Projects tagged ‘telephony’ and ‘voip’

Asterisk is a complete PBX and telephony toolkit in software. It runs on Linux, *BSD, MacOSX, and Solaris. It provides all of the features you would expect from a PBX and more as it enables ... [More] developers to build customized voice applications of many types. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. [Less]

Robust Open Source (GPL) SIP (RFC3261) server implementation. Among features: VoIP SIP registrar/proxy/router/application server, TLS secure communication, AAA, ENUM, LCR, load balancing, NAT ... [More] traversal, OSP, CPL, SNMP, SIMPLE IM&Presence, DNS failover. The source code for development is hosted now in GIT at http://sip-router.org [Less]

FreePBX is a full-featured PBX web application. If you've looked into Asterisk, you know that it doesn't come with any "built in" programming. You can't plug a phone into it and make it work without ... [More] editing configuration files, writing dialplans, and various messing about. FreePBX simplifies this by giving you pre-programmed functionality accessible by a user-friendly web interfaces that allows you to have a fully functional PBX pretty much straight away with no programming required. Some of the features include voicemail, IVR menus, conferencing, paging, ring groups, call routing, queues, and many more. [Less]

FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch. It can be used as a simple ... [More] switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow. We support various communication technologies such as SIP, H.323, IAX2 and GoogleTalk making it easy to interface with other open source PBX systems such as sipX, OpenPBX, Bayonne, YATE or Asterisk. FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols. [Less]

CallWeaver is a community-driven vendor-independent cross-platform open source PBX software project (formerly known as OpenPBX.org). It was originally derived from Asterisk. Now it supports analog and ... [More] digital PSTN telephony, multi-protocol voice over IP telephony, fax, software-fax, T.38 fax over IP and many telephony applications such as IVR, conferencing and callcenter queue management. [Less]

Yet Another Telephony Engine is a next-generation telephony engine; while currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice ... [More] , video, data and instant messaging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses. [Less]

OpenSIPS (Open SIP Server) is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. OpenSIPS, as a SIP server ... [More] , is the core component of any SIP-based VoIP solution. With a very flexible and customizable routing engine, OpenSIPS 'unifies voice, video, IM and presence services in a highly efficient way, thanks to its scalable (modular) design. What OpenSIPS has to offer, comes in a reliable and high-performance flavour - OpenSIPS is one of the fastest SIP servers, with a throughput that confirms it as a solution up to enterprise or carrier-grade class. [Less]

ZOIPER 2.0 Free SIP and IAX is a user-friendly softphone, compatible with Asterisk or any other SIP or IAX capable system. Great VoIP calling experience with features like: SIP + IAX / IAX 2 ... [More] protocols Available codecs – GSM, ulaw, alaw, speex, ilbc STUN support STUN server per account Two accounts DTMF tones sending Echo cancellation Codec settings per account Account password encryption Call history Hold function Quick dial panel Optional Automatic pop-up window for incoming call Call logs Minimize on tray Minimize on start up Always on top Adaptive Jitter Buffer Support for multiple audio devices Address book Quickdial pad Automatic user registration Call transfer Voice mail message information Portable ZoIPer with portable devices (like USB, flashcards, etc.) Multilanguage support [Less]

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. ... [More] 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. Since 1.1rc4, RTP play (voice and RFC2833 DTMFs) is also supported. [Less]

A2Billing complements the Asterisk project by enabling the following features on both TDM and VoIP calls: Traditional calling card services Credit limit on both pre-paid and post-paid customers ... [More] Callback services Residential VoIP services Wholesale minutes termination Monthly/weekly free calling packages Invoicing Paypal, Moneybookers and Authorize.net integration. The project is easy to use and is frequently seen on FreePBX installations to bring accountability to small offices' phone usage. For ITSP and traditional telco wholesale usage it has been seen to easily scale to millions of minutes per month, with 100,000s rates across many trunks. Work is in progress to further enhance A2Billing's scalability and availability. [Less]