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Asterisk

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company name:
Digium, Inc
license: Freeware
minimum requirements: No special requirements.
functional limitations:
Asterisk description
Asterisk is the world's most popular open source telephony project. Under development since 1999, Asterisk is free, open source software that turns 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.

Code for Asterisk, originally written by Mark Spencer of Digium, Inc., has been contributed from open source software engineers around the world. Currently boasting over two million users, Asterisk supports a wide range of telephony protocols. It includes rich support for the handling and transmission of voice over traditional telephony interfaces including analog lines, ISDN-BRI lines and digital T1/E1 trunks. Asterisk also features support for a wide range of VoIP protocols including SIP, IAX and H.323 among others. It supports U.S. and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.

Asterisk is released as open source under the GNU General Public License (GPL), and it is available for download free of charge. Asterisk® is the leading open source telephony project and the Asterisk community has been ranked as a key factor in the growth of VoIP.
What Does Asterisk Do?

Asterisk is like an erector set or a box of Legos for people who want to create communications applications. That's why we refer to it as a "tool-kit" or "development platform". Asterisk includes all the building blocks needed to create a PBX system, an IVR system or virtually any other kind of communications solution. The "blocks" in the kit include:

* Drivers for various VoIP protocols.
* Drivers for PSTN interface cards and devices.
* Routing and call handling for incoming calls.
* Outbound call generation and routing.
* Media management functions (record, play, generate tone, etc.).
* Call detail recording for accounting and billing.
* Transcoding (conversion from one media format to another).
* Protocol conversion (conversion from one protocol to another).
* Database integration for accessing information on relational databases.
* Web services integration for accessing data using standard internet protocols.
* LDAP integration for accessing corporate directory systems.
* Single and mult-party call bridging.
* Call recording and monitoring functions.
* Integrated "Dialplan" scripting language for call processing.
* External call management in any programming or scripting language through Asterisk Gateway Interface (AGI)
* Event notification and CTI integration via the Asterisk Manager Interface (AMI).
* Speech synthesis (aka "text-to-speech") in various languages and dialects using third party engines.
* Speech recognition in various languages using third party recognition engines.


This combination of components allows an integrator or developer to quickly create voice-enabled applications. The open nature of Asterisk means that there is no fixed limit on what it can be made to do. Asterisk integrators have built everything from very small IP PBX systems to massive carrier media servers.

Asterisk In Action: Key Applications

Asterisk As A PBX

Asterisk can be configured as the core of an IP or hybrid PBX, switching calls, managing routes, enabling features, and connecting callers with the outside world over IP, analog (POTS), and digital (T1/E1) connections.

Asterisk runs on a wide variety of operating systems including Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris and provides all of the features you would expect from a PBX including many advanced features that are often associated with high end (and high cost) proprietary PBXs. Asterisk's architecture is designed for maximum flexibility and supports Voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware.
Asterisk As A Gateway

It can also be built out as the heart of a media gateway, bridging the legacy PSTN to the expanding world of IP telephony. Asterisk’s modular architecture allows it to convert between a wide range of communications protocols and media codecs. Asterisk as a feature/media server.

Need an IVR? Asterisk’s got you covered. How about a conference bridge? Yep. It’s in there. What about an automated attendant? Asterisk does that too. How about a replacement for your aging legacy voicemail system? Can do. Unified messaging? No problem. Need a telephony interface for your web site? Ok.
Asterisk In The Call Center

Asterisk has been adopted by call centers around the world based on its flexibility. Call center and contact center developers have built complete ACD systems based on Asterisk. Asterisk has also added new life to existing call center solutions by adding remote IP agent capabilities, advanced skills-based routing, predictive and bulk dialing, and more.
Asterisk In The Public Network

Internet Telephony Service Providers (ITSPs), competitive local exchange carriers (CLECS) and even first-tier incumbents have discovered the power of open source communications with Asterisk. Feature servers, hosted services clusters, voicemail systems, pre-paid calling solutions, all based on Asterisk have helped reduce costs and enabled flexibility.
Asterisk Everywhere

Asterisk has become the basis for thousands of communications solutions. If you need to communicate, Asterisk is your answer.

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 advanced features that are often associated with large, high end (and high cost) proprietary PBXs.

Call Features

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

Call Features

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
Computer-Telephony Integration

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

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

Codecs

ADPCM
G.711 (A-Law & ?-Law)
G.722
G.723.1 (pass through)
G.726
G.729 (through purchase of a commercial license)
GSM
iLBC
Linear
LPC-10
Speex
VoIP Protocols

SIP (Session Initiation Protocol)
IAX™ (Inter-Asterisk Exchange)
H.323
MGCP (Media Gateway Control Protocol
SCCP (Cisco® Skinny®)
Traditional Telephony Protocols

E&M
E&M Wink
Feature Group D
FXS
FXO
GR-303
Loopstart
Groundstart
Kewlstart
MF and DTMF support
Robbed-bit Signaling (RBS) Types. VoIP Telephony Solutions - Digium®, The Asterisk Company. Digium - The Asterisk Company - provides Asterisk Business Edition telephony platform and Switchvox IP PBX software to power VoIP phone systems for small, medium and large businesses. Digium is the creator and primary developer of Asterisk, the industry's first open source telephony platform.
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