Asterisk, converts an ordinary computer into a feature-rich voice communications server. Asterisk makes it simple to create and deploy a wide range of telephony applications and services, including IP PBXs, VoIP gateways, call center ACDs and IVR systems. It is maintained by Debian VoIP Team.
Asterisk supports a wide range of protocols for the handling and transmission of voice over traditional telephony interfaces including H.323, Session Initiation Protocol, Media Gateway Control Protocol, and Skinny Client Control Protocol. Asterisk merges voice and data traffic seamlessly across disparate networks.
It has drivers for VoIP protocols and PSTN interface cards and devices. It routes both inbound and outbound calls. Call details and duration are logged for billing. Calls could be recorded and monitored. It recognizes speech and converts to text. LDAP integration is provided for corporate security. It stores all information in a relational database.
Asterisk has support of IVR, Conference bridge, Automated attendant, Call Queuing, Call Recording, Call Retrieval, Caller ID Blocking, SMS Messaging, VoIP Gateways and lot more.
Hardware requirement: Asterisk requires no additional hardware. It turns an ordinary computer into a communications server. It runs on GNU/Linux, OpenBSD, FreeBSD, and Mac OS X.
API Support: External call management is possible in any programming or scripting language through Asterisk Gateway Interface.
|Tags||communication framework ip-telephony pbx voip call-center|
Activa brings the Asterisk IP PBX to the call center. Built on top of Asterisk, Activa components enable successful call center implementations adding value in areas such as computer telephony, screenpopamp;click2dial, agent control, automatic dialing...
Ekiga (formely known as GnomeMeeting) is an open source SoftPhone, Video Conferencing and Instant Messenger application over the Internet. It provides Audio and Video free calls through the internet. It supports standard telephony features like Call Hold, Call Transfer, Call Forwarding, Call Histroy and Call Monitoring.voip skype-alternative internet-phone sip chat
FusionPBX can be used as a single or domain based multi-tenant PBX, carrier grade switch, call center server, fax server, VoIP server, voicemail server, conference server, voice application server, multi-tenant appliance framework and more. FreeSWITCH™ is a highly scalable, multi-threaded, multi-platform communication platform. We provide several avenues for you to get your system up and running on your own and learn the basics of the system.
FATS - FATS is a Twisted and Fast Asterisk's Telephony Services. Project contains implementation of FastAGI, AMI protocols for the Twisted framework. Using it you can develop fast and pretty services for the Asterisk IP-PBX.
PNX system develops the middle-ware source code to glue Asterisk with a number of powerful telephony products such as: 1) OpenH323 H.323 stack 2) Vovida SIP stack 3) Bayonne Voice Automation Platform The advantages? An advanced IP PBX supporting the wide
A complete contact center solution for the Asterisk PBX, capable of handling inbound and outbound campaigns. Provides an agent toolbar for CTI. For managers an admin panel controls every aspect of the call center. Stunning historical and realtime stats.
Kamailio (successor of former OpenSER and SER) is an open source implementation of a SIP Signaling Server. SIP is an open standard protocol specified by the IETF. The core specification document is RFC3261.kamailio sip voip webrtc volte iot telephony sip-server
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.
This software suite is designed to extend the functionality of the Asterisk PBX through platform-independant web-client applications. Includes the VICIDIAL inbound/outbound call center application. The suite is scalable across multiple Asterisk servers.
asterCRM is a call center software for asterisk based VoIP system, also it has some CRM functions. It provide useful features such as pop-up, predictive dialer, click to call, extension status .... astercrm could work with all asterisk based system.
A lightweight cross platform IP telephony client using the IAX protocol, designed for use with the asterisk open source PBX.
Adhearsion is an open-source voice application development framework. Adhearsion users write applications atop the framework with Ruby and call into their code. Adhearsion rests above a lower-level telephony platform, for example Asterisk, FreeSWITCH or Voxeo PRISM, and provides a framework for integrating with various resources, such as SQL, LDAP and XMPP (Jabber).
Linphone is an internet phone or Voice Over IP phone (VoIP), it helps to communicate freely with people over the internet, with voice, video, and text instant messaging. Linphone is available for Linux, Windows, MacOSX, and for mobile phones: Android, iPhone, Blackberry. It is using SIP protocol and it is compatible with any voip operator using SIP for its network.voip skype-alternative internet-phone sip xmpp chat
The PoC telephony application suite is a set of scripts and libraries for interactive telephony environments such as the Asterisk PBX and Yate.
sipXcom open source unified communications software is the standard for enterprise voice solutions. Voice, chat, IVR, presence and more in one application. It provides complete enterprise grade telephony, unified messaging, Instant messaging, audio conferencing, and many other features. sipXcom can be run along side your existing Cisco, Avaya, Mitel, Shortel or other PBX while you migrate users to the sipXcom system.sip telephony unified-messaging audio-conferencing pbx voice-communication ivr
Routr – a lightweight sip proxy, location server, and registrar that provides a reliable and scalable SIP infrastructure for telephony carriers, communication service providers, and integrators. It also provides with capabilities that are suitable for the enterprise and personal needs. To get involved in the development of this project please contact us at @fonoster.sip webrtc voip instant-messaging ims proxy registrar asterisk kamailio freepbx yate freeswitch fusionpbx sip-server
VoIP Honey project provides a set of tools for building an entire honeynet, thus includes honeywall and honeypot emulating VoIP environments such as Asterisk PBX or OpenSer with fully configurable connections.
Longing for the good old days of Asterisk@Home? Welcome back to the steroid-enhanced version. PBX in a Flash 2.0 is the latest Lean, Mean Asterisk Machine, a high-performance, turnkey Asterisk PBX that's easy to upgrade. Features include CentOS 6.4 and your choice of Asterisk 1.8, 10, or 11 with FreePBX 2.9, 2.10, or 2.11. All versions include Google Voice for free calling throughout the U.S. and Canada at least through the end of 2013. 32-bit and 64-bit ISOs are available for download as
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