Robocallblocker
Configurations for OBI110 and FreePBX/Asterisk to block landline robocallers.
We have an old-school landline phone for home use and have been getting many SPAM calls. This is an approach to blocking those calls using an OBI110 ATA and FreePBX.
It simply asks callers to press a key, and if they do, it then transfers the call to our regular landline phone. If no one answers, it sends the call to a good voice mail extension. If they don't press a key, it sends the call to a SPAM voice mail. A whitelist allows known callers bypasses the IVR Spam filter altogether and rings your phone directly.
The ATA used in this document is an old OBI110 I had purchased a few years ago and finally got around to setting up to communicate with FreePBX on an old HP mini netbook.
(Sources: ObiTalk Forum: Using the Obi 110 as an FXO and FXS Port for FreePBX, FreePBX Documentation wiki)
OBIHAI OBI110 - setting up basic communications between OBI110 and FreePBX
This document assumes that you have your OBI110 connected to your network and telephone jack, and have installed FreePBX 14 (with version 13 of Asterisk) on a dedicated computer, which is connected to the same network as the OBI110.
The OBI110 ip address is assumed to be 192.168.1.13, FreePBX is at 192.168.1.11, and your router is at 192.168.1.1. Change this to reflect your network settings. This document also assumes that your home phone number is 555-555-1234 - change this to your phone number.
1. OBI110 Network Settings
Internet Settings
IPAddress 192.168.1.13 (requires static ip address)
SubnetMask 255.255.255.0
DefaultGateway 192.168.1.1 (ip address of your router)
DNSServer1 192.168.1.1
2. Configure Line Port
Connection from wall telephone jack (rj11) to OBI110 to FreePBX trunk.
The RJ11 jack is connected via a standard telephone cable to OBI110. The OBI110 is then connected to your home LAN using Ethernet cable so that it can connect to your FreePBX server.
2.a. OBI110 Line Port
Physical Interfaces > LINE Port
Enable selected
DigitMap default
InboundCallRoute SP2(5555551234)
(SP2 = SP2 Service (under Voice Services). It puts your telephone number in the SIP2 Inbound route on the OBITrunk1. SP2 is configured on the OBI110. Trunk and associated Inbound route is configured on FreePBX server).
I had problems with some calls not where DTMF tones (key presses) were not being loud enough, to increase sensitivity, reduce from default of 5 to 2:
ChannelRxGain: 2
2.b. ITSP Profile B (ITSP = Internet Service Provider - i.e. FreePBX)
General
Name Line to SP2 to FreePBX Trunk
DigitMap default
Service Provider Info - defaults
SIP
ProxyServer 192.168.1.13 (ip address of FreePBX server)
ProxyServerPort 5060 (default)
RegistrarServerPort 5060 (default)
OutboundProxyPort 5060 (default)
X_SpoofCallerID - selected
X_AccessList 192.168.1.13 (only allow access to OBI110 from FreePBX server ip address)
FreePBX server port 5060 points to chan_sip on FreePBX in this configuration; (this is not a default FreePBX setting)
2.c. Voice Services
SP2 Service
Enable selected
X_ServProvProfile B (points to ITSP Profile B)
X_InboundCallRoute LI (LI = line = cable connected to telephone wall jack)
X_KeepAliveServerPort 5060 (default)
X_UserAgentPort 5061 (default)
SIP Credentials
AuthUserName OBITRUNK1 (used in FreePBX)
AuthPassword password (used in FreePBX)
Calling Features
defaults
2.d. FREEPBX Trunk
Connectivity > Trunks
Trunk Name OBITRUNK1
Outbound CallerID 5555551234 (phone number of line attache to OBI110)
Dial Number Manipulation Rules - none
sip Settings
Outgoing
Trunk Name OBITRUNK1 (must match OBI110 SP2 configuration)
PEER Details
username=OBITRUNK1 (must match OBI110 SP2 configuration)
secret=password (must match OBI110 SP2configuration)
host=dynamic
type=friend
context=from-trunk
qualify=yes
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
Incoming (leave blank)
2.e. FreePBX Inbound Route
Connectivity > Inbound Routes
Route: Obi110
General
Description Obi110
DID Number 5555551234 (phone number of line attache to OBI110)
Set Destination
Extensions > Add new Extension
Advanced - defaults
Fax - defaults
Other - defaults
Applications > Extensions
Extensions: 300
General
Display Name - myextension
Secret (long password generated by FreePBX that will be used to configure the OBI phone, sip1 configurations)
User Manager Settings - defaults
(an extension, which can be accessed using numeric keypad of telephone set is also associated with a FreePBX username, see Admin>User Management)
Voicemail
Enabled - selected
Voicemail Password 1234 (numeric password used to access voice mail from phone handset)
Disable (*) in Voicemail Menu - No
Find Me/Follow Me - defaults
Advanced - defaults
Pin Sets - defaults
Other - defaults
2.f OBIHAI OBI110 - test connection to FreePBX
Status > System Status
scroll down to SP2 Service Status. Should see something like this:
Status Registered (server=192.168.1.13:6060; expire in 46s) help
3. PHONE
This sets up a communication link between your phone and FreePBX (i.e. a line on the OBITRUNK1)
3.a. OBI110 PHONE Port
Physical Interfaces > PHONE Port
Enable selected
DigitMap default
OutboundCallRoute default
CallReturnDigitMaps default
PrimaryLine SP1 Service
Ringer - defaults
Port Settings - defaults
Calling Features
StarCodeProfile None
Timers - defaults
Tip-Ring Voltage Polarity - defaults
3.b. ITSP Profile A (ITSP = Internet Service Provider - i.e. FreePBX)
General
Name Phone to OBI110 to FreePBX Trunk
DigitMap default
SIP
ProxyServer 192.168.1.13
ProxyServerPort 6060 (nonstandard port)
RegistrarServerPort 6060 (nonstandard port)
OutboundProxyPort 6060 (nonstandard port)
X_UseRefer - selected
X_AccessList - selected
X_MWISubscribe - selected (might not be required since not using sip phone)
(Port 6060 corresponds to non-standard port for Chan PJSIP Settings)
3.c. Voice Services
SP1 Service
Enable selected
X_ServProvProfile A (default - points to ITSP Profile A)
X_InboundCallRoute ph (default)
X_KeepAliveServerPort 6060
X_UserAgentPort 6060
(ph = phone)
SIP Credentials
AuthUserName 300 (extension created earlier)
AuthPassword FREEPBXPASSWORD (long password generated by FreePBX when extension was created)
Calling Features
MaxSessions 10
MWIEnable selected
X_VMWIEnable selected (should not be required since not using SIP phone)
MessageWaiting unselected, not default
3.d. OBIHAI OBI110 - test connection to FreePBX
Status > System Status
scroll down to SP1 Service Status. Should see something like this:
Status Registered (server=192.168.1.13:5060; expire in 34s)
3.e. FREEPBX Outbound Routes
Your landline phone now has a route to the FreePBX server.
To actually connect to FreePBX from your phone (via OBI110) you need to dial '**1'. You can do this now.
From here you can dial 300 to get the extension we create and leave a voice mail.
Or, if you want to call someone as you normally would on a telephone, we need to tell FreePBX how to do this. One way is to create a dial code that tells FreePBX to send your keypresses to the PSTN (Public Service Telephone Network). This is how we create that code.
Connectivity > Outbound Routes
Route Settings
Route: 81DialsPOTS
Trunk Sequence for Matched Routes: OBITRUNK1
Dial Patterns
81|
81|1NXXNXXXXXX
81|NXXNXXXXXX
81|NXXXXXX
81|X11
Import/Export Patterns - not applicable
Additional Settings - defaults
(POTS = Plain Old Telephone Service = PSNT)
4. Whitelisting
The Inbound Route on FreePBX decides what happens when someone calls your phone. As part of the initial set up, we told FreePBX to send the call to extension 300, and since your phone is linked to this extension through OBI110, any call coming in will ring on your landline phone.
What can we do to prevent robo calls from making your phone ring?
Spam callers usually use a predictive dialer to connect you to their spam agent. What happens is that the Robocaller calls many nuumbers, and if someone answers, proceeds to try to match them with an available attendant. But the robocallers are dumb, and if you ask them to press a digit, they cannot (not yet at least - voice spammers are beginning to use speech recogniion, more on that later...).
Therefore, to prevent roboCallers, you need require anyone calling you to dial 8 before allowing the call to go through. We can use the FreePBX IVR app to do this.
4.a Record caller prompts
First, we need to record our the messages we want the caller to hear.
Using an Audio recording app (such as Audacity), record the following phrase:
Hello, you've reached 555 555 1234, please press 8 to continue
and save it as a wav file named:
greetingAndPressToContinue.wav
(note: if your browser supports it, you can record your prompt in your browser when adding a new recording - see: Admin > System Recordings, then 'Record in Browser')
4.b Upload recording to FreePBX
Next upload your wavefile to FreePBX
Admin > System Recordings
Add New System Recording
Name greetingAndPressToContinue.wav
Description greeting and press key to Continue
Upload Recording
4.c Setup IVR
on FreePBX: Application > IVR
IVR General Options
IVR Name PressToContinue
IVR Description Press key in order to continue
IVR DTMF Options
Announcement greetingAndPressToContinue.wav
Enable Direct Dial Disabled
Timeout 5 (default of 10 seconds is too long)
Invalid Retries 1 (default is 3; one is plenty to then go to Voice Mail)
Invalid Recording Default
Invalid Destination Extensions: 400 spamExtension
Timeout Retries 1 (default of 3 is frustrating for users who just want to leave a VM but their keypresses are not being recognized - for a variety of reasons)
Timeout Retry Recording Default
Timeout Recording Default
Invalid Recording Extensions: 400 spamExtension
Return to IVR after VM No
IVR Entries
Digits 8
Destination Extensions: 300 myextension (already created)
return No
Applications > Extensions
Extensions: 400
General
Display Name - spamExtension
Secret - default (long password generated by FreePBX)
User Manager Settings - defaults
Voicemail
Enabled - selected
Voicemail Password 1234 (numeric password used to access voice mail from phone handset)
Disable (*) in Voicemail Menu - Yes
Find Me/Follow Me - defaults
Advanced - defaults
Pin Sets - defaults
Other - defaults
4.d Voicemail volume
Settings > Voice Mail
Voicemail Recording Gain 3 db
5. Troubleshooting
5.a fail2ban
FreePBX uses fail2ban. If as you are setting up FreePBX and logging on and logging off, and you get locked out for no apparent reason, it is likely you have been caught by fail2ban.
Go to Admin > System Admin
select: Intrusion Detection
Add the ip address of the computer you are using to remotely access the FreePBX server to the Whitelist section (you may have to do this from a different computer, or restart FreePBX - note you only have a minute or two before fail2ban blocks your ip address again).
Submit your change.
5.b Debugging calls
To debug inbound calls by listening to the full conversation, use the following settings:
First set FreePBX to record all incoming calls
Connectivity > Inbound Routes
edit Obi110 Inbound Route
Other Tab
Call Recording: Force
Next, set FreePBX record entire conversation (including IVR prompts and replies)
Settings > Advanced Settings
Call Recording
Call Record Options: No
(KEYWORD:CALL_REC_OPTION - if set to Yes, it will not begin recording unless a call is bridged to another channel - i.e. connected to an extension)
5.c Time issues
Problem: UC logs have wrong time
Solution: enable NTP (Network Time Protocol)
(see: https://www.freepbxhosting.com/blog/fix-incorrect-time-freepbx/)
SSH into your PBX, confirm that NTP is running:
service ntpd status
To restart NTP:
service ntpd restart
To configure NTP to run at system start:
chkconfig ntpd on
To see which servers your PBX is syncing time with:
ntpq -p
6. Direct Outbound Calling
Default OBI110 outbound call routes send you through FreePBX and then you must use the previously configured FreePBX outbound route to call as you normally would. So to make a call to the PSTN, you need to:
a. press **1 (to access FreePBX )
b. press 81# (to access pstn)
c. dial your number
Another option in the default OBI110 outbound call routes is to:
a. press **8 (to connect to line port on OBI110, bypassing FreePBX altogether)
b. dial your number
A final option in the default OBI110 outbound call routes is to:
a. press # (to connect to line port on OBI110, bypassing FreePBX altogether)
b. dial your number
Since we are not using SIP calling, we don't really need FreePBX for outbound calls, so lets find a more direct way to bypass FreePBX and make calls without having to press **8 or having to press the pound key (#).
DigitMap and OutboundCallRoute - Defaults
DigitMap: ([1-9]x?*(Mpli)|[1-9]|[1-9][0-9]|911|**0|***|#|**1(Msp1)|**2(Msp2)|**8(Mli)|**9(Mpp)|(Mpli))
OutboundCallRoute: {([1-9]x?*(Mpli)):pp},{(<#:>|911):li},{**0:aa},{***:aa2},{(<**1:>(Msp1)):sp1},{(<**2:>(Msp2)):sp2},{(<**8:>(Mli)):li},{(<**9:>(Mpp)):pp},{(Mpli):pli}
Understanding the Defaults
DigitMap
A digit map can be used to perform the following tasks:
• Ensure a complete number is dialed
• Transform dialed digits
• Block numbers from being dialed
Outbound Call Routes
We use a digit map to match or transform a number that’s been entered. Once the digit map has matched the entered sequence, we can then use the outbound call route to direct the call to the correct terminal on the OBi. The outbound call route directs the call from the perspective of the user placing the call from the OBi device.
Terminals on the OBi
Each physical connection on the OBi device is addressed as a logical terminal by the software. OBi devices support the following terminals:
Phone (FXS) Ports - ph1
Line (FXO) Ports - ln1
IP Voice Services - sp1, sp2
...
source: Obi DigitMap CallRoute Tutorial
OutboundCallRoute - Review
Each route in a terminal's OutboundCallRoute is separated by a comma. Lets break them out and analyze each in turn:
{([1-9]x?*(Mpli)):pp}
This route is not required because it routes to pp, which is used for Obi to Obi calls, and therefore can be deleted.
{(<#:>|911):li}
Two rules:
a. If the caller dials # followed by any other digits, then route call directly to Line Port
b. If caller dials 911, then route call directly to Line Port
{**0:aa}
If the caller dials **0, then route call directly Obi110 Auto Attendant
{***:aa2}
If the caller dials ***, then route call directly Obi110 Device Configuration Menu
{(<**1:>(Msp1)):sp1}
If the caller dials **1, then connect call to FreePBX (to listen to voice mail, etc)
{(<**2:>(Msp2)):sp2}
sp2 is used for outbound calling from FreePBX, no reason to access by phone, delete
{(<**8:>(Mli)):li}
If the caller dials **8, then bypass FreePBX, and make a call as you normally would on PSTN.
{(<**9:>(Mpp)):pp}
routes to pp, which is used for Obi to Obi calls, and therefore can be deleted.
{(Mpli):pli}
This route is not required because it routes to pli, which is the primary line which corresponds to SP1 Service, and since we are not using VoIP calling, we can be delete it.
User Defined Digit Map
Let's create a user define digit map to be able to dial a number as one would on a regular phone, bypasing FreePBX.
First we need add a DigitMap entry that filters the types of call we want to go to the OBI110 Line Port:
User Settings > User Defined Digit Maps
label: dm2
DigitMap: (1xxxxxxxxxx|[2-9]xxxxxxxxx|*98)
So now for outgoing calls using 10 digit local number, or long distance number prefixed by 1, you can access the PSTN directly rather than pressing pound key ('#') or using **8 to access the PSTN network.
In addition, my phone providers voice mail uses *98, now I can directly access it even though my phones are connected to the OBI110/FreePBX. It is possible that FreePBX is busy with a call and second call goes to service provider voice mail system, so need a simple way to check it.
(Note: these may need to be changed if you are in a different country or want direct access to other numbers, or just prefix all your calls with # to access the PSTN)
Then add a reference to the User Defined DigitMap you just created (dm2) in the final DigitMap/OutboundCallRoute for the Phone terminal. It should look like this:
DigitMap: (911|**0|***|#|*97|(Mdm2)|**1(Msp1))
OutboundCallRoute: {(<#:>|911):li},{**0:aa},{***:aa2},{(*97):sp1},{(<**1:>(Msp1)):sp1},{(Mdm2):li}
CallReturnDigitMaps: Default
Also added ability to check voice mail on FreePBX by pressing *97, and removed default **8 to access the PSTN network which is not required since can get to PSTN with #.
TODO:
Sending email notifications of Voice Mails
fix extensions to both use pjsip