2600.network is a public service for dial-up users. It's purpose is to allow users of old, vintage, and outdated hardware to dial in with real modems to real systems.

This guide will explain some ways to set up a VoIP ATA so that you can place calls between computers with modems (although any other pair of telephone devices will work.) When done, you will be able to:

• Connect two or more computers with modems directly together using a single ATA
• Connect a computer with a modem to another computer and modem elsewhere in your house over your LAN using two ATAs
• Connect a computer with a modem to another one in someone else's house over the internet

Note that this does NOT involve setting up Asterisk!

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. 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.

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.

The scripts clone, compile, and install Asterisk for you, so if you want to use this with an existing Asterisk install it's going to take some hacking.

A fully featured browser based WebRTC SIP phone for Asterisk.

This web application is designed to work with Asterisk PBX. Once loaded application will connect to Asterisk PBX on its web socket, and register an extension. Calls are made between contacts, and a full call detail is saved. Audio Calls can be recorded. Video Calls can be recorded, and can be saved with 5 different recording layouts and 3 different quality settings. This application does not use any cloud systems or services, and is designed to be stand-alone. Additional libraries will be downloaded at run time (but can also be saved to the web server for a complete off-line solution).

Just what it says on the tin. Assumes a Digital Ocean droplet and you have root access.

OnionComms is server configuration to host chat applications over Tor using onion services. Servers supported:

• Mumble
• Asterisk
• Prosody
• Ejabberd
• Ricochet

A utility to automate the installation, maintenance, and debugging of Asterisk/DAHDI, while integrating additional patches to provide the richest telephony experience. Useful for: Automating installation and maintenance of Asterisk, Asterisk Test Suite, Asterisk Test Framework, DAHDI Linux, DAHDI Tools, and related resources; validating Asterisk configuration; finding common syntax errors in dialplan code; finding missing audio files referenced by the Playback, BackGround, and Read applications; suggesting optimizations that can be made to dialplan code to make it more readable and efficient; generating Asterisk user documentation; debugging Asterisk configuration; generating core dumps, automating PhreakNet boilerplate dialplan installation.

Primarily supported on Debian-based Linux systems. Support has also been added for FreeBSD. Pull requests to add support for other Linux distros or BSD are welcome.

A simple v.23 modem including the data link layer. This is an application for Asterisk. Place a call over SIP and it'll connect via Telnet to a pre-configured service in the dialplan.

This is F.O.B. (Flexible Orange Box), inspired by the popular S.O.B. (Software Orange Box) program for Windows.

When connecting SIP FXS devices with a Class 5 switch, the ATA will not see a Call Waiting presented to it when there is a Call Waiting. Thus, it is necessary to signal the FSK directly to the CPE in-band from the switch. Asterisk does not have any provision to do this, so this needs to be done with an external program.

This program is intended to be a legitimate Type II Caller ID Generator, used for the purpose of Call Waiting Caller ID (Of course, functionally, it can be used just like any other orange box to spoof call waitings if desired - we are not responsible for any misuse of this program). This allows you to send Call Waiting Caller ID to a remote endpoint, even if no Call Waiting is presented to the remote endpoint (e.g. Analog Telephone Adapter), allowing for CWCID to be provided even when advanced bridging capabilites are being used, by "orange boxing" in band for legitimate purposes.

dialup.world is (currently) a three-line dial-up ISP!

Sounds for Asterisk, recorded by Pat Fleet (the original voice of Ma Bell).

A webapp for administering Asterisk from a web browser. Written in PHP. Worked on recently. Asterisk's API doesn't change very much so there probably doesn't need to be. Backed by MySQL. No obvious documentation so it'll need to be messed with to get installed.

A webapp which uses WebRTC and Javascript to test whether or not your STUN or TURN server is working.

Github: https://github.com/webrtc/samples/tree/gh-pages/src/content/peerconnection/trickle-ice

A Voip Wardialer for the phreaking of 2020.

A curated list of telco resources and projects.

A new implementation of STUN and TURN. Full IPv6 support. Supports server authentication with the REST APi by the RFC. Implemented with Erlang.

V.23 Softmodem for Asterisk with some Bildschirmtext-specific stuff in it. Pretends to be a modem but it actually sets up a telnet-like TCP session to an IP address. Like many things with Asterisk, you have to compile everything from source with this module in a particular location.

A curated list of awesome Real Time Communications resources.

A small hobby software PBX for the tildeverse. So far users can make calls, leave voicemails and participate in a multi-user conference with more features to come. The numbers are loosely broken down by tilde, with each tilde having its own prefix "area code." It's set aside for tilde users only, and is not connected to the PSTN - you have to be an active tilde user just to get the admin's attention.

Baresip is a portable and modular SIP User-Agent with audio and video support. Tries to be a Swiss Army knife for SIP and VoIP. Supports encryption. Has an embedded web server with an HTTP (REST?) API for controlling the utility. Modular architecture.