Busy Lamp Field (BLF)
A set of illuminated buttons on a physical phone that provide visual indication for busy (usually red light) and available (usually green light) lines and extension on the PBX system. The buttons can also be pushed to dial an available line or extension. The TeleConsole has software settings to set up BLF. You can read about this and how it is difference from speed dial here.
A telephony service that allows a recipient of a phone call to see the caller’s details, on his phone screen or VoIP software, before picking up the call. Usually both the phone number and name of the callers are provides and with PBX systems it is also possible to add an extension that transfered the call.
Caller ID originated before the days of digital communication and with traditional land lines the caller ID is always identical to the actual phone number of the caller. However, with VoIP and PBX systems alternative numbers can be applied as the caller ID which serves to better identify specific departments, products, or individuals.
A caller usually has the option of blocking (hiding) his Caller ID information by dailing a specific code or with a relevant software option. With Telebroad's TeleConsole you can select or block a caller ID from the setting menu or directly from the dialing screen. Additionally there is a call screening settings feature where a caller is asked to record his name, even if his caller ID is blocked, before the call is passed on to an agent. See also Outbound Caller ID.
Call Flow and Call Segment (Leg)
In a non-PBX phone system the caller's experience is of a direct non-interactive connection with his intended called party (the call may be routed through different switches, but this is transparent to the caller.) In a PBX system, however, a caller has to navigate and interact with menus, directories, queues, and other destinations to reach his intended party. Each destination he navigates or get routed to by the system is called a Call Segment or a Call Leg. The progress of the calls between these destinations (the collection of segments) is referred to as the Call Flow. For example when a caller arrives at the IVR he is at the 1st segment (leg) of the call flow, when he makes a selection and get routed to a support department he is at the 2nd segment (leg), and when an agent answers the call it is the 3rd segment etc.
Call agents are usually not concerned with Call Flow. Call flow information is much more important to call center managers or phone administrators and can be accessed with the call logs function, including any relevant recording in either the ACD Panel or Analytics or by sending this API request.
A network connection method where a dedicated wiring path is established between a source and a destination before the transmission occurs. Circuit switching is traditionally used with analog telephone communication. A continuous wire connection is created in the telephone exchanges once a call is from one telephone is picked up by the recipient. While the connection is in session the only data that travels on it is between the two connected phones. The data is sent continuously and in sequence, unlike the divided data in Packet Switching. This gurantee call quality and reliability at the expense of lesser network capacity (since lines can't be shared).
A data conversion software or hardware that stands for coder-decoder. It converts data from one form to another and then back to the original form. In VoIP communication, speech (analog data) is converted to digital data before it get sent to the recipient where it is converted back to speech on his phone, computer, or mobile device. Codecs usually also perform compression of the data to reduce transmition times. There is a tradeoff between the amount of compression and data-loss, although the resulting reduction in call quality is not meant to be perceptible in most situations. VoIP codecs have a measurements of minimal network speed that will support their acceptable performance. With the mobile version of the TeleConsole you can select between three popular codecs by changing the call quality settings.
DSS stands for Direct Station Selection. It is a type of a soft key on a business desk phone that can be programed by the user to perform a certain function. It is commonly used for BLF/Speed Dial functions but can be assigned to do almost any function on the phone such as transfer, intercom, forward, park and retrieve, call return etc.
A telephone technology, also known as Touch-Tone, for dialling numbers or making selection from IVRs where each button on the keypad generates and is identified by a specific audio signal. With traditional telephones systems (POTS) the DTMF signals are transmitted on the exactly same copper wire that carries the conversation. This is called in-band transmition and it is also used in VoIP systems where the DTMF signals are packaged with the audio data. However, unless an uncompressed codec is used, data compression and network interference can result in loss of the signals. With VoIP the preferred DTMF delivery method, called out-of-band, is to send the signals separately from the actual audio stream.
FXS stands for Foreign Exchange Subscriber. It is a type of port on analog telephony network (or POTS service) that delivers the analog signal and electric current from a phone company (or the PSTN) to an FXO port (see next term) on analog telephone, fax machine, or other analog devices (the FXO port provides an on-hook/off-hook indication to start and end a call). Offices that have existing FXS ports can connect them to an IP network using a VoIP gateway, a device that can convert multiple analog lines to digital VoIP data.
FXO stands for Foreign Exchange Office. It is the port that receives the analog telephony equipment that receives the analog signal delivered from an analog FXS port (see pervious term). Because the signal from the FXS port is continuous, the FXO port is required in order to provides an on-hook/off-hook indication to start and end a call. Offices that have existing equipment with FXO ports only can connect them to an IP network using an ATA adapter.
The Internet Protocol is a communication addressing system used to identify devices on a network and across the Internet. IPv4 (Internet Protocol Version 4) is its fourth revision and most widely used. It is a 32 bit addressing system with a limit of about 4.3 billion devices. With growing number of computing devices (especially smart phones) this limit will be reached in the forseeable future.
To deal with this the IPv6 version was introduced. It is a 128 bit addressing system with more than 3 Duodecillion possible addresses (3 followed by 39 zeros). It also offers additional configuration, routing and privacy benefits. Presently both formats are in use alongside each other and implementation of IPv6 depends on both server and devices support. Your SIP or IP phone are assigned either an IPv4 or IPv6 address when they are provisioned.
Internet Service Provider is a company that provides your business with connectivity to the Internet. The ISP usually provides both the physical wiring infrastructure (cable, phone lines, fiber optics etc) and the networking access that enables Internet traffic over the wiring. In some rare cases the infrastructure and networking are provided by two different companies, but usually your available infrastructure will also dictate your choice of ISP.
Your ISP also gives you a router to facilitate the Internet connectivity. Not all routers may be compatible with VoIP or with Telebroad's services out of the box, but we do have some guidelines of how to adjust them for such functionality. Our support team will occasionally refer you to your ISP if they think an issue with your Telebroad account is due to Internet connectivity problem rather than a problem on our end.