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Glossary of terms

ACD

Automatic Call Distribution is a feature used to route calls in a call center environment to the appropriate person based on factors such as availability, call usage, skills, time, etc.

Agent

An agent is a person on an extension that is a member of a queue.

ATA

An analog telephone adapter allows a traditional analog phone to be used on an Asterisk phone system by converting the analog signals to VoIP and connecting to the Asterisk system as a SIP extension.

AT

Network Address Translation allows for a network to have its own private IP space and still communicate to the Internet through a single public IP address.

BRI

Basic Rate Interface is a communication standard that is used on ISDN connections.

CDR

Call Data Records are the call logs that are kept by the system showing the logging of all of the calls into and out of the system.

Codec

A Codec is a piece of code that encodes or decodes audio using a given type of algorithm.

Conference Bridge

A conference bridge is an feature of the phone system that creates a meeting room that multiple callers can be in to have a multiple-person conference call.

DID

This stands for Direct Inward Dial which refers to the phone number that can be dialled to place a call into your PBX system.

DISA

Direct Inward System Access is a method by which you can connect into the phone system from the outside and have access to functions as if you were connected via an extension.

Firewall

A device that exists at the border of your network and policies the traffic that traverses the network.

Follow-Me

This feature allows a user to configure their extension to send calls to multiple destinations such as their extension and their cell phone.

FXO

The Foreign eXchange Office is the port that connects to an analog phone line. FXO ports are commonly found on IP Gateways and IP PBX / Asterisk Hardware.

FXS

The Foreign eXchange Station is the port that connects to an analog phone device. FXS ports are commonly found on IP Gateways and IP PBX / Asterisk Hardware.

G.711 Ulaw

G.711 is an uncompress audio format thereby providing the best call quality but the highest bandwidth usage.

G.729

The G.729 codec provides excellent call quality and very low bandwidth but requires a license for each channel that will be used.

GSM

GSM is a codec that provides for good call quality as well as very low bandwidth usage.

Hard Phone

This is a hardware phone device. You can browse Hard Phones in our IP Phones section.

IAX

The Inter-Asterisk-eXchange protocol was developed as a simpler and easier-to-manage alternative to the SIP protocol. IAX can traverse NAT far better than SIP can. The IAX protocol is not an industry standard but is native to Asterisk.

iLBC

The iLBC codec provides very good call quality with decent bandwidth usage.

Intercom

An intercom system allows you to call another extension and have their extension automatically answer on speakerphone for a two-way conversation.

ISDN

Integrated Services Digital Network. While use of ISDN circuits has become increasingly rare in the US, it is still very popular in other countries and provides some PRI type features to small to medium businesses. ISDN interfaces are commonly found on IP Gateways and IP PBX / Asterisk Hardware.

ITSP

An Internet Telephone Service Provider is a company that provides inbound or outbound call termination to your PBX over the Internet rather than requiring you to go through a traditional telephone company.

IVR

The Interactive Voice Response system is the menus that are given to your callers that help route the caller to the appropriate destination.

MPLS – Multi Protocol Label Switching

MPLS creates virtual links between distant nodes.  It is highly scalable directs and carries voice and data packets.

Paging

A paging system can user overhead speakers or can page through the speakerphones of the phone devices and allows for a the person who initiated the page to broadcast a one-way message.

PBX

The Private Branch Exchange is the phone system in your office that is the interface between the extensions in your office and the different connections to telephone service providers. Voiplink sells IP PBX's, a newer version of PBX that integrates telecommunications with digital networks. You can learn more about IP PBX's in our IP Phone Systems category.

POTS

This acronym stands for Plain Old Telephone Service, the commonly used analog phone lines found in houses and small-medium businesses. POTS is an analog system and is controlled by electrical loops. It is provided by copper wires run to homes and businesses and is often the cheapest and easiest telephone service to implement. To connect a POTS line to an IP PBX an IP Gateway an IP PBX interface card is commonly used.

Predictive Dialer

A predictive dialer is a software application for outbound calling that monitors the agents that are on calls, anticipates when the next agent will be available, and calls the next number in the least trying to maximize the use of the outbound calling agents.

PRI

Primary Rate Interface is a digital communication standard that is brought in over T1 or E1 circuits. To connect a PRI line to an IP PBX an IP Gateway an IP PBX interface card is commonly used.

PSTN

The Public Switched Telephone Network is the traditional public phone network that carries all traditional phone calls.

Queues

A call queue is typically a very high end PBX feature that allows callers to go into a holding area in the order that they called in and wait for an available agent. A queue is typically used when there are peak times when there are more inbound calls coming in than there are agents available to take the call.

Ring Groups

A ring group is a group of extensions that can be set to ring at the same time. A ring group is usually used when there are typically more agents available to take calls than there are calls coming in.

SIP

Session Initiated Protocol is the industry standard for voice over IP communications. SIP is the common protocol today used for communication between IP PBX's and IP Phones.

SIP Trunks – Session Initiation Protocol

A SIP trunk is a service offered to connect a PBX using VOIP on an internet connection, resulting in cost savings.

Soft Phone

Unlike a hard phone, a soft phone is a software program that you run on your computer that emulates all of the functions of a normal phone and allows you to receive and place phone calls from your computer.

T1/E1

This is the common digital circuit for medium to large businesses. T1 and E1 circuits are a digital service and offer more features than ISDN or POTS lines. A T1 circuit can handle 24 voice calls while an E1 circuit can handle 32 voice calls. To connect a T1 or E1 line to an IP PBX an IP Gateway an IP PBX interface card is commonly used.

Trunk

A trunk is a group of channels between two distinct points. This can be either between PBXs within a corporation or between an organization and a telephone service provider.

VoIP

This is the acronym for any voice traffic that uses Internet protocols. In a typical Asterisk setup, the phones will communicate over VoIP to the phone system and the phone system may communicate over VoIP to an Internet Telephone Service Provider.

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