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Asterisk PABX Features

All the flexibility of Open Source

Asterisk has all the featues that you would expect from an enterprise PABX and is growing every day. One of Asterisk's  great strengths is that it is designed to configurable and extensible. Proprietary systems are not nearly as flexible and in the unlikely event that you cannot find the feature you need listed below then Asterisk can be extended through custom programming either through the creation of dial plans, programming to the documented interfaces or through modification of the Asterisk product itself.

 

FeatureExplanationRequirements

ADSI On-Screen Menu System

 

 

Automated Attendant

Allows callers to be automatically transferred to a user's extension without the intervention of a receptionist. e.g. select 1 for sales, 2 for support...

 

Blacklists

A list of persons or organisations that have incurred disapproval or suspicion and therefore penalised.

 

Blind Transfer

Ability to transfer a call to another extension without the need to wait for the other person to pick up.

 

Call Details Record

A log of all calls made including: source no., destination no., call duration, date, time etc

Third party database if records are to be stored in format other than text file.

Call Forward on Busy

Calls are automatically forwarded to another extension or phone if the phone is busy.

 

Call Forward on No Answer

Calls are automatically forwarded to another extension if not answered after a defined number of rings.

 

Call Monitoring

Allows a supervisor to listen in on a phone conversation.

 

Call Parking

Allows a call to be 'parked' on a virtual extension and then picked up by a third party by dialling in to that extension.

 

Call Queuing

Allows multiple calls to be placed in a queue and answered by the next available operator.

 

Call Recording

Recording of a phone conversation for later playback.

Significant amounts of disk space may be used (approx. 16KB/sec of recording).

Call Retrieval

Page a person who then can pick up the call.

 

Call Routing

Directs the call to the correct extension based on a Direct In Dial (DID) number.

 

Call Snooping

Allows a supervisor to listen in on active calls across a group of phones.

Privacy Policy for staff.

Call Transfer

Transfer a call to another extension.

 

Call Waiting

Allows switching between two simultaneous calls.

 

Caller ID

Displays the callers phone number on the phones screen.

Remote phone must send caller's ID.

Caller ID Blocking

Block a call based on the callers phone number.

 

Call ID on Call Waiting

See the phone number of a second caller whilst talking to the first caller.

 

Calling Cards

 

 

Conference Bridging

Create a conference call between multiple parties at multiple locations using different phone types; e.g. conference call a local extension, remote fixed line, mobile and VoIP connection all in one conference.

 

Database Store/Retrieval

Store call information in a database for later retrieval.

Third party database such as MySQL.

Database Integration

Ability to access a database during a call to provide such information as a customers outstanding balance.  e.g. Asterisk can whisper the callers balance or name in your ear just before putting the caller through.

Third party database.

Dial by Name

Dial a customer by their name using the phones numeric keypad (using the same technique used when sending an SMS).

 

Direct Inward System Access

 

 

Distinctive Ring

Different Rings based on caller ID or selection the caller makes whilst interacting with the auto attendant.

Handset must support multiple ring tones.

Distributed Universal Number Discovery (DUNDI?)

DUNDI is a peer-to-peer system for locating internet gateways to telephony services.

 

Do Not Disturb

Incoming calls are automatically routed to Voicemail, reception or alternate extension.

 

E000

Emergency Services Number

Requires VoIP provider who supports E000

ENUM

Electronic Telephone Numbers Mapping.

 

Fax Transmit/Receive

Via AsterFax

Limited Faxing built in to Asterisk.  Full featured faxing requires AsterFax.

Flexible Extension Logic

 

 

Interactive Directory Listing

Allows inbound callers to lookup a person's extension by their name.

 

Advanced system for call handling allows callers to navigate a menu system by dialling a number offered from a range of options. e.g. Please dial 1 for sales, 2 for admin....  Allows for creation of systems such as making credit card payments, entering of timesheets etc.

 

Local and Remote Call Agents

Staff can log onto the PABX from any phone using a login ID allowing them to make and take calls as if they were in their office.

 

Music on Hold

No external device is required.  Ships with a reasonable selection of tracks.

Music loaded onto server. Asterisk ships with basic selection.

Music on Transfer

Music plays when transferring calls between extensions

As per Music on Hold

Predictive Dialler

Used by outbound call centres.  Starts dialling a number based on statistical model of when it is expected that the next agent will be available.

Third party add on such as 'Vicidial'.

Privacy Manager

If a remote callers ID is blocked, the Privacy Manager prompts the call to manually enter their phone number which can then be used to check against a blacklist or whitelist. The call may then be blocked or let through as appropriate.

 

Overhead Paging

Allows an overhead speaker to be 'dialled into' and an announcement made.

Specialised paging device.

Protocol Conversion

Allows the interconnection of disparate phone networks: e.g. connect a Telstra call to a VoIP call.

Standard protocols supported include: TDM, SIP, H.323, LAX, SCCP.

Remote Call Pickup

Allows a call to be picked up at a remote location.

 

Remote Office Support

Ability to connect phones located in a remote office to the office PABX as local extensions.

Internet or VPN connection at both locations.

Open Settlement Protocol (OSP)

OSP provides a way for ISPs to support billing for voice and fax over IP services.

OSP compliant billing package.

Roaming Extensions

Individuals are given an extension number rather than phones having an extension.  A person logs into the phone in the morning (using a PIN) and all calls to their extension are then routed through to that phone.  Single sign in is also available, so that logging into Windows will also log you into the phone next to the computer.

Single Sign On requires an O/S dependant script. 

Route by Caller ID

The ability to connect a call to a given extension, call queue or group of extensions based on the callers phone number.

 

SMS Messaging

 

Requires an SMS Gateway.

Spell/Say

The ability to have the 'text' read or spelt to the caller. E.g. read out an email.

 

Streaming Media Access

Streaming Media is the rapid transmission of audio and video in packets over the internet.

 

Supervised Transfer

A call transfer made by an automatic device such as a Voice Response Unit which attempts to determine the result of the transfer - answered, busy, ring but no answer - by analysing call progress tones on the line.

 

Talk Detection

Ability to detect when a person is talking.  Useful for detection of answering machines.

 

Text-to-Speech

The ability to have the 'text' read to the caller

Festival - Open Source 'Text-to-Speech' engine

Three-Way Calling

Connect three people into a mini conference call

 

Time and Date

Have the time and date read to the caller

 

Transcoding

Convert between the different methods of compressing voice.

 

Trunking

Connect to the PSTN. e.g. connect a standard telephone line (analog or digital) to the PABX

Requires a trunk card such as Digium TDM400.

VoIP Gateways

A network device that converts voice and fax calls, in real time, between PSNT and an IP network.

 

VoiceMail

Ability to record a message from a caller when you are away from your desk.  Includes ability to deliver the voicemail message via email as well as the standard flashing light on your phone.

 

Computer Telephony Integration

- AGI (Asterisk Gateway Interface

- Graphical Call Manager

- Outbound Call Spooling

- Predictive Dialler

- 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 dial plan across multiple office locations

 

Codecs

- ADPCM

- G.711 (A-Law & μ-Law)

- G.723.1 (pass through)

- G.726

- G.729 (via purchase of commercial license)

- GSM

- iLBC

- Linear

- LPC-10

- Speex

 

Protocols

- LAX? (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 Signalling (RBS) Types

- MFC-R2

 

PRI Protocols

- 4ESS

- BRI (ISDN4Linux)

- DMS100

- EuroISDN

- Lucent5E

- National ISDN2

- NFAS