Asterisk - IP telephony commuincation product suitable for call center

  •        5622

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.



Related Projects

Activa for Asterisk

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 - Software Phone and Video Cnferencing Application

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.


Enterprise telephony recording and retrieval system

Fast Asterisk's Telephony Services

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.

AACC - Asterisk Advanced Call Center

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.

asterisk-php-libs - PHP Classes for Asterisk AMI and AGI interfacing

verbose ( $line, [ $level = 1 ] )---------------------------------Sends the Verbose AGI command with the first argument as the content. The optional second argument is the verbose level (default: 1) to set.answer ()---------Tells Asterisk to answer the call/channelhangup ()---------Tells Asterisk to hangup the call/channelami.php (class AMI)===================A simple class for building AMI-based PHP applications.The host (or IP) for the Asterisk box may be passed as an argument to the class co

AsteriskPNX 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

Kamailio - The Open Source SIP Server

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.


Asterisk is free software that transforms a computer into a communication server. Asterisk lets you build applications like PBX, VoIP gateway, IVR and ACD.

Kiax - an IAX Softphone

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.

Asterisk GUI client

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.

Linphone - Video SIP phone for Desktop and Mobile

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.


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

libasterisk-parseconfig-perl - parse and check syntax config files of IP-PBX Asterisk

parse and check syntax config files of IP-PBX Asterisk

PoC Telephony Applications

The PoC telephony application suite is a set of scripts and libraries for interactive telephony environments such as the Asterisk PBX and Yate.

SipXcom - Unified Communications System

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.

VoIP Switch Download

VoIP Switch is a software platform that serves different purposes and

Mumble - High Quality Voice Chat Software

Mumble is a cross-platform voice over IP application. It is a client - server application where multiple users could use the same server and all their communication is encrypted. It could be used while gaming. It does audio video recording.

Voip Honey

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.