CoreDial Definitions

Call Blocking

"Black list" phone numbers to block them from calling your Hosted PBX.

Customer Service Record (CSR)

A CSR is a record of the customer's account information as stored in their Telecommunication Service Provider's (or carrier's) database. CSRs are held by carriers and provided to customers upon request. A CSR includes the service location, billing address, billing telephone number, account number, and authorized contact information for the account, which may be different from the information printed on the customer's invoice. CSRs are often used when porting numbers and are also known as Internal Account Records, Service Records, and Billing Services Records. A customer's CSR is the most accurate source of porting information and helps ensure a fast, error free process. CSRs are not required for porting toll free numbers. Instead, the customer's most recent invoice is used.

Digital Signal Processor (DSP)

A DSP processes a digital signal converted from an analog signal. VoIP DSPs are application specific and conver analog voice to digital voice over internet protocol (IP).

Denial of Service (DoS)

A DoS attack (or Distributed Denial of Service DDoS) is an attempt to make a network unavailable to its intended users.

Hybrid PBX

A Hybrid PBX is a TDM PBX that is also SIP compatible. The process for enabling SIP varies by device and can range from a simple reconfiguration to complex reprogramming or licensing fees. If the process is too costly or complex, a VoIP gateway can be used to connect a hybrid PBX to a SIP trunk.

Hosted VoIP

A common term for Hosted PBX.

Concurrent Call Utilization

A concurrent call is defined as a call to/from somewhere on the CoreDial network to/from somewhere outside of the CoreDial network such as the PSTN or cellular network. While all calls made from a SIP device on CoreDial service will consume bandwidth on the local network (LAN), not all calls will consume concurrent call paths. It is important to make this distinction, as it is integral in the planning for network capacity and for contracted line count as both represent a cost to the end-user. The diagram attached illustrates the difference between bandwidth utilization and line (concurrent call) utilization in the CoreDial environment. Here you can see the difference between the bandwidth consumed for an extension-to-extension call at the same location when compared to the bandwidth consumed for a call from a single extension to/from the PSTN. You can also see how an internal or extension-to-extension call does not consume any concurrent call paths as all traffic stays within the CoreDial network while a call to or from the PSTN will consume a concurrent call path.

Do Not Disturb

A device or softphone feature that simulates a phone being off-the-hook and sends incoming calls directly into voicemail. Other routing options are also available.

Firewall

A firewall is a key security feature that sits between two networks, such as a company's internal network and the Internet. Firewalls prevents unauthorized people from accessing the internal network.

Extension

A standard extension is an individual user account on the cloud associated with a physical endpoint by a two to six digit number. A standard extension is associated with an endpoint, and the endpoint is associated with a device. The extension provides a pathway to the endpoint and its device. This is different from prem based PBX standard extensions, which are associated directly with devices. Standard extensions on Hosted PBXs are also referred to as SIP extensions. You can recive incoming calls and make outbound calls on a standard extension. Use features such as Disable Outbound Dialing for specific extensions. A cloud extension (i.e. a voice mailbox) is virtual meaning it is not associated with a phyiscal endpoint. You cannot make an outbound call from a cloud extension since it is virtual. A cloud extension forwards (or routes) incoming calls to other extensions. It cannot be associated with an endpoint. cloud extensions are great for external workers at a company whereby calls to their corporate cloud extension can be forwarded to their outside number, i.e. a personal cellphone using a feature like Find Me, Follow Me. Rules for Endpoints and Extensions: -Every device (used for making calls) must have an endpoint configured in the Hosted PBX. -Endpoints can be associated with either physical devices or softphones. -Every endpoint must have at least one extension assigned. -A single endpoint can have multiple extensions. -Standard extensions usually have endpoints. -Cloud extensions never have endpoints.

API (Application Programming Interface)

APIs provide a way for two different programs (or applications to interact with each other. CoreDial's open (public) API exposes active call control and dialing data. SwitchConnex plugins for Salesforce.com, MS Windows Outlook, ClickConnex browser support, and URL Agent allow call information to be passed between SwitchConnex and third party applications. The Dashboard app provides a SwitchConnex interface3 for real-time management of call center features and agents. API developer documentation is available from your CoreDial Partner Success Associate (PSA).

Analog Telephone Adapter (ATA)

ATA refers to a category of devices that connect analog phones to VoIP networks. ATAs generally include at least one RJ-11 (or FXS) phone port and one ethernet port. This allows analog phones to connect directly to a LAN and access an IP PBX or an internet-based VoIP service such as CoreDial's Hosted PBX. There are many varieties of ATAs, and models with multiple phone ports are often called Analog Gateways. Some modems include built-ins ATAs that allow a few remote phones to use an internet VoIP service via direct cable or DSL connection (without a LAN).

Conference Call (3-Way)

After making or receiving a call, a user may "conference in" any third party for a 3-Way call.

Direct Inward System Access (DISA)

Allows remote users to dial into their Hosted PBX from an outside line and make outbound calls that will display Caller ID information from a DID within their office.

Call Routing Time Frames

Allows routing decisions based on time and date. Multiple schedules can be configured for departments with different hours of operation (e.g. business hours, afters hours and holiday hours.

Directed Call Pickup

Allows users to dial *8 plus an extension number to answer call ringing at that extension. Note: This feature does not work with a Group or queue call, only direct extensions.

Listen Live

Allows you to listen in on a selected extension, but not to speak.

Commercials On Hold (By Phone Number)

Allows you to upload multiple On Hold commercials to your Hosted PBBX, and playback can be based on location or queue.

ACD (Automated Call Distributor)

Also called Automatic Call Distribution system, this refers to the software that runs a call center. ACDs are primarily responsible for sending (or routing) incoming calls to the correct destinations and typically include auto attendant functions, features for managing call center agents and queues, tools to collect and report statistics, and customizable routing options.

Agents

Also known as Call Center Agents, this refers to individuals who answer incoming calls from one or more Call Center queues. These users are set up with Agent login credentials in SwitchConnex, and their activities are tracked and recorded by the system. Agent activity can be monitored through some of the ACD Reports and in real-time via the Dashboard App.

Analog Gateway

Also known as PSTN Gateways, these devices generally connect LANs to the PSTN, so that IP (LAN) phones can access the PSTN. They may also connect analog phones directly to IP LANs, so that analog phones can communicate over a VoIP network, (small gateways are often called Analog Telephone Adapters). Analog Gateways are not used to connect legacy PBXs to CoreDial SIP Trunks.

A Record

Also known as an Address Record, this is a DNS (domain name system) parameter that associates an IP address with a domain (or website) name. This allows DNS to locate the correct physical device on the internet when you enter a website name. NOTE: When your SwitchConnex portal was built, an A Record was created to associate the IP address of a CoreDial server with your Portal Domain Name.

Enterprise Session Border Controller (ESBC)

An ESBC is a device that connects communications infrastructure to the public internet, SIP Trunking service providers and/or private networks. Its role is to terminate and reassemble received communications so that it can manage traffic while ensuring that the entire unified communications platform is secure.

IP Address

An IP Address is used to uniquely identify devices on a network and are classified as either 'public' or 'private' IP addresses. The customer's ISP will typically distribute one or more public IP address to the customer and the public IP will be assigned to the WAN interface of the customer's router or modem/router.

ISP (Internet Service Provider)

An ISP is the company you purchase your internet access from. Most ISPs provide additional services such web site hosting, etc. As the internet has grown, the industry has steadily consolidated, and is now dominated by large regional or national ISPs such as Comcast, Verizon, etc.

Letter of Authorization (LOA)

An LOA is a document authorizing a telecommunications provider to act on the customer's behalf. If porting a toll free number, a RespOrg LOA is used instead of a regular LOA.

Endpoint

An endpoint is an IP telephone, softphone, or analog telephone adapter device. Every device (used for making calls) must have an endpoint configured in the Hosted PBX. Every endpoint must have at least one extension assigned however a single endpoint can have multiple extensions.

Availability

Availability refers to the probability of a hardware failure. Availability is determined by dividing the "mean time between failure" by the "mean time to repair" - in other words, how often things break down divided by how long it takes to fix them. "Five nines," i.e. 99.999 percent, is the benchmark common to our industry.

Bandwidth

Bandwidth is the maximum amount of data that can be sent, in one direction, between your device and the internet. It is commonly referred to as internet speed, connection speed, or line speed; and is represented as the amount of data transferred per second. For example, a 500 Mbps connection can send 500 megabits of data in one second. Bandwidth can be misleading for two reasons: 1. It is not the same as througput. Bandwidth rates are based on optimal conditions; therefore, any network problems will cause the actual throughput (data successfully transmitted per second to be lower than the bandwidth. 2. Internet connections are typically asymmetric, meaning they are configured with more download bandwidth (from the internet to you) than upload bandwidth. For example, a 500 Mbps connection can download data at a rate of 500 Mbps, but may only be able to upload data at a rate of 100 Mbps. It's important to consider the following factors are when determining BoIP bandwidth: -VoIP requires the same amount of bandwidth in both directions. -CoreDial uses G711 encoding, which transmits VoIP packets at a rate of 80 Kbps (0.64) Mbps). -VoIP bandwidth requirements should be based on the peak number of concurrent calls. -When adding VoIP to an existing internet connection, the bandwidth must be sufficient for concurrent, peak voice and data transmissions.

BroadSoft

BroadSoft is a global software company that created and continues to maintain the BroadWorks VoIP platform. Cisco purchased BroadSoft in October 2017; however, that hasn't impacted CoreDial Partners' ability to sell BroadWorks Hosted PBx, SIP Trunk, and UC solutions.

BroadWorks

BroadWorks is the most widely used VoIP application platform. It's integration with SwitchConnex (Sept. 2017) allows CoreDial Partners to sell Hosted PBX, SIP Trunk, and UC solutions on the BroadWorks platform (in addition to the Asterisk platform). The BroadWorks platform is typically implemented in medium to large sized companies who have multi-site, large call centers, or require complex telephony solutions.

Broadband

Broadband refers to a general category of long distance, high-speed methods for sending information between devices. It is available on a variety of physical media such as fiber, FiOS, DSL, Wi-Fi (coaxial cable, cellular, satellite, etc; and each technology has its own pros and cons. Regardless of the medium, all broadband implementations share the following traits: -The connection is always available or "always on" -The medium (e.g., wire) is divided into mulitple pathways called channels or bands. -This allows a single medium to simultaneously transmit different types of information (e.g., video and data) to different destinations. -Information is downloaded (received by a device) at a minimum speed of 25 Mbps and uploaded (sent) at a minimum speed of 3 Mbps. Broadband is now the default term commonly used to describe any type of internet connection that isn't dialup.

Bursting

Bursting is a general term that describes a sudden, and usually temporary, spike in the amount of data traveling over a specific connection. In VoIP, bursting occurs when a customer's toatal call activity exceeds the combined capacity of all their connections (or call paths). For example, a customer with 3 prepaid call paths will reach maximum capacity when 3 users are simultaneously on calls (to/from external numbers.) Bursting would occur if a 4th call arrived during that time. CoreDial's Hosted PBX and SIP Trunking solutions can dynamically add more call paths as needed when bursting occurs. This ensures callers will not receive busy signals during peak activity times.

CLEC (Competitive Local Exchange Carrier)

CLECs are companies that have their own access to the PSTN and are able to sell phone services directly to consumers. They may own and operate regional networks (local loops), switches, and inter-switch connections; or they may lease some local or backbone PSTN components form ILECs. CLEC customers are generally unaware of the underlying ILEC relationship. CLECs are regulated by the FCC. It's important to note that CoreDial and CoreDial Partners (Service Providers) are not CLECs.

CPNI

CPNI, or Customer Proprietary Network Information, is the data collected about an individual users calls, such as time, date, duration, and destination number of each call.

Customer Relationship Management (CRM)

CRM software manages all aspects of an organization's interactions customers and prospects. A unified communications solution can make a CRM system more accessible across an organization because it creates a heightened level of accessibility.

Computer Telephony Integration (CTI)

CTI technology enables integrated interaction on a telephone and a computer, such as click-to-call and screen pops.

Call Queue

Call Queues are used to route calls on a first-in first-out basis to the appropriate extension or group. These extensions can be agents logged into the system. Call queues are commonly used with an ACD, where callers hear an announcement such as "Thank you for calling, all available agents are busy, please hold for the next available agent, or press "1" to leave a message". When the call is ready to be routed, the ACD handles the routing rules.

CNAM (CID Name)

Caller ID Name (CNAM) is an enhancement to CID that allows the caller's name to be displayed along with the phone number. CNAM databases are maintained by carriers (who own the numbers) or by third party services. When you receive a call, your phone service provider checks the incoming number against a CNAM database, retrieves the CNAM, and displays it with the CID. THis process is referred to as "dipping" and there is often a fee associated with it. Although the PSTN restricts a CNAM to 15 characters, VoIP does not have this limitation. However, longer CNAMs will only be displayed for calls that originate and terminate on a VoIP network. When a VoIP call terminates on the PSTN, the CNAM will be truncated after the first 15 characters (including spaces). For this reason, most CNAMs are still 15 characters or less. NOTE: CNAM has no correlation to CNAME, which refers to a DNS resource record.

Click-to-Dial

Click-to-Dial is the ability to initiate a phone call from the contact list on your computer with the click of a mouse. CoreDial's browser plugins enable Click-to-Dial functionality for the following browsers: -Microsoft Internet Explorer -Google Chrome -Safari -Mozilla Firefox

Cloud Communications

Cloud Communications are voice and data communications over the Internet. All applications, switching and storage are hosted by third-party entities outside the organization and accessed over the Internet. With cloud communications, there is no major capital expenditure for an in-house PBX system and ongoing costs are more predictable than with a traditional premise-based solution.

Cloud Hosting

Cloud Hosting services provide hosting for websites on virtual servers which pull their computing resource from extensive underlying networks of physical web servers. It follows the model of computing in that it is available as a service rather than a product. This allows a client to tap into their service as much as they need, depending on the demands of their website, and they will only pay for what they use. Examples of cloud hosting can fall under both the Infrastructure as a Service (IaaS) and Platform as a Service (PaaS).

Codec

Codec is an abbreviation of the words "compression" and "decompression". The function of a VoIP codec is to convert an analog voice signal into a compressed digital signal that can be sent over the internet; and to reverse the process on the other end. The two standard VoIP codecs are G.711 uLaw provides the highest quality: 200 bytes/packet, full duplex, 80kb/sec/call, and requires 800kb/sec bandwidth for every 10 concurrent calls. CoreDial recommends you use this codec whenever possible. G.729a provides good quality, but not as high as G.711: 60 bytes/packet, full duplex, 30kb/sec/call, and requires 300kb/sec for every 10 concurrent calls. CoreDial recommends you use this codec only as temporary solution when needed.

Dynamic Host Configuration Protocol (DHCP)

DHCP is a network service that automatically provides an internet protocol (IP) host with its network addrss and other related configuration information such as the subnet mask and default gateway as a person shifts from one network to another. Benefits: The DHCP server minimizes configuration errors caused by manual IP address configuration, such as typographical errors or address conflicts caused by the assignment of an IP address to more than one computer at the same time. It also reduces network administration by i.e. efficient handling of IP address changes for customers that must be updated frequently, such as those for portable computers that move to different locations on a wireless network. DHCP allows the customer to move their device from one place to another without needing to make any configuration changes.

Direct Inward Dial (DID)

DID is used for call routing. Through DID, external callers are able to contact a user directly at their unique phone number. Set up a telephone number to dial directly to a device or extension.

Advanced Call Routing

Enables customers with advanced routing capabilities including call forwarding, and call status (away, busy, unreachable, etc).

Fixed Mobile Convergence (FMC)

FMC is the convergence of the fixed and mobile networks. FMC solutions is the method used to integrate cellular services with private communications networks, by connecting the mobile phone to the fixed line infrastructure, irrespective of whether the user is wired or wireless and regardless of their location.

Automated Billing System

Features on the CoreDial platform include monthly billing, automated credit card charges, Dunning notices, and A/R.

Find Me, Follow Me

Find Me, Follow Me is generally used as a call-forwarding feature. It improves worker productivity and customer service by ensuring that every call reaches the right person, regardless of where he or she is working. The Find Me feature attempts to locate you by dialing up to 5 locations until you either accept or reject the call. The caller is placed on hold until you are found. They are also prompted to announce themselves and are given the option to try the next location or to leave a message. On answering, you will have the option to accept the call, or reject the call. You also have the option to reject the call and leave a short message for the caller. The caller will not know that you have rejected their call if you chose that option.

Incoming Privacy Screening

Force callers with "No Caller ID" or"Blocked Caller ID" to enter a number that will be presented as their Caller ID.

Call Forwarding

Forwards calls via the portal, or via the portal, or via your device or softphone. Calls may be forwarded to any extension or phone number. Note: Device or softphone forwarding functionality may vary by manufacturer.

Frame Relay

Frame relay is a cost-efficient method of data transmission for intermittent traffic between LANs, and between end-points in WANs. They're less expensive than private leased lines because the carrier shares the frame relay bandwidth among many customers. This can have a negative impact on quality, and therefore requires detailed attention to engineering the solution.

Graphical User Interface (GUI)

GUIs allow users to interact with computers through graphical icons and visual indicators as opposed to text-based (command line) interfaces.

Extension Dialing (1-6 Digit)

Hosted PBX extensions can consist of 1-6 digits.

ILEC (Incumbent Local Exchange Carrier)

ILECs are what the original telephone monopoly companies became after the Telecommunications Deregulation Act of 1996. Prior to the Act, these companies were the sole providers of phone service, and they owned and controlled the PSTN infrastructure throughout the US. After the Act, the original companies were divided into regional companies (ILECs) and required to lease PSTN components (E.g., backbone lines) to newly formed, competitor phone companies called CLECs.

Instant Messaging (IM)

IM is a real-time communication over the Internet using text-based messages. Popular consumer-facing examples include G-chat, AIM, and iMessage. IM is usually a central feature of unified communications.

IP Phones

IP phones plug directly into the network and perform analog-to-digital and/or digital-to-analog conversions.

ITSP (Internet Telephony Service Provider)

ITSP is the standard industry designation for businesses that sell VoIP services. CoreDial Partners (Service Providers) are ITSPs. NOTE: The FCC's definition of ITSP is Interstate Telecommunications Service Providers, which refers to companies that sell multi-state phone services via the PSTN. The FCC categorizes businesses that sell VoIP services (e.g., CoreDial Partners) as Interconnected VoIP Providers.

Circuit Switched Network

In telephony, a circuit refers to single, dedicated path (usually a wire) between two points through which audio signals can travel in both directions. A switch is a device that receives a signal from one location and forwards (or switches it to another location (either the final destination or another switch en route to the destination). A circuit switched network combines both technologies. It's a group of geographically distributed switches connected together via backbone circuits, and end point devices such as phones are connected to the switches via local circuits (phone wires). This allows a connection (or circuit) to be established between any two end points on the network regardless of their actual locations. The PSTN is a world-wide circuit switched network.

Analog

In telephony, analog refers to a process of converting sound into electronic signals and sending them between two devices. Analog signals travel in a continual stream (or wave) that fluctuates as the sound level changes. The PSTN (public switched telephone network) was originally built with analog technology; however, it's now mostly digital. The terms analog, landline, and POTS (plain old telephone service) typically refer to a single connection to the PSTN such as a traditional household phone line.

IP (Internet Protocol)

Internet Protocol is one of the underlying components that allows devices to communicate on the internet. All internet communication (including VoIP) uses a group of protocols that work together, and each one has a specific function. The role of IP is to read the destination addresses on packets (of data) and deliver them to the correct location. IP can only read IP Addresses, which are formatted as four numbers (between 0-255) separated by periods such as 1.150.20.235. Each device on the internet requires a unique IP address.

Jitter

Jitter is a measure of the variation in packet arrival times from phone to server. Jitter is generally caused by network congestion, which is created by delay, variation or timing issues when the packets arrive. The result is poor and/or unacceptable voice quality. Jitter often manifests itself as a "delayed response" from one caller to the other caller or users sounding like they are talking over each other. Sometimes jitter could sound like "squawking" noises, if packets are played out of order. In order to maintain quality voice communications, try to look for Jitter measurements of less than 5ms. The best solution for jitter is to increase the capacity of the congested device or link.

Local Number Port (LNP)

LNP is the process of transferring standard telephone numbers (or DIDs) from one carrier to another. This allows customers to keep their existing phone number(s) when they switch telecommunications service providers. It is commonly known as Porting or LNP.

Latency

Latency is the time it takes for a caller's voice to be transported - packetized, sent over the network, depacketized and replayed - to the other person. Too much latency is bad, making for a disjointed conversation flow. Ideally, latency should not exceed 150 milliseconds (one-way). Geographical distance (i.e. wireless) or a lower-speed network connection can cause latency issues to get worse.

Managed IP Telephony Services

Managed IP Telephony Services is another phrase for "Hosted" services. Typically, the end-customer business customer owns the IP PBX and related equipment, while the carrier or VAR provides management and maintenance for the phone system.

Mean Opinion Score (MOS)

Mean Opinion Score is a test that has been used in telephony networks for decades to obtain a user's view of quality on the network. In measuring VoIP, it is a calcuation based on performance over the IP network in which it is carried. The range is 1 to 5, where 1 is lowest perceived audio quality and 5 is the highest. A typical range for VoIP would be 3.5 to 4.2. Use an Edge Device that collects MOS data. Common Edge Device brands are Adtran, Cisco, Edgemarc, Peplink, and Sonic Wall. Note that scores are an aggregate of many factors and must be read in the correct context. For example, a two hour call with poor quality during the last few mintues will have a high MOS value. These devices also track additional performance factors. You can measure the quality of a given network connection at a specific slice in time by using the CoreDial VoIP readiness test - http://test.sipregistration.com The test must be run from the customer's location.

IP Telephony

More commonly referred to as Voice over IP (VoIP), IP telephony uses the IP network to carry voice communications, replacing the public switched telephone network.

Conference Bridge

Multiple on-site and outside callers can simultaneously participate in password-protected conference calls. Callers can be assigned "talk/listen" or "listen only" status.

Converged Network

Network convergence (also called media convergence) refers to using a single network for multiple types of media such as vocie and data. Converged networks are becoming more commonplace as technology improves and costs decrease; therefore, this module focuses on optimizing VoIP in converged networks. Converged networks can share the same LAN infrastructure, WAN infrastructure, or both. A common reason for switching from a dedicated to converged network is the lower cost of a single communications line. With proper configuration and management, converged networks are highly reliable. SOHO (small office home office) networks with up to five VoIP phones are essentially very small converged networks. Because of the relatively light traffic loads, VoIP performance is usually acceptable without changing the LAN configuration. SOHO networks are also called flat networks. Dedicated and SOHO networks are generally simpler to configure and manage, and use the same VoIP optimization techniques as converged networks.

Control

On CoreDial's platform, you are able to make changes as needed, including scalability, API integration, and an intuitive UI (user interface).

Call Hold

Place calls on hold and play music or a commercial while a caller on hold.

Call Detail Records

Real-time call logging is available within the portal. information displayed includes call origin, destination, duration, date and time, and call type (International, On-Net, etc.).

Call Control

Refers to the software within a telephone switch that supplies its central function. Call control decodes addressing information and routes phone calls from one end point to another. It also creates the features that can be used to adapt standard switch operation to the needs of users. Examples include Call Waiting and Call Forward on Busy.

Call Recording

Selectively record calls for training or documentation purposes. Note: Requires a dedicated server in the cloud.

Live Person Answering

Set up a telephone number to ring a specific extension or a Ring Group - sequentially or simultaneously. This option enables your company to use a live person to answer the caller instead of an Auto Attendant.

ACD Reports

Seven ACD Reports are available in SwitchConnex. Collectively, they provide a thorough picture of Call Center (or ACD) operations. The Realtime Console Report displays live statistic for queues and agents, and is updated in 30 second intervals. The other six reports provide a variety of statistics about agents, queues, and call volumes. They can be run on demand and filtered by date, time, queue, agent, or phone number; and the results can be exported.

Barge

Short for "Barge In", this call center feature allows an authorized user to listen in (hear both parties) during an active call. The call center agent is aware the call is being monitored, but the outside party is not. Barge is typically used to monitor or train call center agents. CoreDial's platform includes the following Barge options: -Listen Live: the authorized user's phone is muted while they are listening to a call. -Barge- the authorized user can speak to the call center agent during a call. The outside party cannot hear the Barge conversation. The option is commonly referred to as "whisper".

Auto Attendant (AA)

Short for Automated Attendant, AAs are also known as "Virtual" Attendants, Receptionists, or Operators. Their primary function is to automate the process of answering and routing incoming calls. AAs answer calls with a prerecorded message that offers a menu of options for completing the call. Callers select an option by pressing a button on their phone, and the AA sends (or routes) the call to the correct destination. The SwitchConnex AA can route calls to 19 different destinations such as voicemail, external numbers (including mobile), ring groups, extensions, company directories, other AAs, general mailbox, etc.

Carrier

Short for Common Carrier, this refers to a company that sells phone services to the general public, and also owns the transport lines and equipment. Telephone service is a public utility, so carriers are regulated by the FCC and required to charge all customers the same price (for the same service). Following the Telecommunications Deregulation Act of 1996, carriers were categorized as either ILECs or CLECs. NOTE: CoreDial is not a carrier.

IP PBX

Short for Internet Protocol Premise Based Private Branch Exchange, this is a prem based PBX that connects to a LAN. It requires IP phones, which also connect to the LAN versus connecting directly to the PBX (as with a standard prem based PBX). All communications between the in-house phone and the IP PBX use IP. On the outbound side, the IP PBX can use a standard telephony interface (e.g., PRI) or a SIP trunk to reach the PSTN. IP PBXs use more software controls than standard PBXs and also have more features. Although IP PBXs are often referred to as "digital" PBXs, this can be misleading because standard (non-IP) PBXs can be either digital or analog. Therefore, not all digital PBXs are IP PBXs can be either digital or analog. Therefore, not all digital PBXs are IP PBXs.

Accounting Reports

SwitchConnex Accounting Reports provide information on accounts receivable, invoices, payments, revenue, credits, sales tax, E911 addresses, contract maturity, and cancelled accounts. Details are available on a per customer basis, and summary information can be viewed for all customers.

Activity Reports

SwitchConnex provides three Activity Reports (Call Activity, Hosted Fax, and Virtual Path) that provide usage statistics for the past 90 days. Reports for Call Activity and Hosted Fax display inbound/outbound volume, and Virtual Path Usage displays virtual paths and concurrent calls. All three reports can be viewed on demand; additionally, the Call Activity report can be automatically generate and sent via SFTP. See ACD for additional call center statistic.

Asynchronous Transfer Mode (ATM)

The ATM standard was developed in the 1980s to transmit voice, video, and data over a single connection. It encapsulates network traffic into small, fixed-sized cells (or packets) and transmits them over a virtual, point-to-point, high-speed connection. This differs from internet protocol (IP), which uses variable sized packets and multi-point connections. As IP evolved and became more efficient, it is essentially replaced ATM and became the dominant standard for transmitting voice, video, and data over a network.

CALEA

The Communications Assistance for Law Enforcement Act (CALEA) is a federal law that requires VoIP telecommunications service providers to cooperate with court ordered electronic phone surveillance operations. CoreDial Partners must file a CALEA SSI Plan with the FCC. This is a document that provides contact information and identifies internal company procedures to be followed in the event an investigation is required.

Federal Communications Commission (FCC)

The FCC provides regulatory oversight of VoIP services at the federal level. The FCC also coordinates collection and dissemination of information to other federal agencies. All VoIP Service Providers must comply with the same set of federal regulations. Note: State laws regulating VoIP vary on a state-by-state basis. In some states, local entities (e.g., counties, municipalities, etc.) impose additional VoIP regulations.

Firm Order Completion (FOC) Date

The FOC Date is the date when the local carrier will commit to transferring the customers existing phone number to another carrier.

E911

The term "E911" refers to the legal requirement for ITSPs (Internet Telephony Service Providers) to ensure that dialing 911 (from any device) will connect to the appropriate emergency response center (PSAP) and automatically transmit the caller's address and phone number. E911 laws are imposed at the state and local levels and vary by region. For example, on state may require that addresses include an office or cubicle number; and another state may only require the building and floor number. It is the ITSP's responsibility to identify E911 laws and configure the SwitchConnex E911 Location feature correctly for each Customer Account. NOTE: There are no federally mandated E911 procedures such as registrations or fees; however, the FCC does require that ITSPs comply with E911 laws in each area where they are providing services.

Disaster Recovery

These are plans that are in place to provide a fluidity of business operations in the event of a service emergency or natural disaster, by backing-up data and managing that data for a rapid recovery. CoreDial aims to virtually eliminate business downtime should your local area network (LAN) or wide area network (WAN) communications network go down. Your communications solution is still running on our cloud environment, so your customers, vendors, and company calls can keep flowing.

FXS and FXO

These are the names give to ports that are used by analog phone lines (also known as POTS - Plain Old Telephone Service) or phones.

Hosted Fax

This feature allows you to send faxes from your computer (as a PDF attachment), which can then be received via email or routed to a physical fax device. This feature provides immediate access to faxes - anywhere, anytime, and from any device.

Hosted PBX

This is a "software-based" PBX service that is available through the internet. It's an alternative to a traditional "prem-based" PBX and provides the same functions (e.g., auto attendants, voice mail, extensions, etc.)/ The physical hardware and software for the Hosted PBX are owned and managed by a VoIP provider such as CoreDial. A Hosted PBX requires VoIP phones. These are connected to the internet, and all telephone services are delivered through the internet. The end user customer manages the Hosted PBX through a GUI that is similar to the prem-based PBX screens. They pay an ISP for internet access and an ITSP for Hosted PBX access and VoIP phone services. CoreDIal manages the call flow between the Hosted PBX and the PSTN. A Hosted PBX is commonly referred to as a Virtual PBX or Hosted VoIP.

BTN (Billing Telephone Number)

This is a 10-digit number that the local phone company (ILEC or CLEC) uses to bill a customer for services. When carriers issue phone numbers to customers, they select one of the numbers to be the account billing number. This is referred to as the BTN, and it's used to bill the customer for all services delivered to a specific location. For example, a customer might have 5 phone numbers (DIDs) for their business. One number is designated as the BTN, and services for all 5 numbers are billed under that BTN. Customers with multiple locations may have different BTNs for each address. Contrary to popular belief, phone numbers are actually owned by the carriers who issue them versus the customers or businesses who use them. Regardless of whether a customer is using VoIP or non-VoIP phone service, there is always a carrier and a BTN associated with their account. The BTN is required in order to move (or port) a customer's phone numbers onto CoreDial's platform. If the customer doesn't know their BTN, they can request it from their current phone service provider.

Integrated Access Device (IAD)

This is a customer premise device that provides access to WANs and the internet. It aggregates multiple channels of information such as voice and data across a shared access link to a service provider. Examples of an access link include a T1 line or a DSL connection. The IAD is installed by the service provider on the customer's premises and thus allows the service provider to control and manage the features and operation oft he access link.

Hop

This is a part of a signal's journey from the source to the destination. Data packets pass through bridges, routers and gateways along the way. Each time a packet is passed to the next device, a hop occurs.

Busy Call Forwaring

This phone system feature automatically forwards an incoming call to another destination if the called number is busy or has DND turned on. In SwitchConnex, this is one of several Inbound Dialing Rules that can be configured for an extension. Calls can be sent to any routable destination such as another phone number or extension, ring group, voice mailbox, etc.

Attended Transfer

This refers to a notification process that occurs before transferring a call. It is also known as Announced, Warm, or Supervised Transfer. When a call needs to be transferred, it is first put on hold. The intended recipient is then called and notified of the party on hold (who cannot hear the exchange). If the recipient agrees to accept the call, the transfer is completed.

BYOD (Bring Your Own Device)

This refers to the general practice of employees using personal devices to access work networks and applications instead of (or along with) employer provided devices. In telephony, many people define BYOD as using personal smartphones for business purposes. The SwitchConnex Hosted PBX platform supports BYOD in the following ways: -Smartphones: The ClickConnex Mobile App allows an employee to use their own iOS or Android device to make/take calls from their work telephone number. It also provides access to the following Hosted PBX features: call forwarding, DND, contacts, voicemail, and Virtual Attendants. -Softphones: SwitchConnex supports softphone applications running on computers or smartphones. This means the application itself (instead of a physical phone) can be configured as a Hosted PBX extension. This becomes a BYOD solution if the softphone is running on an employee's device. -Desk Phones: The system includes a generic BYOD template that can be used to provision phones without preconfigured SwitchConnex templates. This is a good option for customers who want to use their pre-existing SIP phones or who need to integrate a few specialized phones.

Caller ID (CID)

This refers to the process of displaying a caller's 10 digit number on the recipient's phone, while the phone is ringing. This allows the recipient to identify the caller before answering the phone. By default, carriers transmit CID with outbound calls; however, most phones allow users to modify the default behavior. CID is also referred to as Calling Line Number. CID Name (CNAM) is an enhancement to CID that displays the caller's name along with the phone number. See the CNAM entry for more information. When calls are placed through a PBX, the CID and CNAM for each phone can be customized. The terms CID, CNAM, and Calling Line ID are often used interchangeably.

Dedicated Network

Traditionally, different types of media such as voice and data had their own physical (combined LAN/WAN) infrastructures, known as dedicated networks. For example, voice networks consisted of handsets,PBXs, and a connection ot the PSTN. Data networks included computers, switches, routers, firewalls, and a connection to the internet. Dedicated networks are highly reliable and can provide guaranteed WAN access speeds. Dedicated LANs/WANs provide the best possible VoIP service. They are easier to install, manage, and troubleshoot but are also more expensive to operate.

Call Park

Unlike a call placed on hold, a parked call may be picked up from another extension.