Why choose an Asterisk based phone system from IPGate?

Asterisk is the world’s leading open source telephony platform. It’s a software product that can turn a general purpose computer into a sophisticated VoIP communications server.

Asterisk has become one of the most feature rich, scalable, sophisticated communication servers available today. Implemented as an on premise business phone system or as a hosted PBX, Asterisk remains free to download and comes fully featured with none of licensing restrictions of proprietary systems.


Maximum Functionality, Minimum Cost

Our IP-PBX have advanced features such as Unified Messaging (voice-mail, fax, email), IVR, multi-party conferencing, call recording, virtual fax, integration with instant messaging clients such as GTalk or Skype, support for multiple companies on one platform, remote extensions, reporting, web-based management etc. Ideal for small, medium or large organizations, our IP-PBX solutions provide an unrivalled feature set only found on high-end platforms at a price to suit the cost conscious enterprise or administration.

 

Integration with Existing Business Applications

Since our IP-PBX solutions are just another software application and are based on open bricks, integration with your CRM, ERP, websites, directories and other IT platform becomes a straightforward exercise.

 

We do the job with your IT team

Rather than doing the installation purely on our own, we work in collaboration with your IT team and involve its staff in the project. This gives them the autonomy to manage the IP-PBX platform by themselves.

 

No infrastructure disruption

Our approach is to integrate as much as possible with our customers' existing IT infrastructure. Our solutions run on standard server hardware (HP, Dell etc.) that is most of the time supplied by the customer. We also deploy in virtual server environments (VMware, KVM etc.). No appliances! We use the infrastructure our customers have invested in.

 

Get Started

 

Where Did It Come From?

The Asterisk project started in 1999 when Mark Spencer released the initial code under the GPL open source license. Since that time it has been enhanced and tested by a global community of thousands. Today Asterisk is maintained by the combined efforts Digium and the Asterisk community.

 

 

What Can You Do With Asterisk?

Asterisk is a framework for building multi-protocol real-time communications applications and solutions. Asterisk is to realtime voice and video applications as what Apache is to web applications: the underlying platform. Asterisk abstracts the complexities of communications protocols and technologies, allowing you to concentrate on creating innovative products and solutions.

You can use Asterisk to build communications applications, things like business phone systems (also known as PBXs), call distributors, VoIP gateways and conference bridges. Asterisk includes both low and high-level components that significantly simplify the process of building these complex applications. See the Asterisk Applications section for more examples.

 

What Makes Asterisk Different?

Asterisk is open source, which means you can get under the hood, see how it works and make any changes or enhancements you like. Asterisk is flexible and lets you define the a solution that truly fits your requirements. Asterisk is stable, reliable and in production on thousands of systems worldwide. Asterisk is free to use.

 

What Do I Need To Know To Use Asterisk?

It depends. The Asterisk framework itself is built by developers for developers. If you want to create applications and solutions with Asterisk you will need a working knowledge of Linux, script programming, networking and telephony.

If you're not a developer you can still take advantage of the power of Asterisk by using pre-packaged solutions built on Asterisk or by working with an Asterisk integrator or consultant. For a comprehensive list of pre-built solutions, see the Asterisk Exchange community marketplace.

Applications

 

The Swiss Army Knife Of Communications

Asterisk is in use today as the core engine of many communications applications. While business phone systems (also known as IP PBXs) are the most common, Asterisk includes components that allow it to serve a wide range of functions. This section of the Asterisk.org site is intended to help you understand how Asterisk influences some of the most common applications. Keep in mind that these are only a small sample of the thousands of things that have been built using Asterisk.

  • Business Phone System / IP PBX
  • VoIP Gateway
  • Voicemail Server
  • Conference Bridge
  • Call Center
  • IVR Server
Ultimate Flexibility

You might wonder how a single software package can serve so many different functions. The key is in Asterisk's modular design. Asterisk includes hundreds of components that can be combined to build amazing stuff. Combine the SIP channel, the PSTN interface channel and some Dialplan script and you have a gateway. Change the Dialplan to drop calls into a ConfBridge session and you have a conference server. Alter it once more to route calls into voice mailboxes and you have a voicemail server. Tie it all together and you have an amazingly powerful phone system.

Amazing Potential

With Asterisk, you have the potential to tie communications into any application or business function. The ultimate goal of Unified Communications is to build multi-modal communications capabilities into the applications you use. Asterisk makes this easy. Communications-enable your salesforce automation or CRM system using the Asterisk Manager Interface. Set your workforce free by adding mobility and remote worker capabilities. Up your customer service efficiency and delight your customers by implementing web-based callback and intelligent queuing.

 

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 advanced features that are often associated with large, high end (and high cost) proprietary PBXs. The list below includes a sample of the features available in Asterisk.

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
  • Flexible Extension Logic
  • Interactive Directory Listing
  • Interactive Voice Response (IVR)
  • Local and Remote Call Agents
  • Macros
  • Music On Hold.h
  • 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
Call Features
  • 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
  • Voice-over IP
  • Allows for integration of physically separate installations
  • Uses commonly deployed data connections
  • Allows a unified dialplan across multiple offices
Speech
  • Cepstral TTS
  • Lumenvox ASR
Codecs
  • ADPCM
  • CELT (pass through)
  • G.711 (A-Law & μ-Law)
  • G.719 (pass through)
  • G.722
  • G.722.1 licensed from Polycom®
  • G.722.1 Annex C licensed from Polycom®
  • G.723.1 (pass through)
  • G.726
  • G.729a
  • iLBC
  • Linear
  • LPC-10
  • Speex
  • SILK
VoIP Protocols
  • Google Talk
  • H.323
  • IAX™ (Inter-Asterisk eXchange)
  • Jingle/XMPP
  • MGCP (Media Gateway Control Protocol
  • SCCP (Cisco® Skinny®)
  • SIP (Session Initiation Protocol)
  • UNIStim
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)
ISDN Protocols
  • AT&T 4ESS
  • EuroISDN PRI and BRI
  • Lucent 5ESS
  • National ISDN 1
  • National ISDN 2
  • NFAS
  • Nortel DMS100
  • Q.SIG

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.