Open Source BSD Internet Phone Software
Sort By:
Internet Phone Software for BSD
Browse free open source Internet Phone software and projects for BSD below. Use the toggles on the left to filter open source Internet Phone software by OS, license, language, programming language, and project status.
-
Gen AI apps are built with MongoDB AtlasMongoDB Atlas is the developer-friendly database used to build, scale, and run gen AI and LLM-powered apps—without needing a separate vector database. Atlas offers built-in vector search, global availability across 115+ regions, and flexible document modeling. Start building AI apps faster, all in one place.
-
The #1 AI-Powered eLearning PlatformTransform your content into engaging eLearning experiences with Coursebox, the #1 AI-powered eLearning authoring tool. Our platform automates the course creation process, allowing you to design a structured course in seconds. Simply make edits, add any missing elements, and your course is ready to go. Whether you want to publish privately, share publicly, sell your course, or export it to your LMS, Coursebox has you covered.
-
1
trixbox CE is an easy to install, VOIP phone system based on the Asterisk PBX. trixbox is designed for home or office use. trixbox CE includes CentOS linux, mysql, and all the tools needed to run a business quality phone system. (formerly asterisk@home)
">
-
2
Elastix
Unified Communications Server
Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android. -
3
Open Phone Abstraction Library (OPAL) is a C++ multi-platform, multi-protocol library for Fax, Video & Voice over IP and other networks. Also included is the Portable Tool Library (PTLib) which is a C++ multi-platform abstraction library and collection o
-
4
Sipp is a performance testing tool for the SIP protocol. Its main features are basic SIPStone scenarios, TCP/UDP transport, customizable (XML-based) scenarios, dynamic adjustment of call-rate and a comprehensive set of real-time statistics.
-
We help you deliver Virtual and Hybrid Events using our Award Winning end-to-end Event Management PlatformEventsAIR have been anticipating and responding to the ever-changing event industry needs for over 30 years, providing innovative solutions that empower event organizers to create successful events around the globe.
-
5
This project implements a simple STUN server and client on Windows, Linux, and Solaris. The STUN protocol (Simple Traversal of UDP through NATs) is described in the IETF RFC 3489, available at http://www.ietf.org/rfc/rfc3489.txt
-
6
Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).
-
7
Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like. The project moved to https://github.com/chan-sccp/chan-sccp
-
8
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.
-
9
This project has been superseded by OpalVoip (https://sourceforge.net/projects/opalvoip/) and H323Plus (https://sourceforge.net/projects/h323plus/) The OpenH323 project provides full featured, interoperable, Open Source implementation of the ITU H.323 teleconferencing protocol that can be used by personal developers and commercial users without charge.
-
Pylon is an All-in-one B2B Support Platform for modern B2B businesses.We bring together everything a post-sales teams team needs including a ticketing system, B2B omnichannel integrations (Slack Connect, Microsoft Teams), modern chat widget, knowledge base, AI support bot, account management, customer marketing, and more.
-
10
IHU is a VoIP application for Linux (using Qt and Speex), with low latency, crypted stream, minimal use of bandwith, and without intermediary servers. It is the easiest way to talk real-time with your friends (like phone) on the internet or LAN.
-
11
-
12
">
-
13
A lightweight cross platform IP telephony client using the IAX protocol, designed for use with the asterisk open source PBX.
-
14
The Cisco IP Telephony Services project contains many IP phone services and utilities for the Cisco 79XX phones and CallManager. This project houses several scripts, services, and administration tools. Recruiting! Contact me and become a developer.
-
15
-
16
GNU Gatekeeper (GnuGk)
H.323 Gatekeeper for VoIP and videconferencing
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/ -
17
This is a Java wrapper for Ffmpeg. The JMF (Java Media Framework) plugins system lets one use JMStudio or other application to play H.263/rfc2190 streams. Support for H.263+ and MPEG-4 is currently under test, please contact us for further information.
-
18
Open79XX XMLDirectory is an open PHP-based XML menuing system for providing on-screen services to the Cisco 79XX Series IP Phones. It contains dynamically generated phone directories, on-screen memos, links to other services, and many more options.
-
19
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.
-
20
Voxy - the simplest way to integrate VoiceXML applications in Asterisk. Configure your dial plan with the URL of your VoiceXML application and it's done.
-
21
ISDN CAPI Channel driver (chan_capi) for the Asterisk Open Source VoIP Platform. THIS PROJECT HAS BEEN MOVED TO chan-capi.org
-
22
CafeSip - Look what Java and SIP can do
A suite of open-source tools and frameworks for creating SIP apps
Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java. -
23
TelDaBase ist ein php/mySQL-Skript zur Aufteilung und Auswertung der Telefonrechnung (Rechnung Online der Deutschen Telekom AG) eines gemeinschaftlichen Telefons (z.B. WG,Studentenwohnheim); asterisk cdr-Datensätze werden optional automatisch ausgewerte
-
24
The XTapi project is aimed at someday growing up to be a reference implementation for JTAPI. At this time it is not the intention to cover every JTAPI package, just the Core and Media packages are covered.
-
25
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.