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.

A curated list of awesome Real Time Communications resources.

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.

HTML5 SIP client. Written in Javascript, throw it onto a page and configure it to contact a SIP gateway of some kind. Uses webRTC built into all modern web browsers. Does audio as well as video. Can contact the PSTN if the gateway can.

Github repo: https://github.com/DoubangoTelecom/sipml5

Javascript SIP client for VoIP. NodeJS backend. Uses WebRTC built into every modern browser. Found in the NPM collection, so it's easy to install (in theory).

A venerable FOSS SIP client with many possible user interfaces, include an HTML5 one that uses javascript. Also has a CLI version

Jitsi is a F/OSS multiprotocol text chat/voice over IP/video chat application. It supports a large number of instant messaging and VoIP protocols, not only for IM but as transport and control channels for VoIP and video conferencing. If there's a way to get your streams through it'll do it. Natively supports OTR for protecting chats. Implements SRTP and ZRTP for encrypted conferencing. Supports DNSSEC. Aims to replace Skype as a more secure, open source package.