CapiSuite is an ISDN telecommunication suite providing easy-to-use telecommunication functions which can be controlled from Python scripts. It uses a CAPI-compatible driver for accessing the ISDN-hardware, so you'll need a CAPI-compatible driver. CapiSuite is distributed with two example scripts for call incoming handling and fax sending. They provide a multi-user answering machine with fax recognition, remote inquiry, and fax receiving/sending functions.
ADM (Asterisk Desktop Manager) aims to integrate your desktop with the Asterisk PBX and hardware IP phone by providing some useful features such as automatic on-call volume reduction, one click dialing (from the clipboard), CRM integration via a browser popup, BlueTooth presence detection and automatic call redirection when you walk out of the office, and transfer of the current call from the desktop.
Appkonference is a high performance voice/video conferencing system for Asterisk. It is a fork of appconference, and it focuses on reliability and scalability. Appkonference has been tested on both Asterisk 1.4 and 1.6.X. Both Asterisk installations supported more than 1200 participants at a time.
SipExchange is a softswitch that provides standard SIP services like location, proxy, and presence. Using the SipExchange application, service providers can offer VoIP telephone services to their subscribers as well as other services based on voice, video, and instant messaging. SipExchange supports many of the standard subscriber features offered by the traditional telephone exchanges and PBXs. In addition, SipExchange supports external call control capabilities which service providers and software developers can use to create new features and services rapidly and plug them into the SipExchange application. SipExchange works with standard SIP phones that adhere to the SIP protocol standards. Its software architecture is flexible, scalable, and easily extensible. It runs as an enterprise application inside the JBoss server and takes advantage of many services that a J2EE server provides. SipExchange provides a portal-based user interface with which system administrators can manage subscribers and features as well as perform other routine operations. From the portal, subscribers can manage their profiles, view the call detail records, and customize the features to which they have subscribed. Service providers can easily add additional content to the portal and customize the look and feel.
Generic JTAPI, JAIN JCC, and JAIN Jcat is a framework to allow for the rapid development of Java telephony specification implementations. It does this by reducing the "service provider" coding requirements by an order of magnitude and by providing common services. Also included are sample service providers, including layered providers that can be combined with others to support multiplexing, JAIN JCC support, JAIN Jcat (preliminary), and and remote access.
Voxpak is a GUI for playing, editing, renaming, etc. voice and fax messages. It includes scripts for popping up sticky-notes or requesters with caller ID info, renaming voice messages to date+callerid, using Python and pyGTK. A small Kaptain script is also available as an alternative.
The Alamin GSM SMS Gateway is a group of daemons that allows you to send/receive SMS messages from any GSM device that supports AT+ commands (GSM modems or GSM mobile phones) or supported by Gnokii. A client program allows you to send messages from any IP client. An SMTP interface is provided to allow MTAs to send SMS directly to the GSM network. IMP (Incoming Message Processor) modules allows you to extend functionality to implement banking, network administration, bd querys, etc. from a GSM mobile phone.
Callweaver is a community driven software PBX project. The most important differences between Callweaver and Asterisk are built-in STUN support, the use of SpanDSP for better codecs and full T.38 fax over IP support, Sqlite instead of Berkeley DB, universal jitterbuffer, POSIX timers to avoid Zaptel timing dependencies, greater speed, more efficient dialplan execution, and greater stability.
Flite For Asterisk provides the "Flite" dialplan application, which allows you to use the Flite TTS Engine with Asterisk. This module invokes the Flite TTS engine locally, and uses it to render text to speech. It supports voice selection and 8kHz or 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs. It works with Asterisk 1.6.x, 1.8, and 10.