Open Source FreeBSD Telephony Software

Gammu is a cellular manager for mobile phones/modems. It contains libraries and functions for ringtones,logos,phonebook,SMS,etc. (used by external software), a command line version (with backup/restore) and SMS gateway (with MySQL and PostgreSQL supp

FusionPBX is a multi-tenant PBX and voice switch for FreeSWITCH, a highly scalable, multi-threaded, cross-platform communication platform. FusionPBX can be used either as a single or domain based PBX, and offers several functionalities: as carrier grade switch, call center server, voicemail server, fax server, conference server, VoIP server, multi-tenant appliance framework and so much more. It also offers unlimited extensions, device provisioning, music on hold, call parking, automatic call distribution and many other superb features. Give your company the business functionality and corporate level phone system it needs with FusionPBX!

Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.

Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

A lightweight cross platform IP telephony client using the IAX protocol, designed for use with the asterisk open source PBX.

New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on https://github.com/jselbie/stunserver More details on the project's website: http://www.stunprotocol.org

TurnServer is a implementation of Traversal Using Relay around NAT (TURN) protocol. This protocol allows a client to obtain IP addresses and ports from such a relay.

SCMxx allows you to exchange certain types of data with mobile phones made by Siemens Mobile. It works with the S25, S35i, M35i, C35i, SL45, S45 and ME45 and probably others.

pcapsipdump is libpcap-based SIP sniffer with per-call sorting capabilities. It writes SIP/RTP sessions to disk in a same format, as "tcpdump -w", but one file per SIP session (even if there is thousands of concurrent SIP sessions). Getting started: http://pcapsipdump.sf.net/

The project has moved! Please find current versions at https://www.gnugk.org/ The GNU Gatekeeper (GnuGk) is a full featured H.323 gatekeeper under GPL license. It supports VoIP and videoconferencing and includes Radius and database support and many call routing functions. The project has moved! Please find current versions at https://www.gnugk.org/

Bayonne is supported from Savanna.gnu.org. Please visit there. Sourceforge is generally depreciated except for mailing lists, but updated releases will be posted here from time to time.

4PSA VoipNow PHP Demo Client is an Open Source software written in PHP demonstrating the 4PSA VoipNow SOAP API interface. The application handles a client registration and a phone terminal creation.

The next generation of the Openh323 Gatekeeper project that provides high availability and scalibility for carrier grade, available freely under GNU GPL.

The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit

meicd - UNIX command line run daemon that connects to Nortel Meridian MAX MEI TCP/IP ports. Records PBX event information for use in generating reports and and tracking call flow. Simply decodes the protocol and dumps the data into flat files.

The OSP Toolkit is a client side implementation of the ETSI OSP VoIP Peering protocol (ETSI TS 101 321). The OSP Toolkit project was begun in 1998 and the code has been incorporated into many commercial and open source VoIP products.

Open Source SIP Back to Back User Agent) and Billing system. Has an Embeded TCL interpreter making session control and session mediation limitless in flexibility. Useful for: Routing Engine, Calling Cards, SBC, Peering, Wholesale and more!

A Web Control Panel Application for the OpenSIPS, which is intended for both system and user provisioning. It features more than 18 tools, covering important functionalities (MI,statistics) and modules (acc,siptrace,drouting,dialplan) of OpenSIPS. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips-cp .

A set of perl scripts which senses wether a particular bluetooth device is within range, also featuring sync with IrMC 1.1 compliant devices.

Phone Call Cost is a program that in the end, aims to be able to pull length of call time from a phone that displays such information and multiply it by a known rate (given to the program via the UI (or hard coded in by the end user)) and return the cost

SAMs emulates a basic answering machine. It maintains a single voicemail box that automatically detects faxes and is remotely accessable with a password. Also enables remotely activating an Internet connection, and changing the greeting.

This project has been relocated to http://github.com/sippy/rtpproxy http://github.com/sippy/b2bua

H323 voice & video proxy appropriated for border and zone controlling (without RAS). Designed for heavy load.

US mirror of answerd, a software answering machine. (Records a voice telephone message and emails it to you as a sound file.) The author of answerd is Vyacheslav Zamyatin (slawa@medinf.chuvashia.su)