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Save on Calls!
Asterisk PBX is the most versatile, best value for money PABX.
Like any PBX, it allows attached telephones to make calls to one another, and to connect to other telephone services including the public switched telephone network (Telkom/Neotel/Premicell) with the added advantage of Voice over Internet Protocol (VoIP) services.

Business Benefits:

  • Low cost: Very low hardware and power requirements helps keep down the total cost of ownership
  • Low risk: Simple installation and maintenance means lower risks and greater system availability
  • Improved efficiency: Reducing the time it takes to make critical decisions, and keeping the entire team connected
  • Simplicity: One platform for all communication needs, accessible from from virtually anywhere

Exceptional Features:

  • Anywhere, anytime communications with mobility support
  • High quality, exceptional reliability with click-to-install simplicity
  • Complete High-Definition HD Voice IP Telephony platform with a feature rich IP PBX, including:
    • HD Voice SIP soft-phone
    • Inbound/Outbound Call handling (answer, hold, transfer, drop, record)
    • Conference Calls, adhoc and scheduled (add, drop, record conference, mute, un-mute, breakout)
    • Phone preferences, and presence management
    • Call management (voicemail, divert, group, transfer, rules)
    • Compatible with major software and hardware platforms


No need for extra cards or licensing to activate/install features such as Voicemail,Voicerecording or adding more extentions.

To add more extentions it is as easy as buy phone, configure and use!!!

Large companies with branches can now have all of their offices linked with one PBX,make free calls between branches and route calls via the least expensive route.


Coupled with Save on Calls! you can and will save thousands of rands.

Remember that because Asterisk is software based it will NEVER become Outdated. Lifelong software Upgrades is free .


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.

See the Asterisk Glossary for a list of terms.

Call Features

ADSI On-Screen Menu System
Alarm Receiver
Append Message
Automated Attendant
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
Fax Transmit and Receive (3rd Party OSS Package)
Flexible Extension Logic
Interactive Directory Listing
Interactive Voice Response (IVR)
Local and Remote Call Agents
Music On Hold
Music On Transfer:
- Flexible Mp3-based System
- Random or Linear Play
- Volume Control

Call Features

Predictive Diallers
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
VoIP Gateways
- Visual Indicator for Message Waiting
- Stutter Dial tone for Message Waiting
- Voicemail to email
- Voicemail Groups
- Web Voicemail Interface

Computer-Telephony Integration

AGI (Asterisk Gateway Interface)
Graphical Call Manager
Outbound Call Spooling
Predictive Dialler
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 Allows a unified dialplan across multiple offices




G.711 (A-Law & μ-Law)
G.723.1 (pass through)
G.729 (through purchase of a commercial license)

VoIP Protocols

SIP (Session Initiation Protocol)
IAX™ (Inter-Asterisk Exchange)
MGCP (Media Gateway Control Protocol
SCCP (Cisco® Skinny®)

Traditional Telephony Protocols

E&M Wink
Feature Group D
MF and DTMF support
Robbed-bit Signalling (RBS) Types
MFC-R2 (Not supported. However, a patch is available)

PRI Protocols

BRI (ISDN4Linux)
Lucent 5E
National ISDN2

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 - 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, auto attendant, multilevel menus, authentication, call-back, 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 signalling commonly used over T1 and E1 interfaces.

Encode - The process of converting an analogue 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 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 variants 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 touch-tone phone. (I.E. Push 1 for sales, push 2 for support, etc..)

MGCP (Media Gateway Control Protocol) - A VOIP Protocol that has both signalling 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 signalling. The rest are B channels and used to transport audio.

PSTN (Public Switched Telephone Network) - Originally a network of fixed-line analogue 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 signalling 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) - The Zaptel project has been renamed 'DAHDI' as of May 2008. DAHDI is a series of drivers for telephony hardware devices.

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