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 Asterisk Open Source PBX Phone System What is Asterisk?
Asterisk® is a complete IP PBX in software. It 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® is released as open source under the GNU General Public License (GPL), meaning that it is available for download free of charge. Asterisk® is the most popular open source software available, with the Asterisk Community being the top influencer in VoIP. Supported Platforms Asterisk® is primarily developed on GNU/Linux for x/86 and runs on GNU/Linux for PPC along with OpenBSD, FreeBSD, and Mac OS X. 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. Supported Hardware 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 Digium®, the creator of Asterisk®. Features Asterisk-based telephony solutions offer a rich and flexible feature set. Asterisk® offers both classical PBX functionality and advanced features which interoperates with traditional standards-based telephony systems and Voice over IP systems. Supported Protocols Asterisk® supports a wide range of protocols for the handling and transmission of voice over traditional telephony interfaces including H.323, Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP). Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol Asterisk® merges voice and data traffic seamlessly across disparate networks. The use of Packet Voice allows Asterisk® to send data such as URL information 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.
Main Asterisk Feature Definitions | Voicemail | This module allows you to configure Voicemail for a user or extension | | Follow Me | Much like a ring group, but works on individual extensions. When someone calls the extension, it can be setup to ring for a number of seconds before trying to ring other extensions and/or external numbers, or to ring all at once, or in other various 'hunt' configurations. Most commonly used to ring someone's cell phone if they don't answer their extension. | | IVR | Creates Digital Receptionist (aka Auto-Attendant, aka Interactive Voice Response) menus. These can be used to send callers to different locations (eg, "Press 1 for sales") and/or allow direct-dialing of extension numbers. | | Misc Destinations | Allows creating destinations that dial any local number (extensions, feature codes, outside phone numbers) that can be used by other modules (eg, IVR, time conditions) as a call destination. | | Queues | Creates a queue where calls are placed on hold and answered on a first-in, first-out basis. Many options are available, including ring strategy for agents, caller announcements, max wait times, etc. | | Ring Groups | Creates a group of extensions that all ring together. Extensions can be rung all at once, or in various 'hunt' configurations. Additionally, external numbers are supported, and there is a call confirmation option where the callee has to confirm if they actually want to take the call before the caller is transferred. | | Time Conditions | Creates a condition where calls will to one of two destinations (eg, an extension, IVR, ring group..) based on the time and/or date. This can be used for example to ring a receptionist during the day, or go directly to an IVR at night. | | Conferences | Allow creation of conference rooms (meet-me) where multiple people can talk together. | | Music on Hold | Uploading and management of sound files (wav, mp3) to be used for on-hold music. | | PIN Sets | Allow creation of lists of PINs (numbers for passwords) that can be used by other modules (eg, trunks). | | Paging and Intercom | Allows creation of paging groups to make announcements using the speaker built into most SIP phones.
Also creates an Intercom feature code that can be used as a prefix to talk directly to one person, as well as optional feature codes to block/allow intercom calls. | | Recordings | Creates and manages system recordings, used by many other modules (eg, IVR). | | Call Forward | Allows a user to automatically forward incoming calls to any number | | Call Waiting | Provides an option to turn on/off call waiting | | Do-Not Disturb | A user can restrict their line from ringing and send all calls directly to voicemail | | Info Services | Provides a number of applications accessable by feature codes: company directory, call trace (last call information), echo test, speaking clock, and speak current extension number. | | DISA | DISA Allows you 'Direct Inward System Access'. This gives you the ability to have an option on an IVR that gives you a dial tone, and you're able to dial out from the freePBX machine as if you were connected to a standard extension. It appears as a Destination. | | Call Back | Allows you to setup a callback destination that calls a user back and provides access to an application. An example of this would be a caller that dials your system, disconnects, is called back and then provided a DISA application to make a phone call. |
Asterisk Full Feature 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
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
| 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 | Protocols
IAX™ (Inter-Asterisk Exchange)
H.323
SIP (Session Initiation Protocol)
MGCP (Media Gateway Control Protocol
SCCP (Cisco® Skinny®) Traditional Telephony Interoperability
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 | |
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