VOIP PBX Designing and Implementation

Our features are of the deepest and strongest!
We gave life to our first virtual PBX in 2008 and we’ve been continuously adding capabilities since...

Creative People offers solutions to ensure that once your VoIP phones are up and running, they work efficiently and effectively to best meet the needs of your business.

 

The investment in VOIP SIP trunks can give a quick and substantial return-on-investment (ROI), since with a SIP trunk connected directly to the telephony service provider, the end user can dispense with costly ISDN BRIs and PRIs, replacing them with service that can cost significantly less. In most cases the PBX must be an IP-based PBX, communicating with all endpoints over IP, but it may just as well be a traditional digital or analog PBX. The requirements are that an interface for SIP trunking connectivity is available (Ethernet interface) and SIP protocol is supported.

 

We gave life to our first virtual PBX in 2008 and we’ve been continuously adding capabilities since. Our features are of the deepest and strongest, including professional greetings, music on hold, custom auto-attendant, call forwarding, fax with no fax machine, queuing, call recording, conferencing and…the list goes on and on!


 

 

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.

See the Asterisk Glossary for a list of terms.

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
Music On Transfer:

  • Flexible Mp3-based System
  • Random or Linear Play
  • Volume Control

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 Hold Music
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

Asterisk Gateway Interface (AGI)
Asterisk Manager Interface (AMI)
Asterisk REST Interface (ARI)
Outbound Call Spooling

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

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
GSM
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.

 

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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.