GSMLIB is a library to access GSM mobile phones through GSM modems. Features include: modification of phonebooks stored in the mobile phone or on the SIM card, reading and writing of SMS messages stored in the mobile phone, sending and reception of SMS messages. Additionally, some simple command line programs are provided to use these features.
Sofia-SIP is a SIP user agent library, compliant with the IETF RFC3261 specification. It can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication services. The primary target platform is GNU/Linux. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center.
ADM (Asterisk Desktop Manager) aims to integrate your desktop with the Asterisk PBX and hardware IP phone by providing some useful features such as automatic on-call volume reduction, one click dialing (from the clipboard), CRM integration via a browser popup, BlueTooth presence detection and automatic call redirection when you walk out of the office, and transfer of the current call from the desktop.
GYach Enhanced is a feature-rich, improved version of the original Gyach. It is the first Yahoo! client for Linux with voice chat capabilities. It offers almost all of the features you would expect to find in the official Windows Yahoo! client. The program offers support for chat, conferences, buddy lists, and My Yahoo content. In addition, Gyach Enhanced offers many features not available in the official Yahoo! clients for Windows, Mac, and Linux. Webcam support is under development and planned for the future. Unlike the original Gyach, GYach Enhanced is designed for Linux only.
Callweaver is a community driven software PBX project. The most important differences between Callweaver and Asterisk are built-in STUN support, the use of SpanDSP for better codecs and full T.38 fax over IP support, Sqlite instead of Berkeley DB, universal jitterbuffer, POSIX timers to avoid Zaptel timing dependencies, greater speed, more efficient dialplan execution, and greater stability.
Sipp is a performance testing tool for the SIP protocol. It includes a few basic SipStone user agent scenarios (UAC & UAS) and establishes and releases multiple calls with the INVITE and BYE methods. It also reads XML scenario files describing any performance testing configuration. It features the dynamic display of statistics about running tests, periodic CSV statistics dumps, TCP, UDP, or TLS over IPv4 or IPv6 over multiple sockets or multiplexed with retransmission management, regular expressions and variables in scenario files, conditional branching, and dynamically-adjustable call rates. RTP play (voice, video, and RFC2833 DTMFs) is also supported.