as a SIP call to 2233435945@sip2sip.info
Lenny may cause a risk with the "Can you here me now" scam, as Lenny's dialog contains about four "Yes" words in a row after about 2 minutes in to the recording.
The truth about the 'Can you hear me?' scam - CNET
https://www.cnet.com/how-to/can-you-hear-me-scam/
Feb 1, 2017 - The truth about the 'Can you hear me?' scam ... you. This scam has previously targeted businesses but is now allegedly going after individuals.
I wouldn't worry too much about the "Can you hear me" scam and Lenny. Let them produce part of a recording to prove you agreed to anything. Now let them prove you are a senile British/Australian gentleman. In seconds you can produce the Lennt recording showing how they cut and pasted an answer supposedly from you.
let's start an army of "Silent Hello Robo-Answerers".
When you get a robo/telemarketer, punch through and set the phone down. Don't say or do anything until they hang up (or ma bell or ma Google hangs up for you).
Now imagine if hundreds of thousands of robocalls were answered that way. The people doing the robocalling would have to hire a huge staff to handle the "silent hellos". Their queues would be full and without handling each silent hello, they may lose the chance at the few people who fall for the call. It would kill the robo-call business.
And it is kinda fun to listen to them "hello, hello, are you there, can you hear me, hello..."
Spread the word, Help build an army of us who answer the calls, punch through and then be silent (or put by the radio). An army of "Silent Hello Robo-Answerers"
Update: It turns out that this technique is affecting robocallers and is why some systems now require a 2nd button press. It is an attempt to overcome those who start their answering machine message with the 1 DTMF tone. So maybe there is something we could do with AA where the message starts with a 1 DTMF tone.