Noojee Receptionist for Asterisk provides a software-based interface for receptionists. It provides the ability to dispatch calls using a database of all staff and external parties or by phone or extension number. When looking up internal extensions, the status of each extension is displayed, saving the receptionist time as they no longer have to dial the number to determine if the party is available. When an extension is busy, the call can be placed into a "hold and transfer" mode that will automatically transfer the caller when the extension becomes available. It can also periodically inform the caller that the party is busy and gives the option to continue waiting, leave a voice message, or be put back through to the receptionist.
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.
Mediastreamer is a portable C library that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications. It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM, and AMR), video codecs (MPEG4, H263, H264, and Theora), sound card I/O, wav file streaming, webcam video capture, echo-cancellation, conferencing, parametric equalization, and various other utilities. It has a modular design that makes it extensible through plugins. This is the media-streaming component of linphone, a GPL SIP video phone.
Sipcat is a software IP telephony system that helps businesses increase productivity and reduce telecommunication costs. It allows you to start running your own voice over IP PBX in a matter of minutes. It contains a complete Linux distribution, including Asterisk and all required 3rd party software.
VoxForge collects transcribed user-submitted speech audio files (collectively called a "speech corpus") to create acoustic models for use with speech recognition engines such as HTK, Julius, ISIP, and Sphinx. The current focus is on collecting audio to create acoustic models for command and control applications on a PC, and for voice over IP telephony speech recognition applications, i.e. IVR (Interactive Voice Response).