Abandoned Call- A call or other type of contact to a call center that is abandoned before the call is answered. Calls in which the caller hangs up almost immediately may or may not be counted as abandoned. When configuring settings on an ACD a minimum length of time can be set before it would be counted as an abandoned call. Incoming abandoned calls are most likely the result of longer wait times that callers will not endure.
Agent- A Call Center Agent is a person who actively handles incoming incoming or outgoing phone calls for a business. Each agent can be associated with a peripheral and can be a member of one or more skill groups, allowing for a skills based routing of calls.
Agent out call- An outbound call made by an agent.
(ATB) All Trunks Busy- The state of a trunk group when all trunks are in use. The trunk group cannot accept any new inbound or outbound calls in this state. The ICR will track the amount of time during which all trunks in a trunk group are busy.
Analog- The traditional (older) predominant method of providing residential and small-business telephone service where a voice conversation is carried over a single pair of copper wires.
Announcement- A recorded verbal message played to a caller.
Answer wait time- The elapsed time from when a call is received to when it is actually answered. Answer wait time is the sum of delay time, queue time, and ring time.
Answered call- A call is counted as answered when it reaches an agent or was routed to an ACD queue.
Area code- A three-digit prefix code used to indicate the destination area for long distance or "toll" calls in North America. Also known as Numbering Plan Area (NPA).
Asterisk- An Open Source PBX system that is recognized as the standard by the open source development community and is available under the GPL licence. It has proved to be a stable, reliable and robust telephone exchange system that supports hardware which will connect IP phones and devices to traditional analogue and ISDN telephone lines and has the ability to accomplish and integrate with numerous feature sets.
(ACD) Automatic Call Distributor- A programmable device at a call center that routes and manages incoming calls to targets or agents within that call center. An Intelligent Call Router determines the target or agent for a specific call, and then that call is sent to the ACD associated with that target. The ACD must then complete the routing as determined by the ICR. Many companies offering sales, service and/or support use ACDs to validate callers, make outgoing responses or calls, forward calls to the right party, allow calls to be recorded, gather usage statistics, balance the use of phone lines, and provide other services.
(ANI) Automatic Number Identification- A feature that provides the billing phone number of the phone from which a call originated or the phone number itself.
(ATA) Analog Telephone Adapter- A device that coverts traditional analog voice signals to digital signals which can then be transmitted over the Internet typically using an Ethernet RJ45 connection.
(BRI) Basic Rate Interface- An Integrated Services Digital Network (ISDN) configuration that consists of two bearer channels (B channels) of 64 kpbs each which van be bonded for a total of 128 kpbs and one data channel (D channel) of one 16 kpbs. The B channels are used for voice or user data and the D channel is used for any combination of data, control or signaling.
Bufferbloat- is an excessive amount of latency that is caused when a router or switch is configured to use extremely large buffers. Packets in a first-in first-out system become queued for long periods causing higher latency degrading VoIP and other time sensitive applications and the entire network.
Busy Lamp Field- A set of lights or LEDs, found primarily on an attendant console that visually indicates which phones on the system are in use. (CDR) Call Record Detail- Details about a specific call that includes duration, origination, destination, and billable information, as well as other pertinent information.
Call Center- A communications solution centered on inbound and/or outbound calls that uses PBX features such as attendants, queues, predictive dialing, etc., to manage voice calls.
Chat- Chat messaging differs from traditional texting in that it facilitates real-time, live synchronous communication rather than one-way exchanges. It enables back-and-forth conversations, informal interactions, and instant exchanges of messages, including multimedia content and group chats.
(CLID) Calling Line ID- Information about the billing telephone number from which a call originated. The CLID might be the entire phone number, the area code, or the area code plus the local exchange (NPX).
Carrier- A company that provides telecommunications circuits. Carriers include the local telephone companies and larger telephone providers like AT&T, Verizon, Sprint and Level3.
CentOS (Community ENTerprise Operating System)- CentOS Linux is an enterprise class Linux server distribution software derived from sources freely provided by Red Hat. It is used as the operating system for some Asterisk based IP-PBXs.
(CO) Central Office- The local office of the telephone company, where the switching equipment is housed that can switch calls locally or to long-distance (LD) carrier phone offices. The subscriber home and business lines are connected on what is called a local loop.
Codec- A term that comes from the enCOder/DECoder or Compressor-Decompressor process used for software or hardware devices that can convert a data stream. Two VoIP codecs often used are G711, a non-compressed codec, and G729, a codec that uses compression to lower bandwidth requirements.
Contact Center- An omni-channel communications solution to manage voice calls, and other types of communications, like chat, SMS, and social media apps. Along with advanced PBX features, a contact center offers integration with CRM apps and other tools which offer instantaneous information to allow agents to better service customers as an essential part of the communications solution.
Convergence- Referred to in the telecom and IT world as the integration or connection of different systems. Convergence will allow for communication and "meeting" of separate systems to allow for better efficiencies.
(DHCP) Dynamic Host Configuration Protocol- is a communications protocol that lets network administrators centrally manage and automate the assignment of Internet Protocol (IP) addresses, subnet masks, default gateway, and other IP parameters as nodes come online, as opposed to manually configuring static IP addresses for each device. Servers or routers can act as DHCP servers, but typically only one such device should actively be used at any given time to assign IP addresses on a LAN. This method for dynamically managing and automating the assignment of IP addresses lets a network administrator supervise and distribute IP addresses from a central point and automatically send a new IP address. DHCP uses the concept of a lease or amount of time that a given IP address will be valid for a node.
(DID) Direct Inward Dialing- A service that allows an enterprise to allocate individual phone numbers to each person within its PBX system. Offered by telephone companies for use with their customers' PBX systems, they allocate a range of numbers all connected to their customer's PBX. As calls are presented to the PBX the number that the caller dialed is also given, so the PBX can route the call to the desired person or other feature within the organization.
(DNS) Dynamic Name Service- is an IP standard hierarchical naming system to translate a name into a numerical IP address for the purpose of locating and addressing these devices world-wide.
(DSL) Digital Subscriber Line- Phone technology that allows a broadband internet digital connection to be carried over existing copper phone lines, while still allowing the phone service carry analog signals over the same line.
(DTMF) Dual-Tone Multi-Frequency- is a set of 16 tones used for in-band signaling between phone switches and telephones. Each key that is pressed generates two tones at specific frequencies, so that a voice or regular sound transmissions cannot mistakenly imitate the tones. One tone is generated from a high-frequency group and the second tone from a low frequency group.
FreePBX- is a standardized implementation of Asterisk and is based around a graphical web-based configuration interface and other tools which can be used to configure Asterisk as an alternative to directly editing text files.
FSX (Foreign eXchange Subscriber)- An RJ11 port used to connect in premise to an analog office phone, fax machine or other analog device.
FXO (Foreign eXchange Office)- designates a telephone signaling interface (port) that receives POTS, aka plain old telephone service. An IP-PBX appliance requires the use FXO ports to accept each active copper telephone line from the telco service provider.
H.323- An international standard for multimedia communication over packet-switched networks, including LANs, WANs, and the Internet. It was designed primarily for IP networks with multipoint voice and video conferencing capabilities, but may also operate over other packet-switched networks. H.232 addresses call control and management for both point-to-point and multipoint conferences, as well as gateway administration of media traffic, user participation and bandwidth.
High Availability- A characteristic of a system which aims to ensure a higher level of uptime operational performance, and/or for a higher than normal lifetime period. By eliminating single points of failure and by using redundant network designs incorporating almost seamless crossover in the event of a failure, systems can keep in operation without a failure that causes a loss of service.
Hosted VoIP- (also known as a Cloud Based PBX,
and/or hosted PBX) is where the PBX hardware and software
that handle calls for a business or person resides with a provider
off-site on the Internet. Through signaling, calls and other
communications are initiated and routed to other parties. The advanced
features, such as forwarding, music on hold, auto-attendants,
conferencing, etc., are handled by the provider's software and hardware
rather than on premise with the business.
Benefits associated with a Hosted VoIP service rather
than a traditional phone system, like an on-premise IP-PBX, are lower initial costs. A Hosted VoIP service
costs much less in time and money to set-up than an on-premise PBX. In many cases, hosted VoIP providers
offer no set-up fees for a
hosted VoIP system and a business can be up and taking calls in a matter of hours.
(IAX) Inter-Asterisk eXchange protocol- An Asterisk PBX protocol, (Now most commonly refers to IAX2), that usually carries both signaling and data on the same path and is used to enable VoIP connections between Asterisk servers as well as client-server communication.
(ISDN) Integrated Services Digital Network- is a circuit-switched telephone network system designed to allow digital transmission of voice and data over ordinary telephone copper wires, resulting in better voice quality than an analog phone. It offers circuit-switched connections (for either voice or data) and packet-switched connections for data in increments of 64 kb. The two standard levels of ISDN are the (BRI) Basic Rate Interface and the (PRI) Primary Rate Interface .
(IVR) Interactive Voice Response- An integrated software information system that speaks to callers and uses menus and voice responses. By using touch-tone keypad entries to interact with the software you'll get voice responses with real time data.
Jitter- is the variation in the time between packets arriving at the endpoint, which is caused by network congestion, timing drift, or route changes. As data loads increase and decrease, routers on the Internet can create slightly different times that individual packets take to travel from one point to another point. This variation in time is known as jitter.
Key System- A business phone system that allows users to access more than one telephone line and more than one line from more than one telephone, and place a line on hold in order to answer or initiate calls on other lines. Key systems usually include an intercom capability that allows users on different telephones in the system to communicate with one another. The number of station and phone lines served by a key telephone system is limited by the size of the key service unit. On a typical key telephone system, each incoming phone line appears on a separate button on every key telephone set, which is typically a small incandescent lamp or light-emitting diode and lets the user tell if the line is in use. The same lamp flashes at a certain rate when an incoming call is ringing on the line, and flashes at a different rate when a call is on hold.
Latency- is how much time it takes for a packet to reach its destination. Contributors to network latency include: propagation; the time it takes for a packet to travel between one place and another at the speed of light, transmission; the speed of travel over the medium itself (fiber, wireless, copper), routing; each gateway node takes time to examine and possibly change the header in a packet, storage delays; Within networks at each end of the journey, a packet may be subject to storage and hard disk access delays at intermediate devices. and packet size; the size of the data packet introduces delay in a round trip since a larger packet will take longer to receive and return than a short one. Higher delay times can be an issue, especially for VoIP, where voice delay can be recognized with latency higher than 150 milliseconds.
(LEC) Local Exchange Carrier- The local phone company, sometimes referred to as the Telco, that is responsible for delivering calls within a local area.
(LERG) Local Exchange Routing Guide- Is a database of the first 6 digits of a telephone number, updated on a regular basis, that provides information for routing telephone calls over the Public Switching Telephone Network, as well as, enables identification of what local company the number belongs to.
(LNP) Local Number Portability- is the ability of a US telephone customer to retain their phone number if they switch to another local telephone provider.
(MPLS) Multi-Protocol Label Switching- An IETF initiative that integrates layer 2 information about network links (latency, bandwidth, utilization) into layer 3 or IP within a autonomous network, which greatly improves IP-packet exchange. These advancements give network operators the ability and flexibility to re-route traffic around failure points, congestion and bottlenecks for a more robust stable network for their network users.
(NAT) Network Address Translation- An Internet standard allowing a local network to use one public IP address to connect to the Internet and a set of local IP addresses to identify each PC or device in the local network. The router translates or uses a conversion process to send data from the WAN to the correct device on the LAN that is using a private IP address.
On-premise- Used to define or represent phone equipment, such as the PBX phone system, which resides inside the user's premise or premises. This can include co-location facilities, a second location or the "phone closet" inside the business. Typically matched with PBX, an on-premise PBX resides within the user's facilities or business, unlike a hosted PBX or cloud based PBX which resides with a provider.
Packet Loss- During a transmission some of the data that is sent in packets are lost due to latency, congestion at a router or other network problems. Packet Loss is specifically problematic to VoIP and even a small amount can result in significant voice degradation. Since most VoIP transmissions use UDP protocols the delivery is not guaranteed and in the event that packets are lost they will not be requested to be retransmitted.
PBX (Private Branch Exchange), is a business telephone system using a network made of hardware and software components. The technology manages the routing of calls inside the organization and outside the organization and offers advanced features for handling calls. A PBX can use VoIP and landline connected calls. An IP-PBX is a PBX system with IP (Internet Protocol) connectivity and may provide additional audio, video, instant messaging communication and integrate with other advanced applications such as SalesForce or other CRM tools.
(POE) Power over Ethernet- is a technology for wired Ethernet LANs by which spare wires in the Ethernet cables are used to provide a voltage to power devices, such as IP Phones. This allows the connected devices, (IP Phones) to forego the need for power adapters.
(POTS) Plain Old Telephone Service- The traditional telephone service which uses analogue voice signaling from the phone to the local exchange.
(PRI) Primary Rate Interface- The Primary Rate Interface (PRI) serves as a telecommunications interface catering to the communication needs of enterprises and offices. Comprising 23 B channels and one data (D) channel for control functions, a PRI line delivers a total T1 bandwidth of 1.544 Mbits. Each B-channel is capable of transmitting data, voice, and video, while the D-channel handles control and signaling information. With a track record of three decades, PRI, using circuit-switched technology, ensures high reliability and call quality. However, recent trends show carriers are gradually phasing out copper PSTN connections, compelling users to transition to SIP.
(QOS) Quality of Service- The quality of transmission rates, error rates, and other characteristics of a particular type of protocol, mainly audio or video, over a network. Latency, packet loss, network jitter, echo and other factors may contribute to the quality of a call and the idea that these conditions can be measured, improved, and guaranteed in advance is deemed QOS. There are Quality of Service solutions offered by some providers that reserve bandwidth across their network, give priority to specific applications or protocols and by using the Internet's Resource Reservation Protocol (RSVP), packets passing through a gateway host can be expedited based on policy and reservation criteria arranged in advance.
Router- A router is a device connected to at least two networks that determines the next network point to forward a packet to. The decision of which way to send each information packet is based on it's current understanding of the networks that it is connected to.
(RTP) Real Time Transport Protocol- An Internet protocol that functions for end-to-end network connections to transmit data packets for applications that use audio or video. RTP combines its data transport with a control protocol (RTCP), which makes it possible to monitor data delivery for large multicast networks. Monitoring allows the receiver to detect if there is any packet loss and to compensate for some delay jitter. Both protocols work independently of the underlying transport layer and network layer protocols. The RTP header contains information that tells the receiver how to reconstruct the data and describes how the codec bit streams are packetized. Typically, both Session Initiation Protocol (SIP) and H.323, use RTP, which runs on top of the User Datagram Protocol (UDP) Asterisk typically uses UDP ports 10000 to 20000 for its RTP traffic.
SD-WAN (Software Defined WAN)- Software-defined wide-area network uses software based management that tries to eliminate the complexity of packet routing by an understanding of applications, their performance and the policies that are placed. Using software-defined networking (SDN) technology to prioritize bandwidth allocation and dynamic path selection for critical applications, like voice and video, according to policies and rules has multiple benefits over expensive MPLS. As more companies turn to cloud services for numerous applications the need for SD-WAN is increasing, especially for organizations with hybrid networks or multiple locations.
(SIP) Session Initiation Protocol- is a signaling protocol based upon a set of open standards managed by the IETF for Internet conferencing, telephony, and instant messaging. It is a request-response protocol, dealing with requests from clients and responses from servers initiating an interactive user session. Because of the open standards approach SIP devices are generally interoperable between different vendors. In a SIP based VoIP system most of the intelligence is in the IP Phones rather than centrally in the PABX.
Switch- A switch is a device that keeps a record of the MAC addresses of all devices connected to it and then channels incoming data from any of multiple input ports to the specific output port that will take the data toward its intended destination.
Trunk- Telecommunication refers to trunking as a technology for providing network access to multiple clients simultaneously by sharing a set of circuits, carriers, channels, or frequencies, instead of providing individual circuits or channels for each client. SIP Trunk- With regard to SIP trunks; the term "trunk" goes back to the early telephone days. When used today with SIP it doesn’t apply to a physical collection of wires; rather, the trunk represents a combination of “virtual” lines.
Unified Communications (UC)- is a term that describes the integration of enterprise communication services where each service is available from a unified user interface. Services typically include telephone calls, voicemail, Email, fax, video conferencing, data sharing, and integration with applications that handle a companies' CRM packages. UC can incorporate any and all communications and is not defined by any particular set or piece, rather it is a concept where all of a business communications are in one unified system.
UCaaS - A category of an "as a service" in which UC is Unified Communications delivered by a cloud PBX service. Using data centers to host the servers a provider delivers unified communications as a service over IP. With large robust Internet pipes available at lower costs than in the past enterprise sized companies are now paying for communication services in a service provider model. Some of the services that are part of UCaaS are voice and telephony, conferencing solutions, including video, messaging, and different communication apps.
Virtual Number- A VoIP phone number that is for a different location than where the user or company resides. For instance it could be a local telephone number from another state or country. Calls being made to the number from that local area are treated as a local call. The call is then routed to a user's primary VoIP number.
Virtual Phone System- Presently, "virtual phone services" specifically use forwarded virtual numbers (only) that then terminate to an existing phone service. These affordable services receive incoming calls placed to the virtual number directly through their cloud PBX allowing customers to apply big business phone system features. Some of the more prominent are virtual automated attendants that then route calls to extensions depending on the time and day. These virtual phone systems are favored with small start-ups and entrepreneurs as they are more affordable than traditional business VoIP services. Users typically continue to retain their cell phones and cell phone service which maintains their outgoing calls, unlike traditional VoIP service providers which connect directly with the PSTN.
(VPN) Virtual Private Network- Using tunneling, IP traffic is typically securely encrypted between two endpoints so that the transmission travels securely over a public TCP/IP network. Instead of using a dedicated, real-world connection such as leased line a VPN uses "virtual" connections routed through the Internet from the one end point, such as a business's private network, to the remote endpoint site, typically an employee.
WebRTC- Real Time Communications over the Web- an open source project based on HTML5 and JavaScript with the objective of setting parameters for using the web to advance real time communication, such as voice, peer to peer, video and data transfer all directly from the browser without the need to download additional applications.