Elastix is a unified communications software that integrates the best tools available for Asterisk-based PBXs into a single, easy-to-use interface. It also adds its own set of utilities and allows the creation of third party modules to make it the best software package available for open source telephony. The goals of Elastix are reliability, modularity and ease-of-use. These characteristics added to the strong reporting capabilities make it the best choice for implementing an Asterisk-based pbx.
Elastix can easily deploy scalable IP-PBX solutions to the enterprise. Businesses can connect their distant branch locations, while maintaining all major PBX features available at their corporate office. Business headquarters, branch offices as well as remote workers and home offices can benefit from feature-rich productivity suite including voicemail, multi-line conferencing, call forward and many other features.
Monday, December 28, 2009
Digium
Digium® is the innovative force behind Asterisk®, the world's most widely used open source telephony platform. In addition, Digium develops the software, hardware, and other components you'll need to create a unified communications solution for a small or medium business.
Digium provides a variety of VoIP communications solutions to fit the needs of small, medium and large businesses. The Digium product line includes turn-key business phone systems as well as telephony interface cards, Asterisk software and other components needed to create custom telephony solutions.
Combine free Asterisk open source software, Digium interface telephony cards, and Asterisk components to build Custom Telephony Solutions. Asterisk can be configured as an IP PBX, a voice mail server, IVR server, VoIP gateway, call recorder, ACD and other telephony applications.
Asterisk was originally built as a PBX and today represents an astonishing 18% of global market for business telephone systems. The base feature set includes many of the most popular and powerful PBX functions.
VoIP Gateway
Tuesday, December 15, 2009
Asterisk - Digium
About Digium
Founded in 1999, Digium is the creator, sponsor, and innovative force behind Asterisk, the world's most popular open source telephony software.
Vision and Mission
Digium's vision is a world based upon open communications. Our mission is to transform the way businesses acquire and operate their communications systems through the application of open source software.
About Digium
Digium Inc., the Asterisk® Company, created, owns and is the innovative force behind Asterisk, the most widely used open source telephony software. Since its founding in 1999, Digium has become the open source alternative to proprietary communication providers, with offerings that cost as much as 80 percent less. Digium offers Asterisk software free to the open source community and offers Asterisk Business Edition and Switchvox IP PBX software to power a broad family of products for small, medium and large businesses. The company's product line includes a wide range of hardware and software to enable resellers and customers to implement turnkey VoIP systems or to design their own custom telephony solutions.
Used in combination with Digium's telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures. Digium's offerings include VoIP, conferencing, voicemail, legacy PBX, IVR, auto attendant, media servers and gateways, and application servers and gateways.
About Asterisk
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 TDM protocols for the handling and transmission of voice over traditional telephony interfaces, featuring VoIP packet protocols such as SIP and IAX 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.
Founded in 1999, Digium is the creator, sponsor, and innovative force behind Asterisk, the world's most popular open source telephony software.
Vision and Mission
Digium's vision is a world based upon open communications. Our mission is to transform the way businesses acquire and operate their communications systems through the application of open source software.
About Digium
Digium Inc., the Asterisk® Company, created, owns and is the innovative force behind Asterisk, the most widely used open source telephony software. Since its founding in 1999, Digium has become the open source alternative to proprietary communication providers, with offerings that cost as much as 80 percent less. Digium offers Asterisk software free to the open source community and offers Asterisk Business Edition and Switchvox IP PBX software to power a broad family of products for small, medium and large businesses. The company's product line includes a wide range of hardware and software to enable resellers and customers to implement turnkey VoIP systems or to design their own custom telephony solutions.
Used in combination with Digium's telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over IP, TDM, switched and Ethernet architectures. Digium's offerings include VoIP, conferencing, voicemail, legacy PBX, IVR, auto attendant, media servers and gateways, and application servers and gateways.
About Asterisk
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 TDM protocols for the handling and transmission of voice over traditional telephony interfaces, featuring VoIP packet protocols such as SIP and IAX 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 PBX
Asterisk is a complete telecommunications platform.
From caller ID to long distance, anything your telephone system can do, Asterisk can do better and cheaper.
Asterisk was designed to be able to do everything a traditional telephone system can do, and much, much more. Asterisk is, at it's heart, a PBX system. However, it includes a whole host of telephony features such as voicemail and call conferencing.
Asterisk creates a PBX that rivals the features and functionality of traditional telephony switches. Asterisk is cost-effective, low-maintenance, and flexible enough to handle all voice and data networking.
With Asterisk software, Telephony hardware, and a common PC, anyone can replace an existing switch or complement a PBX by adding VoiceOverIP, voicemail, conferencing, and many other capabilities. Asterisk integrates with analog phones and most standards-based IP telephone handsets and software.
Asterisk greatly reduces the cost of traditional telecommunication technology and operation, and moves voice over IP, VoIP, to the mainstream. Asterisk integrates a pre-existing analog telephone network with the benefits of IP technology, greatly reducing costs.
From caller ID to long distance, anything your telephone system can do, Asterisk can do better and cheaper.
Asterisk was designed to be able to do everything a traditional telephone system can do, and much, much more. Asterisk is, at it's heart, a PBX system. However, it includes a whole host of telephony features such as voicemail and call conferencing.
Asterisk creates a PBX that rivals the features and functionality of traditional telephony switches. Asterisk is cost-effective, low-maintenance, and flexible enough to handle all voice and data networking.
With Asterisk software, Telephony hardware, and a common PC, anyone can replace an existing switch or complement a PBX by adding VoiceOverIP, voicemail, conferencing, and many other capabilities. Asterisk integrates with analog phones and most standards-based IP telephone handsets and software.
Asterisk greatly reduces the cost of traditional telecommunication technology and operation, and moves voice over IP, VoIP, to the mainstream. Asterisk integrates a pre-existing analog telephone network with the benefits of IP technology, greatly reducing costs.
Monday, December 14, 2009
Asterisk Features List
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.
Asterisk Features List
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 & μ-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
MFC-R2 (Not supported. However, a patch is available)
PRI Protocols
4ESS
BRI (ISDN4Linux)
DMS100
EuroISDN
Lucent 5E
National ISDN2
NFAS
Q.SIG
Asterisk Features List
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 & μ-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
MFC-R2 (Not supported. However, a patch is available)
PRI Protocols
4ESS
BRI (ISDN4Linux)
DMS100
EuroISDN
Lucent 5E
National ISDN2
NFAS
Q.SIG
Asterisk Glossary
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.
Asterisk Glossary
ACD (Automatic Call Distributor) - A device or system that distributes incoming calls to a specific group of terminals that agents use. It is often part of a computer telephony integration (CTI) system.
CODEC (Coder/Decoder) - A software library that contains the algorithms necessary to convert an analog signal to and from a digital one. Examples: G.711 G.729 GSM
Context - The dialplan is composed of one or more extension contexts. Each extension context is itself simply a collection of extensions. Each extension context in a dialplan has a unique name associated with it. The use of contexts can be used to implement a number of important features, such as security, routing, autoattendant, multilevel menus, authentication, callback, privacy, macros, etc...
DAHDI (Digium Asterisk Hardware Device Interface) - A high density kernel telephony interface for PSTN hardware.
Dialplan - A dial plan establishes the expected number and pattern of digits for a telephone number. This includes country codes, access codes, area codes and all combinations of digits dialed. For instance, the North American public switched telephone network (PSTN) uses a 10-digit dial plan that includes a 3-digit area code and a 7-digit telephone number. Most PBXs support variable-length dial plans that use 3 to 11 digits. Dial plans must comply with the telephone networks to which they connect.
E&M (Ear & Mouth) A type of signaling commonly used over T1 and E1 interfaces.
Encode - The process of converting an analog signal into a digital signal that can be manipulated easily by a computer.
FXO (Foreign Exchange Office) - A device usually found on the customer end that is powered by the channel and can interface into the telephone company's network. Digium makes FXO modules that interface with PSTN lines using FXS signalling in either Loopstart(fxs_ls) or the more common Kewlstart(fxs_ks) modes.
FXS (Foreign Exchange Station) - A device usually located on the telephony company's property, a FXS device send power through a channel to a phone on the other end. Digium makes FXS modules that interface with PSTN phones using FXO signalling in either Loopstart(fxo_ls) or the more common Kewlstart(fxo_ks) modes.
G.711 - An uncompressed codec that samples a 64kbps channel at 8 bits per sample using pulse code modulation. The Two varients of G.711 are known formally as uLaw and aLaw.
G.729 - The G.729 codec is an industry standard which allows for stuffing more calls in limited bandwidth to utilize IP voice in more cost effective ways. A typical call consumes 64Kbps of voice bandwidth. G.729 reduces the call to 8Kbps (normal IP overhead adds to this number). Many people are using Asterisk with G.729 to replace expensive gateways.
GSM - A compressed speech codec that uses a rate of 13 kbps.
H.323 - A VOIP protocol that was deployed early and is widely adopted.
IAX (Inter-Asterisk eXchange) - A VOIP protocol designed to be much more NAT friendly. IAX currently only transports audio.
IVR (Interactive Voice Response) - An automated voice system that allows callers to navigate a phone system and be directed to the correct extension by pressing a series of numbers on a tuch-tone phone. (I.E. Push 1 for sales, push 2 for support, etc..)
MGCP (Media Gateway Control Protocol) - A VOIP Protocol that has both signaling and control and was designed to reduce complexity between media gateways.
Open source - An approach to the design, development, and distribution of software, offering practical accessibility to a software's source code.
PBX (Private Branch Exchange) - A telephone exchange that serves a particular business or office, as opposed to one that a common carrier or telephone company operates for many businesses or for the general public.
PRI (Primary Rate Interface) - A PRI is a truly digital circuit, containing 24 ISDN channels. One of these channels is a D channel and used for signaling. The rest are B channels and used to transport audio.
PSTN (Public Switched Telephone Network) - Originally a network of fixed-line analog telephone systems, the PSTN is now almost entirely digital and includes mobile as well as fixed telephones. The network works in much the same way that the Internet is the network of the world's public IP-based packet-switched networks.
REN (Ringer Equivalency Number) - A number which represents the ringer loading effect on a line. A ringer equivalency number of 1 represents the loading effect of a single traditional telephone set ringing circuit. Most modern telephones probably will have a REN lower than 1. The total REN expresses the total loading effect of the equipment on the ringing current generator (FXS). The REN of a Digium FXS board is 5 (representing "extension," i.e., parallel-connected telephones). The actual number of devices on the line may be greater than the REN limit, if their respective individual RENs are less than 1.
SIP (Session Initiation Protocol) - A signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). SIP adoption amongst hardware and software vendors continues to expand.
TDM (Time Division Multiplexing) - A processes of splitting one medium into two or more channels by using timed segments to transmit information.
Transcode The process of converting a channel with one type of encoding to a different type of encoding in real time.
VoIP (Voice Over Internet Protocol) - A general method for transporting voice through the internet.
Zaptel - The Zaptel project has been renamed 'DAHDI' as of May 2008. DAHDI is a series of drivers for telephony hardware devices.
Asterisk Glossary
ACD (Automatic Call Distributor) - A device or system that distributes incoming calls to a specific group of terminals that agents use. It is often part of a computer telephony integration (CTI) system.
CODEC (Coder/Decoder) - A software library that contains the algorithms necessary to convert an analog signal to and from a digital one. Examples: G.711 G.729 GSM
Context - The dialplan is composed of one or more extension contexts. Each extension context is itself simply a collection of extensions. Each extension context in a dialplan has a unique name associated with it. The use of contexts can be used to implement a number of important features, such as security, routing, autoattendant, multilevel menus, authentication, callback, privacy, macros, etc...
DAHDI (Digium Asterisk Hardware Device Interface) - A high density kernel telephony interface for PSTN hardware.
Dialplan - A dial plan establishes the expected number and pattern of digits for a telephone number. This includes country codes, access codes, area codes and all combinations of digits dialed. For instance, the North American public switched telephone network (PSTN) uses a 10-digit dial plan that includes a 3-digit area code and a 7-digit telephone number. Most PBXs support variable-length dial plans that use 3 to 11 digits. Dial plans must comply with the telephone networks to which they connect.
E&M (Ear & Mouth) A type of signaling commonly used over T1 and E1 interfaces.
Encode - The process of converting an analog signal into a digital signal that can be manipulated easily by a computer.
FXO (Foreign Exchange Office) - A device usually found on the customer end that is powered by the channel and can interface into the telephone company's network. Digium makes FXO modules that interface with PSTN lines using FXS signalling in either Loopstart(fxs_ls) or the more common Kewlstart(fxs_ks) modes.
FXS (Foreign Exchange Station) - A device usually located on the telephony company's property, a FXS device send power through a channel to a phone on the other end. Digium makes FXS modules that interface with PSTN phones using FXO signalling in either Loopstart(fxo_ls) or the more common Kewlstart(fxo_ks) modes.
G.711 - An uncompressed codec that samples a 64kbps channel at 8 bits per sample using pulse code modulation. The Two varients of G.711 are known formally as uLaw and aLaw.
G.729 - The G.729 codec is an industry standard which allows for stuffing more calls in limited bandwidth to utilize IP voice in more cost effective ways. A typical call consumes 64Kbps of voice bandwidth. G.729 reduces the call to 8Kbps (normal IP overhead adds to this number). Many people are using Asterisk with G.729 to replace expensive gateways.
GSM - A compressed speech codec that uses a rate of 13 kbps.
H.323 - A VOIP protocol that was deployed early and is widely adopted.
IAX (Inter-Asterisk eXchange) - A VOIP protocol designed to be much more NAT friendly. IAX currently only transports audio.
IVR (Interactive Voice Response) - An automated voice system that allows callers to navigate a phone system and be directed to the correct extension by pressing a series of numbers on a tuch-tone phone. (I.E. Push 1 for sales, push 2 for support, etc..)
MGCP (Media Gateway Control Protocol) - A VOIP Protocol that has both signaling and control and was designed to reduce complexity between media gateways.
Open source - An approach to the design, development, and distribution of software, offering practical accessibility to a software's source code.
PBX (Private Branch Exchange) - A telephone exchange that serves a particular business or office, as opposed to one that a common carrier or telephone company operates for many businesses or for the general public.
PRI (Primary Rate Interface) - A PRI is a truly digital circuit, containing 24 ISDN channels. One of these channels is a D channel and used for signaling. The rest are B channels and used to transport audio.
PSTN (Public Switched Telephone Network) - Originally a network of fixed-line analog telephone systems, the PSTN is now almost entirely digital and includes mobile as well as fixed telephones. The network works in much the same way that the Internet is the network of the world's public IP-based packet-switched networks.
REN (Ringer Equivalency Number) - A number which represents the ringer loading effect on a line. A ringer equivalency number of 1 represents the loading effect of a single traditional telephone set ringing circuit. Most modern telephones probably will have a REN lower than 1. The total REN expresses the total loading effect of the equipment on the ringing current generator (FXS). The REN of a Digium FXS board is 5 (representing "extension," i.e., parallel-connected telephones). The actual number of devices on the line may be greater than the REN limit, if their respective individual RENs are less than 1.
SIP (Session Initiation Protocol) - A signaling protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP). SIP adoption amongst hardware and software vendors continues to expand.
TDM (Time Division Multiplexing) - A processes of splitting one medium into two or more channels by using timed segments to transmit information.
Transcode The process of converting a channel with one type of encoding to a different type of encoding in real time.
VoIP (Voice Over Internet Protocol) - A general method for transporting voice through the internet.
Zaptel - The Zaptel project has been renamed 'DAHDI' as of May 2008. DAHDI is a series of drivers for telephony hardware devices.
Sunday, December 13, 2009
Open Source Asterisk PBX
Trixbox is an open telephony platform utilizing the best of the open source telephony tools into one easy-to-install package. Based on an enhanced LAAMP (Linux, Apache, Asterisk, mySQL, PHP) the trixbox dashboard provides easy to use web-based interfaces to setup, manage, maintain, and support an complete IP PBX system.
Succeeding Asterisk@Home 2.8, trixbox has been loosely described as 'Asterisk@Home 3.0'. It offers improved stability and the promise of an upgrade process that doesn't require you to wipe the entire install and start over.
Subscribe to:
Posts (Atom)