IP Telephony

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Move past the POTS

Plain Old Telephone Service (POTS) has not changed much in the last 100 years. Sure, you have Call Waiting and Caller ID, but what you really need is a modern phone system.

There are many proprietary PBX Vendors but when you buy one of their systems, you are overpaying for a system that locks you in to their way of thinking.

Asterisk: The IP Telephony Revolution

Asterisk is an open-source PBX that runs in software on an inexpensive commodity PC. It is highly flexible and scalable.

Connections to the outside world

Asterisk can connect to the outside world through POTS, supporting up to 24 lines in a single PCI card, T-1 interfaces, supporting up to 96 lines in a single PCI card or through a SIP proxy, where your available lines are limited only by your Internet bandwidth.

Internal Phones

Asterisk can provide internal dial tones to plain analog phone sets, with up to 24 internal lines provided by a single PCI card. It can also provide service to your SIP phones, whether physical phones or handsets. The number of SIP phones is limited only by the Memory and Processing power of your host PC and the available network bandwidth. On a switched 100Mbps Ethernet network with a modern host PC, thousands of concurrent internal lines are feasible.

Features

Great, so I can connect all my phones, but what will it do for me? How is this for a 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 (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:

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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