STUNT BANANA

Minimalist Asterisk Caller ID Spoofer and Secondary VOIP Line Configuration Built for AWS

Contact: DilDog (twitter.com/dildog) ([email protected])

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Introduction

Things to know:

1. STUNT BANANA provides a Caller ID spoofing mechanism much like SpoofCard and other available services, but at a much reduced cost, if you don't mind doing the setup yourself and having a much more minimal UI.
2. STUNT BANANA also allows you to host new phone numbers (DIDs) for your devices and use a SIP Phone app, such as Zoiper to place and receive calls, as well as get voicemail for those lines sent your email as MP3 files.
3. Spoofing Caller ID is not illegal. Impersonating other people and committing fraud is. If you bulk call people with spoofed caller IDs, your SIP trunk provider will notice and you will get taken down and possibly receive criminal charges. Don't be dumb.

Acknowledgements

Props to Jonathan Stines for his blog entries about this subject:

• https://blog.rapid7.com/2018/05/24/how-to-build-your-own-caller-id-spoofer-part-1/
• https://blog.rapid7.com/2018/07/12/how-to-build-your-own-caller-id-spoofer-part-2/

I started with this and decided to strip out all the stuff that I thought was unnecessary to make it more lightweight and secure, and to document the configuration as much as possible.

Usage

Once set up, here's the basics of using STUNT BANANA:

To use the spoofer application:

• From any phone, dial the DID (phone number) you assigned the inbound DISA.
• Key in the passcode you chose, followed by #.
• Key in the extension chosen for the spoofer application (31337 in the sample config) followed by # and wait for the prompt
• Then type up to 15 numbers to use as the caller id, followed by #. Wait for acknowledgement. You should pick an area code and exchange that actually exists otherwise many mobile devices and/or carriers will reject the Caller ID as unauthentic and display 'Unknown' on the device. To display international caller id may require adding a + and country code to the beginning. Depending on your carrier and trunk provider, you may need additional prefix digits such as 1 to make things work.
• Then type the number you wish to call, followed by #. Wait for acknowledgement.
• The number will be dialed now, with your chosen caller id, and you can have a nice conversation when your party picks up the phone.

To use a VOIP phone with the system:

• Install a SIP phone application such as Zoiper and follow the instructions below to set up the application.
• Place calls to and from whatever DID you choose to purchase from your SIP trunk provider.
• Dial *0 to get to the PBX internal DISA dialtone, from which you can call internal extensions and reach the spoofer application, or dial the extensions of any other VOIP phones local to your STUNT BANANA installation. Note that while SIP is encrypted here, the audio is still done over RTP which is -not- encrypted. To build a completely private SIP phone, we would need to add VPN capability to STUNT BANANA, at which point you might as well just install Wire.
• Zoiper lets you specify caller ID directly in the application, which is respected by STUNT BANANA specifically, so you can also easily perform ID spoofing that way as well. If the caller ID is not specified in Zoiper, the default caller id specified in the STUNT BANANA configuration for this device will be used.
• Text messaging is not supported at this time, however your SIP trunk provider likely has APIs and text-via-email support for allocated DIDs, you might want to set that up outside of STUNT BANANA for your users.

Setup

• Ensure you have an AWS account and you can log into it.

• You might want a mail server you can access to send voicemails from. If you need one, SendGrid works well and is cheap.

• Provision an Ubuntu instance, 18.04/Bionic works as of this writing. (t3.small should be sufficient in many cases). Other Debian-derived OS may work, but I can't guarantee anything.

• Generate an ssh key on that instance and ensure you can git clone from GitHub with it.

• Allocate the instance an external EIP

• Prepare a publicly-resolvable domain name for the EIP, with a DNS A record pointing to it.

• SSH into the box and set up the public name as the instance's hostname

  sudo hostnamectl set-hostname yourname.example.com
  sudo reboot
  

• Get yourself an account with a SIP trunk provider (such as QuestBlue, don't try to use Twilio for this)

  • provision a SIP trunk with the EIP you allocated for your instance
  • note down the SIP trunk provider's inbound IP address
  • allocate a DID you would like to use for the DISA in-dial. This is the number you'll be calling to get access to the spoofer service and any other dialplans you set up.
  • allocate any other DIDs you want to provision for SIP devices/VOIP apps/etc.

• Set up the following security group inbound rules (at least)

  ports	protocol	addresses	description
  10000-20000	udp	0.0.0.0/0,::/0	ports for rtp inbound
  5060	udp	sip.trunk.ip/32	SIP Trunk UDP inbound
  5061	tcp	0.0.0.0/0,::/0	SIP over TLS for devices
  5060	tcp	sip.trunk.ip/32	SIP Trunk TCP inbound
  80	tcp	0.0.0.0/0	used by Let's Encrypt for HTTP auth
  443	tcp	0.0.0.0/0	used by Let's Encrypt for HTTP auth

  The SIP TCP inbound rule may not be required by your SIP trunk provider. The 80/443 ports are intended for use by Let's Encrypt to produce SSL certificates used by SIP-over-TLS. This is to protect your devices' SIP accounts from being eavesdropped on or having their credentials stolen.

  You will also want any rules required to ssh into the machine from a trusted location and access to/from the local VPC if you're in one.

• SSH into the box and clone this repository recursively:

  git clone --recurse-submodules [email protected]:stuntbanana/stuntbanana.git
  cd stuntbanana
  

• Set up Let's Encrypt:

  ./setup-letsencrypt.sh [email protected]
  

  You'll be providing a contact email for the SSL certificate. If you don't want to use Let's Encrypt, skip this step, but you'll need to modify other configuration files to point to your SSL certificates, because you REALLY don't want to run SIP without encryption to your devices. If you get hacked you can easily end up with a $20,000 phone bill.

• Set up Asterisk:

  ./setup-asterisk.sh
  

  This may prompt you for what your country's international dialing code. US country code is '1'. If you're elsewhere, look it up and type it in.

  This will install Asterisk as a non-root 'asterisk' user. The default branch of Asterisk used is a custom fork that has some changes to pjproject's dns resolver. The goal of these changes and many of the default configuration files is to open the minimum number of network-facing TCP/UDP ports and minimize the attack surface. Given that RTP wants to use a large number of high ports for VOIP audio, requiring a large firewall hole, having pjproject DNS bind to a random high port is undesirable.

  Note to devs: As this is not a general-purpose installation of Asterisk, -many- of the modules in modules.conf have been disabled to proactively reduce the attack surface. Be aware that changing the dialplan or doing your own development on STUNT BANANA may require adding modules to this file. Good luck, sometimes it's not obvious which modules are required. Turning on autoload=yes in that file will help determine if a broken script requires a module.

• Configure STUNT BANANA:

  You will need to add a bunch of files by hand to the location /etc/asterisk/private to make things work. Templates you can use are available at the Configuration section of this document.

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Configuration

The Asterisk configuration for STUNT BANANA requires several configuration files to be added manually in the /etc/asterisk/private directory. These are #included into the main configuration at various points, and will contain private credentials that have no business in a git repository.

/etc/asterisk/private/default-trunk

remote_hosts = your.sip.trunk
outbound_auth/username=yourusername
outbound_auth/password=yourpassword

This file is in Asterisk Configuration File format.

Replace your.sip.trunk with the endpoint of your SIP trunk provider. (for QuestBlue this is sbc.questblue.com at the time of this writing)

Replace yourusername and yourpassword with the credentials required by your SIP trunk provider. (for QuestBlue this is name of your SIP trunk for both fields)

/etc/asterisk/private/from-internal

exten => 31337,1,Goto(spoofer,s,1)
exten => 1000,1,Gosub(internal-dial,s,1(endpoint_name))

This file is in dialplan format.

The first line is the extension to use for the spoofer tool.

The second line is an example of adding an extension for a SIP device. Replace 1000 with the desired extension, and endpoint_name with a name for the endpoint. If you have more than one SIP device to provision, you can duplicate this line as many times as necessary with the appropriate changes. If you have no SIP devices, you can remove the second line.

/etc/asterisk/private/from-pstn

exten => 1235551212,1,Goto(disa,s,1)
exten => 3214442323,1,Goto(from-internal,1000,1)

This file is in dialplan format.

The first line is the DID for the DISA system, telling calls to that number to go the DISA dialplan context.

This second line is for a SIP device, connecting the external phone number to the internal extension specified in the /etc/asterisk/private/from-internal file. If you have more than one SIP device to provision, you can duplicate this line as many times as necessary with the appropriate changes. If you have no SIP devices, you can remove the second line.

/etc/asterisk/private/passwd-disa

123456

This file is in plain text format.

Seriously, change this value to something secure. It's the password to your DISA system, in plaintext. If someone guesses this you're going to be unhappy. The password must be numeric as it will be keyed in on your phone's dialpad.

/etc/asterisk/private/phones

[endpoint_name](DefaultPhone)
inbound_auth/username=deviceusername
inbound_auth/password=devicepassword
endpoint/set_var=DEFAULT_CALLERID="Your Name"<3214442323>
endpoint/set_var=VOICEMAIL_BOX=1000

This file is in Asterisk Configuration File format.

Replace endpoint_name with the endpoint name you chose in /etc/asterisk/private/from-internal as well as choosing a username, password, and caller id to use for the line. The VOICEMAIL_BOX can be set to anything but keeping it the same as the extension chosen for this endpoint would ensure there are no conflicts later if you add more lines. The deviceusername and devicepassword you choose are going to be the ones you put into the VOIP application to log into the system. If you do not make deviceusername the same thing as endpoint_name you will have to specify it seperately in your voip configuration

/etc/asterisk/private/vm-boxes

1000 => 1234,Joe Blow,[email protected],,attach=yes|tz=pacific|delete=yes

This configuration file is in Asterisk voicemail.conf context format.

The format for each line is:

voicemailboxnumber => passcode, name, email, secondary email, options

secondary email does not need to be specified. passcode would be used if accessing the voicemail through a non-trusted path, though this is currently unused. The voicemailboxnumber should be the same as chosen in /etc/asterisk/private/phones for each SIP device. The email specified will receive copies of voicemails left for the device when busy or unanswered, as MP3 file attachments.

/etc/asterisk/private/vm-email

fromstring=STUNT BANANA
[email protected]

This configuration file is in Asterisk voicemail.conf [general] format.

The fromstring will be used as the name on the 'from' address for voicemail emails. The serveremail should be an address you can send email from via your email server.

Email Setup

To send email from STUNT BANANA you will need to set up a sendmail compatible mail transport on your STUNT BANANA installation. The setup scripts automatically install the ssmtp package, which provides sendmail compatibility without needing to run an email daemon. You will need an external email provider, one that works well is sendgrid.

To configure email, edit this file:

/etc/ssmtp/ssmtp.conf

mailhub=mailserver:port
UseSTARTTLS=YES
FromLineOverride=YES
AuthUser=authuser
AuthPass=authpassword
TLS_CA_File=/etc/ssl/certs/ca-certificates.crt

mailserver:port should be set to an email server you have access to. Avoid port 25 since you'll need to work with Amazon Support to get outbound access to that port. For sendgrid, it's smtp.sendgrid.net:587. authuser is your username or api key for the email server authpassword is your password or api secret for the email server

Zoiper Setup

A word of caution, there are -two- Zoiper apps in the Apple App Store. If you intend to purchase Zoiper Premium, you might be better off just getting the seperate Zoiper Premium app rather than subscribing via Zoiper Lite. The subscription to one will not get you access to the other.

If you have Zoiper Premium, you can turn on Incoming Calls/Push Notifications so your registered SIP phone will receive calls even when the app is sleeping. Push Transport should be set to TCP, as TLS transport for some reason -does not work-. This is only for the push notifications themselves not the SIP/RTP traffic.

Create an Account

• Account name can be anything but for convenience, you might want to name it the same thing as your DID, so you always remember which phone numer you're coming from.
• Your Domain should be the hostname of your STUNT BANANA installation.
• User name should be the endpoint_name chosen in /etc/asterisk/private/phones
• Password should be the devicepassword chosen in /etc/asterisk/private/phones
• Caller ID can be left blank to use the DEFAULT_CALLERID chosen in /etc/asterisk/private/phones or specified in either numeric or "Full Name"<Phone Number> format.
• Auth Username needs to be specified as deviceusername if it is not the same as endpoint_name
• Network Settings/Transports should be set to TLS. Enable whatever the app tells you to at this point.
• Network Settings/Protocol suite should be set to TLS v1
• Number rewriting should be turned on for your convenience unless it's giving you trouble