Open Source Telephony

Cast Technologies deploys its Open Source Telephony solution based on the Asterisk® - the Open Source PBX!

Easily build your own multiprotocol PBX on Linux!

Asterisk is a complete PBX in software. It runs on Linux, BSD and MacOSX and provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.

Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response and Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway). Check the Features section for a more complete list.

Asterisk needs no additional hardware for Voice over IP. For interconnection with digital and analog telephony equipment, Asterisk supports a number of hardware devices, most notably all of the hardware manufactured by Asterisk's sponsors, Digium™. Digium has single and quad span T1 and E1 interfaces for interconnection to PRI lines and channel banks as well as a single port FXO card and a one to four-port modular FXS and FXO card.
Also supported are the Internet Line Jack and Internet Phone Jack products from Quicknet.

Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces. Asterisk supports US and European standard signaling types used in standard business phone systems, allowing it to bridge between next generation voice-data integrated networks and existing infrastructure. Asterisk not only supports traditional phone equipment, it enhances them with additional capabilities.

Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol, Asterisk merges voice and data traffic seamlessly across disparate networks. While using Packet Voice, it is possible to send data such as URL info and images in-line with voice traffic, allowing advanced integration of information.

Asterisk provides a central switching core, with four APIs for modular loading of telephony applications, hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network.

Asterisk is primarily developed on GNU/Linux for x/86. It is known to compile and run on GNU/Linux for PPC along with OpenBSD, FreeBSD, and Mac OS X Jaguar. Other platforms and standards-based UNIX-like operating systems should be reasonably easy to port for anyone with the time and requisite skill to do so. Asterisk is available in Debian Stable (in version 1.0.7), and the current maintainership is done by a team, the Debian VoIP Team.

Asterisk was originally written by Mark Spencer of Digium, Inc. Code has been contributed from open source coders around the world, and testing and bug-patches from the community have provided invaluable aid to the development of this software.

Asterisk is growing fast with new features added frequently to the CVS tree. Mark Spencer and numerous contributors from around the world contribute new code and patches on a daily basis. To stay up-to-date on the growing feature list of Asterisk, please visit Digium's website for more information on subscribing to the Asterisk mailing lists.

Asterisk® Features

Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk offers both classical PBX functionality and advanced features, and interoperates with traditional standards-based telephony systems and Voice over IP systems. Asterisk offers the features one would expect of a large proprietary PBX system such as Voicemail, Conference Bridging, Call Queuing, and Call Detail Records.

• ADSI On-Screen Menu System
• Alarm Receiver
• Append Message
• Authentication
• Automated Attendant
• Blacklists
• Blind Transfer
• Call Detail Records
• Call Forward on Busy
• Call Forward on No Answer
• Call Forward Variable
• Call Monitoring
• Call Parking
• Call Queuing
• Call Recording
• Call Retrieval
• Call Routing (DID & ANI)
• Call Snooping
• Call Transfer
• Call Waiting
• Caller ID
• Caller ID Blocking
• Caller ID on Call Waiting
• Calling Cards
• Conference Bridging
• Database Store / Retrieve
• Database Integration
• Dial by Name
• Direct Inward System Access
• Distinctive Ring
• Distributed Universal Number Discovery (DUNDi™)
• Do Not Disturb
• E911
• ENUM
• Fax Transmit and Receive (3rd Party OSS Package)
• Flexible Extension Logic
• Interactive Directory Listing
• Interactive Voice Response (IVR)
• Local and Remote Call Agents
• Macros
• Music On Hold
• Music On Transfer
• Flexible Mp3-based System
• Random or Linear Play
• Volume Control
• Predictive Dialer
• Privacy
• Open Settlement Protocol (OSP)
• Overhead Paging
• Protocol Conversion
• Remote Call Pickup
• Remote Office Support
• Roaming Extensions
• Route by Caller ID
• SMS Messaging
• Spell / Say
• Streaming Media Access
• Supervised Transfer
• Talk Detection
• Text-to-Speech (via Festival)
• Three-way Calling
• Time and Date
• Transcoding
• Trunking
• VoIP Gateways
• Voicemail
• Visual Indicator for Message Waiting
• Stutter Dialtone for Message Waiting
• Voicemail to email
• Voicemail Groups
• Web Voicemail Interface
• Zapateller

• AGI (Asterisk Gateway Interface)
• Graphical Call Manager
• Outbound Call Spooling
• Predictive Dialer
• TCP/IP Management Interface

• TDMoE (Time Division Multiplex over Ethernet)
• Allows direct connection of Asterisk PBX
• Zero latency
• Uses commodity Ethernet hardware
• Voice-over IP
• Allows for integration of physically separate installations
• Uses commonly deployed data connections
• Allows a unified dialplan across multiple offices