Call Processing Feature/Function Glossary and Definitions

This appendix is an abridged glossary of voice calling features that are available on leading PBX systems. This glossary is intended to be representative of the most popularly requested customer features, but it does not include all of the currently available features. Each PBX system feature set is unique, and feature capabilities differ across manufacturers’ models. Although the typical PBX station user may commonly use only a small fraction of these features under normal working conditions, each feature has some productivity and/or cost savings value. PBX systems that lack more than 15 percent of these features cannot be said to be competitive in the marketplace.

Station User Features

Add-on Conference

This feature allows a station user to add a third party to an existing two-party conversation.

Automatic Callback

This feature is used when a dialed station is busy. When the feature is activated, the system automatically attempts to call the desired station until the line is free. The calling party is alerted that the called party is available. This saves wasted time dialing when encountering busy signals.

Automatic Intercom

This feature provides a talking path between two voice terminal users. A station user presses a programmed automatic intercom button and lifts the handset, or vice versa. The called user receives a unique intercom alerting signal, and the status lamp associated with the dial or automatic intercom button, if provided, flashes.

Bridged Call Appearance

This feature allows the same line appearance to be programmed to appear on more than one telephone. It is very useful in manager–assistant relationships and call answering position environments. A bridged call appearance can reduce the number of abandoned or lost calls and allows the coverage station to prescreen calls for the called party.

Call-Back Last Internal Caller

This feature allows a station user to automatically consult and call back the last internal caller to the station (unanswered call) by implementing the feature code.

Call Forwarding—All Calls/No Answer/Busy

This feature allows a station user to divert all incoming calls to another programmed station. All Call activation diverts all incoming calls to the station; no answer activation diverts calls after a programmed number of rings; busy activation diverts incoming calls when the station is busy. The features are useful when a station user is away from the desk area, wishes to receive calls at another station, or when there is a desire not to receive calls, but the user wants the call answered. The most common coverage station is a voice messaging port. Call forwarding features decrease abandoned or lost calls and improve call coverage service for the calling party.

Call Forwarding—Follow Me

This feature allows a station user to activate the call forwarding feature from a remote telephone by changing an existing forwarding destination. It provides station users with the capability of changing call forwarding destinations without returning to their desks, and can be used to “follow” station users around the system if they wish to receive calls at different stations.

Call Forwarding—Off Premises

This feature allows a station to forward all calls to an off-premises location outside the system. It allows station users to receive calls at a programmed outside telephone line when they are out of the office.

Call Forwarding—Ringing

This feature allows a station user of a multiple line voice terminal with display to forward an incoming call during the ringing period to another station. The station user can read the display for screening information, such as CLID or calling party name, before activating the feature through a programmed feature button or dial access code. The rerouted call destination is input by the station user after feature activation. The operation is transparent to the calling party.

Call Forwarding—Selective Multiple Line

This feature allows a station user of a multiple line voice terminal to selectively call forward any or all line appearance numbers.

Call Hold

This feature allows a station user to place an existing call in a hold state when there is another incoming call or the station user must leave the desktop area for more than a few seconds. Call hold provides station users with the flexibility of handling multiple concurrent calls without re-establishing the connection after finishing another call.

Call Park

This feature allows a station user to “park” a call at the received station, effectively placing the call in a hold state, retrieve the call at another station, and continue the conversation. A second party can also retrieve the parked call if notified by the first party. The feature provides station users with mobility and eliminates the need to return calls.

Call Pickup

The feature allows a station user to retrieve and answer a call directed to another station (direct), any station in the station user’s assigned call pickup group (group), or another call pickup group (designated group). The station user presses a programmed call pickup button or dials the desired feature access code to implement the feature.

Call Transfer

This feature allows a station user to divert an existing call to another station within the system. It eliminates the need for the original calling party to hang up and redial another telephone line to reach the desired or proper called party.

Call Waiting

This feature notifies a station user engaged in conversation that there is another incoming call to the station line number. The notification is usually a special tone or display signal on the telephone. Call waiting reduces lost calls, improves customer service, and reduces the number of calls forwarded to another station or messaging system.

Consecutive Speed Dialing

For speed dialing all station number digits are registered as the speed dial code. This feature allows a common set of partial station number digits to be registered as the speed dial code, and allows the station user to dial the remaining digits of each number to establish the call.

Consultation (Broker) Hold

This feature allows a station user to place one party on hold and confer with a third party on another line. This feature reduces call backs and improves customer service. Some systems allow the station user to toggle back and forth between two lines.

Customer Station Rearrangement

This feature allows station users to physically relocate their multiple line voice terminals internal to the system. When a station user moves between locations, the voice terminal station (number and COS) is logically transferred. This service is activated as follows: dialing the “moving” feature code followed by a personal code; the terminal is out of service; the terminal is disconnected, moved, and reconnected in the new location; the terminal is reinitialized by dialing its extension number, followed by the personal code; the station rings immediately and when answered, the set is validated. The former and latter locations must support the same category of voice terminal.

Discrete Call Observing

This feature allows a supervisor to monitor a conversation between an assistant and a caller on a preselected line. While the supervisor is listening, the voice terminal microphone is off and the assistant is informed that the feature is activated by a notice on the voice terminal display. During monitoring, the supervisor can take over the call.

Distinctive Ringing

This feature allows the system administrator to define distinct ringing patterns for different call types, such as internal, defined internal line, external, private network, emergency, and private line. Station users can use this feature as a call screening device to decide which calls to answer and which are to be forwarded to a coverage station. The number of distinct patterns differs greatly between different system models.

Dial by Name

This feature allows a station user with a voice terminal equipped with an alphabetical keyboard to call an internal extension or external number by typing in a name using last name, first name, or initials. The directory database can be locally stored in the voice terminal or accessed from a centrally located database in the PBX system or application.

Do Not Disturb

This feature allows a voice terminal user to request that no calls, other than priority calls, terminate at a particular extension number until a specified time. At the specified time, the system automatically deactivates the feature and allows calls to terminate normally at the extension.

Elapsed Call Timer

This feature provides a display of the elapsed time when a multiple line voice terminal is connected to any trunk circuit.

Emergency Access to Attendant

This feature allows emergency calls to be placed to an attendant with special priority status. Calls can be placed automatically when the telephone is in an off-hook state or by dial access. The attendant receives visual and audible feedback when the call is received. The feature is important for situations requiring immediate attendant access.

Executive Busy Override

This feature allows a station user to cut into an internal party’s conversation. This feature decreases call backs, saves time reaching the called party, and decreases calls sent to coverage positions.

Executive Calling

This feature allows a station to be assigned VIP class status. The feature allows a VIP station to send a special ringing signal to a called station when idle and automatically send multiple tone bursts to that station when busy.

Executive Access Override

This feature allows a station user to connect a call to an internal extension that is in call forward or do not disturb mode. The service is authorized by COS level.

External Paging with Meet-Me

This feature allows a station user or attendant to dial a local paging equipment access code and connect both parties automatically after the paged party has answered the page and dials a special access code.

Facility Busy Indication

This feature provides multiple line telephone users with a visual indication of the busy or idle status of internal station numbers, trunk groups, hunt groups, or paging zones. Station users can monitor the activity of frequently called numbers with this feature, eliminating encountered busy signals.

Group Listening

This feature allows a station user of a multiple line voice terminal with an integrated speakerphone to place a call using the handset and activate the terminal’s built-in speaker, to allow others to listen to the conversation while the station user continues talking through the handset.

Hands-free Dialing

This feature allows a station user of a voice terminal with a built-in speaker to dial and monitor a call without lifting the handset.

Hands-free Intercom

This feature allows a station user of a voice terminal with a built-in speaker to answer a voice call without lifting the handset. The incoming voice call is heard over the speaker.

Help/Information Key

This feature provides a station user of a multiple line voice terminal with display immediate access to help menus for terminal programming and feature access procedures. Information is displayed by the system in user-friendly way. If a feature access code is changed, it will be displayed automatically in the feature menu. Service consultation shows all the relevant functions and their associated feature codes. This feature allows self-training on the voice terminal and also reduces the need for paper labels on telephone features.

Hot Line

This feature automatically dials calls to preassigned internal stations, off-premises stations, or feature access codes when the handset is lifted. It eliminates the need to dial a number or access code, thereby simplifying and accelerating the process.

Incoming Call Display

This feature, available on telephones with display fields, provides visual notification to the station user of the calling party’s station number or incoming trunk group name. The calling party’s name may also be displayed with the station number. The feature is a screening device to decide whether to answer or divert the call.

Individual Attendant Access

This feature allows users to access a specific attendant console. Each attendant console can be assigned an individual extension number.

Intercom Dial

This feature allows multiple line voice terminal users to gain quick access to select other voice terminal users within an administered group. Calling voice terminal users lift the handset, press the dial intercom button, and dial the one- or two-digit code assigned to the desired party. The called user receives an alerting tone, and the status lamp associated with the Intercom button, if provided, flashes.

Last Number Redialed

This feature stores the last number dialed by the station user and allows the station user to automatically dial the number by using a programmed feature button or feature access code. It simplifies the calling process, reduces misdialed calls, and saves time.

Line Lockout

This feature removes single line voice terminal extension numbers from service when users fail to hang up after receiving dial tone signals, fol- lowed by intercept tone signals. The intervals for each tone signal are administrable.

Loudspeaker Paging Access

This feature provides station users or attendants dialing access to voice paging systems. This is useful for paging purposes regardless of the station user’s location within the premises environment. It is often used with the call park feature.

Malicious Call Trace

This feature allows a station user to notify a predefined set of station positions that a malicious call is in process. The notified station users can then gather information and data about the call to identify the calling source. The feature is useful when a CLID or ANI is not displayed.

Manual Intercom

This feature allows a station user to call a manual intercom group member by pressing the manual intercom button. All member of a manual intercom group share a common signaling path. When the manual intercom button is pressed, a special tone burst is sent over the voice terminal speakers of all group members. When a group member answers, a speech path is established.

Manual Originating Line Service

This feature connects single line voice terminal users to the attendant automatically when the user lifts the handset. The attendant code is stored in an abbreviated dialing list. When the manual originating line service voice terminal user lifts the handset, the system automatically routes the call to the attendant using the hot line service feature.

Manual Signaling

This feature allows a voice terminal user to signal another voice terminal user. The receiving voice terminal user hears a short burst of tone. The signal is sent each time the button is pressed. If the receiving voice terminal is already being alerted with an incoming call, manual signaling is denied.

Message Waiting

This feature enables multiple line appearance voice terminal users, by pressing a designated button on their own terminals, to light the status lamp associated with the message waiting button at another multiple line appearance voice terminal. Activating the feature causes the lamp to light on the originating and receiving voice terminals. Either terminal user can cause the lamp to go dark by pressing the button.

Multiparty Conferencing

This feature allows multiappearance voice terminal users to set up multiparty conferences (typically between four to eight station users) without attendant assistance.

Music on Hold

This feature provides music to a party that is on hold, waiting in a queue, parked, or on a trunk call that is being transferred. The music lets the waiting party know that the connection is still in effect. The system provides automatic access to the music source.

Off-hook Alarm

This feature provides a special alerting tone to a station user who does not hang up the handset after receiving a busy signal. The tone signal is sent after a programmed interval.


This feature allows a station user to temporarily prevent outgoing calls from the voice terminal. The selection of an external line by feature code, programmable key, or supervision key is controlled. Dialing the appropriate feature code followed by the personal code reactivates direct access.

Paging/Code Call Access

This feature allows voice terminal users, attendants, and tie trunk users to page with coded chime signals. Multiple individual paging zones can be provided.

Personal Speed Dialing

This feature allows a station user to program personal speed dial numbers at the station instrument. The speed dial feature is activated by using a programmed feature button or access code and pressing a one-or two-digit access code. The feature simplifies the dialing process, saves time, and decreases misdialed numbers.

Personalized Ringing

This feature allows users of certain voice terminals to uniquely identify their own calls. Each user can choose one of a number of possible ringing patterns.

Priority Calling

This feature provides a special form of call alerting between internal voice terminal users. The called voice terminal user receives a distinctive, administrable alerting signal.

Private Line

This feature provides a dedicated trunk for direct access to or from the public network for multiple line appearance voice terminal users.

Privacy—Attendant Lockout

This feature prevents an attendant from re-entering a multiple-party connection held on the console unless recalled by a voice terminal user.

Privacy—Manual Exclusion

This feature allows multiple line appearance voice terminal users to keep other users with appearances of the same extension number from bridging onto an existing call.

Recall Signaling

This feature allows an analog station user to place a call on hold and consult with another party or activate a feature. After consulting with that third party, the user can conference the third party with the original party by another recall signal or return to the original party by flashing the switchhook twice.

Ringer Cutoff

This feature allows the user of a multiple line appearance voice terminal to turn certain audible ringing signals on and off. The feature does not affect visual alerting.

Ringing Tone Control

This feature allows station users of multiple line voice terminals to select from a menu of ringing tone melodies and to adjust the volume level of ringing.

Save and Redial

This feature allows a station user to save a specific dialed number and then redial the number at a later time. The station user stores and redi-als the number by pressing a save and redial feature key.

Secondary Extension Feature Activation

This feature allows a multiple line voice terminal station user to access a line appearance of another extension, and program a limited set of features, such as call forwarding and call pickup, from that extension.

Send All Calls

This feature allows users to temporarily divert all incoming calls to coverage regardless of the assigned call coverage redirection criteria. The feature also allows covering users to temporarily remove their voice terminals from the coverage path.

Step Call

This feature allows a station user or attendant, after dialing a busy station, to dial an idle station by simply dialing an additional digit. The feature can be implemented only if the dialed digits of the first dialed number and the second number are identical, except for the last digit.


This feature allows a station user of a multiple line voice terminal to store a particular number for later use. A store/redial key is programmed and assigned to this function.

Supervisor/Assistant Calling

This feature allows a station user with a multiple line voice terminal, who is an assistant to a supervisor, to use a call appearance of the supervisor’s station to screen calls for the supervisor and announce and/or transfer calls to that extension. The assistant can also dial the supervisor during a busy condition and send a message waiting notification to the supervisor.

Supervisor/Assistant Speed Dial

This feature allows a pair of station users to use a programmed feature key to direct speed calls between a supervisor and an assistant, even if forwarding is validated.

Text Messages

This feature allows station users to leave a short text message for other internal users. Messages are stored in the main system database, and are available for selection via a menu on display-based voice. Calling parties can also receive messages from a voice terminal station that are preselected by the called party during no answer or busy conditions. There may be three structures of messages: preprogrammed fixed messages fully defined by system management, part programmable messages defined by system management but to be completed by the station user if the voice terminal has an alphanumeric keyboard, and fully programmable messages written entirely by the station user and offered only by sets provided with an alphanumeric keyboard.

Timed Queue

When a multiple line voice terminal station user originates an outgoing trunk call and encounters a no answer or busy condition, the timed queue feature can be implemented. After pressing a programmed feature button or dialing the feature access code, trunk seizure is repeated and the external station number is dialed after a predetermined interval.

Trunk Flash

This feature enables multifunction voice terminals to access CLASS features that are provided by the far-end CO switching system located directly behind the PBX system. CLASS services are accessed by a sequence of flash and dial signals from the station on an active trunk call. The feature can decrease the number of trunk lines connected to the PBX system by performing trunk-to-trunk call transfers at the far-end CO, which eliminates the use of a second trunk line for the duration of the call and frees the original trunk line for the duration of the call. It can also be used to set up a conference call with a second outside call party, which eliminates the need for a second trunk line for the duration of the call.

Trunk-to-Trunk Connection

This feature allows a station user to conference together two outside trunk calls and abandon the connection without dropping the two trunk-to-trunk connections.


PBX Systems Management and Administration: System Diagnostics and Maintenance

The primary objective of system maintenance is to detect, report, and clear trouble as quickly as possible with minimum disruption of service. Periodic tests, automatic software diagnostic programs, and fault detection hardware allow most troubles to be traced to an individual assembly in the system. System diagnostic functions include:

  • Monitoring of processor status

  • Monitoring and testing of all port and service circuit packs

  • Monitoring and control of power units, fans, and environmental sensors

  • Monitoring of peripherals (voice terminals and trunk circuits)

  • Initiating emergency transfer and control to backup systems

  • Originatng alarm information and activate alarms

There is a specific maintenance strategy and plan for each of these hardware elements monitored by the system.

The maintenance subsystem software is responsible for initializing and maintaining the system. This software continuously monitors the system and maintains a record of detected errors. The maintenance subsystem also provides a user interface for on-demand testing and contains two general categories: system alarm troubles that are automatically reported to a local maintenance terminal or a remote maintenance center and user-reported troubles resulting from service problems at individual station user terminals.

The major part of maintenance is system-alarmed troubles. PBX system diagnostic circuitry detects and reports most problems automatically. When the trouble is repaired and no longer detected, the alarm is automatically retired. It is not necessary for personnel to retire alarms after a problem is corrected. Dedicated maintenance circuit packs or daughterboards are used in fault detection and repair at many system levels, including the common control complex, expansion cabinets and carriers, and a variety of trunk interface cards, particularly those used for digital trunk connections. Almost all circuit packs have LED indicators to indicate alarm conditions (red) if the system has detected a fault in that circuit pack. A yellow alarm condition indicates the system is running tests on that circuit pack, and a green condition indicates that the circuit pack is operating without problem. In-line error detection circuitry checks for correct operation.

Maintenance tests can be periodic or on demand. Periodic tests run automatically at fixed intervals on a specific schedule. Usually, short tests are run hourly or less; long tests are run every 24 hours. Demand tests are run by the system when it detects a need or by personnel when required during trouble-clearing activities. Demand tests include the periodic tests, and other tests are required only when trouble occurs. Some nonperiodic tests may be disruptive to system operation. From a terminal, personnel can initiate the same tests the system initiates, and the results are displayed on the terminal screen.

If any part of the system fails any portion of the periodic tests a preset number of times, the system automatically generates an alarm. There are three alarm types common to most systems:

  1. Major alarms—Failures that cause critical degradation of service and require immediate attention.

  2. Minor alarms—Failures that cause marginal degradation of service but do not render a crucial portion of the system inoperable. This condition requires action, but its consequences are not immediate. Problems that cause minor alarms might be impaired service in a few trunks or stations or interference with one feature across the entire system.

  3. Warning alarms—Failures that are localized and cause no noticeable degradation of service. Warning alarms are not reported to the attendant console or a remote location.

The PBX system can usually send an alarm to any customer device such as a light, automatic dialer, a bell, or other equipment. The alarm activation level field on the system parameters maintenance screen must be administered to indicate the alarm level (major, minor, warning, or none) that activates the alarm device.

If the maintenance software detects an error condition related to a specific maintenance element, the system will automatically attempt to repair a problem or operate around it. If a hardware component incurs too many errors, an alarm is generated. Records of each error and alarm are stored. The error log is a record of system errors and can be accessed from a SAT. The error log is useful for analyzing problems that have not caused an alarm or when alarms cannot be retired by replacement of hardware. When errors result in alarms, the alarms are listed in the alarm log. This log can be displayed on a terminal. If several alarms are active, the alarm log can be used to determine the alarms that should be cleared first.


PBX Systems Management and Administration: Performance Management

PBX performance management records and reports are typically available for the following system measurements.

Trunk Usage and Traffic

Trunk traffic records are kept for all inbound and outbound calls and identify the trunk group and trunk channel, time of call, and duration of call. Individual trunk line counters can measure the number of call attempts, blocked trunk lines, and traffic intensity (Erlangs). Outgoing counters can measure the number of outgoing attempts, successful calls overflowing to another route, calls lost due to blocking, blocked trunks in measurement, and traffic intensity (Erlangs). Incoming trunk route counters can measure the number of incoming call attempts, trunks in the measurement, number of blocked trunks in the measurement, and traffic intensity (Erlangs). Similar statistics are measured for two-way trunk routes.

Attendant Consoles

Attendant counters can measure all attendants in the system or individual attendants positions. Record measurements include number of answered calls, number of calls initiated by attendant, accumulated handling time for all calls, accumulated handling time for recalls, accumulated handling time for calls initiated by attendant, accumulated total delay time for recalls, number of answered recalls, number of abandoned attendant recalls, accumulated waiting time for abandoned calls to an attendant, accumulated waiting time for abandoned recalls, and accumulated response time for all types of calls.


Station counters can measure individual stations or station group traffic statistics such as number of calls, number of stations in the measurement, number of blocked stations in the measurement, and traffic rating (Erlangs).

Traffic Distribution

Traffic distribution across the internal switching network can be measured for each local TDM bus, traffic over each highway bus, and traffic across the center stage switch by each switch network interface link.

Busy Hour Traffic Analysis

Busy hour traffic analysis measurements for trunks, stations, and the internal switch network can be performed. Busy hour traffic intervals can be programmed for any time of day. Erlang ratings are calculated for individual trunk lines, individual trunk groups, and all trunk groups. CCS ratings are calculated for individual stations or groups of stations.

Processor Occupancy

System call processing performance is measured in terms of BHCs (attempts and completions). The percentage of maximum call processing capacity is reported for programmed intervals. Threshold reports can be generated to monitor system load factors.

Threshold Alarms

For a variety of system hardware devices, it is possible to define a congestion threshold value and measure generated alarms. Alarms are recorded in an alarm record log. The types of devices that can be tracked are tone receivers, DTMF senders and receivers, conference bridges, trunk routes, and modem groups.

Feature Usage

Feature usage counters for selected station features (e.g., call forward, call transfer, add-on conference) and attendant system features (e.g., recall, break-in) can be measured and reported for programmed intervals.

VoIP Gateways

IP-PBX systems collect and store data to track the usage and performance of IP gateway devices, IP phones, and VoIP trunk calls. VoIP information reports include tracking of IP gateway devices and calls that pass through each gateway, gateway congestion, assignment of services or routes to gateways, tracking of phone numbers dialed or originating off-site numbers, and IP gateway addresses.


CDR data is compiled for all successful incoming and outgoing trunk calls. Call records can be stored in multiple formats (fixed and programmed) per output device. Fixed formats typically conform to standards published by leading call accounting software suppliers, or are proprietary to the PBX system. Programmable formats provide a flexible means to incorporate new data elements in the call record. A variable format allows a record to be defined in terms of its content (from a set of available data elements) and the position of the data elements in the record. This method can be used to construct custom formats.

A system administrator may define programmable CDR formats based on available CRD field data records. Call record fields typically include:

  • Date

  • Time

  • Call duration

  • Condition code (categorizes information represented in the call record)

  • Trunk access codes

  • Dialed number

  • Calling number

  • Account code

  • Authorization code

  • FRL for private network calls

  • Transit network selection code (ISDN access code to route calls to a specific interexchange carrier)

  • ISDN bearer capability class

  • Call bandwidth

  • Operator system access (ISDN access code to route calls to a specific network operator)

  • Time in queue

  • Incoming trunk ID

  • Incoming ring interval duration

  • Outgoing trunk ID

Reports can be generated for any or all of the call record field data.

CDR data is not usually compiled for intraswitch calls (station to station, station to attendant), calls terminated by busy signals, and calls with no answer. When CDR was introduced as a system option in the early 1980s, memory storage was expensive. Any call that did not incur a direct expense was not recorded and stored. Today, PBX systems based on nontraditional designs, such as CTI-based server systems, can collect, store, and process these data records for reporting purposes. Traditional circuit switched PBXs may optionally record a limited number of intraswitch calls for select calling stations or capture data for all calls using optional CTI solutions.

CDR records and call accounting reports are vital to the monitoring and management of the PBX system. It is important to monitor call costs and usage to:

  • Bill system subscribers for their communications network use

  • Budget and allocate usage charges by department

  • Resell telephone services to outside clients

  • Monitor PBX effectiveness

  • Gather statistical data for performance benchmarking

  • Prevent or minimize telephone system abuse and unauthorized access

  • Verify monthly service provider bills

There are several optional PBX reports that are useful to system administrators.


Directory records can include each subscriber’s name, with a variety of phone numbers such as primary, published, listed, emergency, and alternate, and authorization code information, job title, employee number, current employment status, and social security number.


Inventory records and management is used to administer any kind of inventory product part: PBX common equipment (cabinets, carriers, circuit cards), voice terminals and module options, jacks, and button maps. The reports allow administrators to accurately re-charge items. Inventory can be tracked by data such as user, system (PBX or other networks), jack, serial number, asset tags, trouble calls, recurring and nonrecurring costs, and general ledger codes. The inventory management system also includes records containing the following data: purchase date, purchase order number, depreciation, lease dates, and manufacturer and warranty information.


Cabling records keep track of all cable, wire pairs, distribution frames, wiring closets, and all connections (including circuits) down to the position and the pair level. Records include starting and ending locations, description, type, and function. Individual cable lengths are maintained and automatically added, as is the decibel loss for the entire path. Information can be provided on the status of all cable runs and the number of pairs they contain, the status of the pairs, and the type of service they provide.


PBX Systems Management and Administration: Administration Sequence

After the system is installed, the system administrator must enter the translation data into the system memory via the SAT. Translation data is taken from survey sheets and previous system records and provides a blueprint of what needs to be programmed into the system configuration. When entering the translation data into the system, the system administrator should periodically save the translations on tape. This creates a nonvolatile copy of the translation already entered into the system. If a power outage or system failure occurs, the translation data saved on the tape will not have to be entered again.

PBX system features should be entered into the switch in an ordered manner. The following is the recommended order in which data should be entered into the system:

  1. Login and password (change password, if necessary)

  2. Dial plan

  3. Feature access codes (FACs)

  4. System features (class of service and class of restriction)

  5. Console parameters

  6. Attendant consoles

  7. System parameters

  8. Voice terminals

  9. Data modules

  10. Network connection channels

  11. Bridged line assignments

  12. Group assignment (hunt groups, call coverage, pickup groups, etc.)

  13. Trunk groups

  14. Paging/code call zone assignments

  15. ARS table

Before the customer database data is programmed into the system, the system administrator should review the system hardware configuration to assess the available port circuit interface boards and design layout. Many administration interface screens will require the administrator to input a port or slot identifier. A port or slot is an address that describes the physical location of the installed equipped. Port addresses consist of cabinet, port carrier, card slot, and port circuit card termination identifiers. Each hardware component has a multidigit identifier, and the combination of the hardware component identifiers is the port address.

Dial plan and FACs must be administered before voice terminals, hunt groups, pickup groups, coverage groups, and attendant consoles can be administered. Default values for the dial plan can be changed if they do not meet customer requirements. A standard dialing plan usually supports four- or five-digit extension numbers, but some customers may require more extension digits for system subscribers. For example, a seven-digit dial plan may be required for multisystem intelligent transparent private network configurations. Default FACs can also be changed, but the number of digits assigned to the FAC must agree with that of the dial plan.

The dial plan is used by the system to interpret dialed digits and know how many digits are expected for different call types, such as intercom calls or trunk calls. An important element of the dial plan is the first-digit table. The first digit dialed by a station user may have any one of the following codes: attendant, dial trunk access, extensions, feature access, and miscellaneous (used when more than one code begins with the same digit and requires a second-digit dial table).

Regarding first-digit dialing, North American station users are accustomed to dialing 9 for dial trunk access and 0 to reach an attendant console. In most of Western Europe, the first-digit dial trunk access code is 0, which usually proves confusing to American tourists. Many station users do not understand that access codes are programmable by system administrators, although most systems use the default values programmed by the manufacturer, such as 9 for trunk access in North America.

Miscellaneous codes are usually required when there might be a problem interpreting the first dialed digit. For example, when local 911 emergency services were first introduced, major PBX problems occurred. PBX systems that were programmed to recognize 9 as the first-digit dial trunk access code did not recognize the second dialed digit, 1, as a valid area or exchange code. System administrators were forced to reprogram their first digit tables to interpret a 911 call. Similarly, the revised North American Dialing Plan (NADP) introduced in the mid-1990s forced a reprogramming of the dial plan because trunk calls outside of the local area code required dialing a 1 before an area code for interpretation by local central office switches, and digit restrictions for the second area code digit (previously 0 or 1) were modified. Continuing changes in PSTN dialing requirements require constant updates of PBX dial plans.

Once the dial plan and FACs have been assigned, the system administrator can add voice terminals to the system. A variety of programming commands simplifies the configuration process. For example, the duplicate command can add the same types of voice terminals, instead of repetitive programming of similar information. The terminal extension number, location, type, and user name are entered on the display form with labeled blank fields. For an IP-PBX system, IP voice terminals require similar data entry for the voice station display screen, but also require entry of IP addresses, MAC addresses, and voice codec information. IP addresses are usually assigned by a DHCP server but can be manually administered. QoS programming for IP voice terminals is the responsibility of the LAN administrator, as is performance monitoring of IP telephony metrics, such as call delay and packet loss.

A common misconception is that IP-PBX systems don’t require traditional MAC administration because IP voice terminals can be initialized via a DHCP server or physically moved to different LAN connector outlets without administration programming. Despite these capabilities, subscriber and voice terminal parameters must be input for all IP peripherals. With regard to moves, IP-PBX system IP voice terminals require similar data entry for the voice station display screen, but also require the entry of IP addresses, MAC addresses, and voice codec information. IP addresses are usually assigned by a DHCP server but can be manually administered. All PBX systems support a customer station rearrangement feature that allows the movement of digital telephones between telecommunications outlets without administration intervention, exactly like an IP terminal.

Attendant consoles must be added one at a time, and for reliability, attendant consoles should not be assigned to the same circuit pack. Data modules can be assigned after voice terminal administration. Some data modules must be added during voice terminal administration if the voice terminal has a data module. Other data modules can be added separately.

Network connection channels are used to provide switched data access for the following features and functions:

  • CDR

  • SATs

  • Remote SATs

  • Property management system (PMS) link

  • PMS log printer

  • Journal printer

  • Recorded announcements

  • System printer

Group assignments can be programmed after voice terminals are added. The following groups can be administered:

  • Abbreviated dialing (system, group, enhanced)

  • Hunt groups

  • Call coverage answer groups

  • Pickup groups

  • Intercom groups

  • Terminating extension groups

  • Trunk groups

The ARS tables support network access to the PSTN and private networks. Trunk groups must be programmed to ARS. Access to private network facilities include the following network interface types:

  • DS1 interface

  • TTTN

  • Private tandem network


  • VPN



Qsig is an inter-PBX signaling system designed for multiple PBX system platform networks. The proprietary nature of IFTN solutions restricted customer configuration flexibility to a single supplier’s product platform. Qsig in its current form originated during the 1990s as a standardization effort by the IPSN Forum, a group of Western European PBX equipment suppliers, with Siemens and Alcatel at the forefront of the movement. IPSN work efforts were handed off to the ECMA and the International Telecommunications Union (ITU) for the formalization of issuing standards and specifications. Qsig is based on the ITU’s Q.93x series of recommendations for basic services and generic features and Q.95x series for supplementary services.

The major benefits for developing Qsig were outlined in the Qsig handbook originally published by the IPNS Forum.

Vendor independence. The nonproprietary nature of Qsig, based on open international standards and supported by all of the leading global PBX suppliers, allows customers to configure an intelligent communications system network when using equipment from more than one supplier.

Guaranteed interoperability. A memorandum of understanding (MoU) signed by the leading global suppliers signifies commitment to Qsig specifications, facilitates interoperability performance tests, and assures customers that they will be able to operate communications networks with a mix of supplier equipment.

Free-form topology. Qsig does not impose the use of a specific network topology, so it can be implemented with any network configuration: meshed, star, main/satellite, etc. Existing networks can, regardless of their topology, be upgraded to Qsig. Newly designed networks can be installed with the most effective and economical topology.

Unlimited number of nodes. There are no nodal limits for a Qsig network. New nodes can be added as needed.

Flexible numbering plan. Qsig does not impose any number plan restrictions for the network, thereby allowing customers to freely adopt customized numbering plans.

Flexible interconnection. Qsig will work over any type transmission network for linking PBX systems, including two- and four-wire analog tie lines, digital leased lines (including ISDN PRI and BRI), radio and satellite links, and VPN services provided by interchange carriers. Associated transmission delays are managed and controlled according to Qsig specifications.

Public ISDN synergy. There is network service compatibility between public and private ISDN transmission facilities. Applications developed for desktop terminals connected directly to a public ISDN network will also be available to desktop terminals provisioned within the Qsig-based customer private network.

Supplementary services for corporate users. Qsig supports private communications features beyond those defined for public ISDN networks, including caller name ID, call intrusion, do not disturb, path replacement, operator services, mobility services, and call completion on no reply.

Feature transparency. Features and functions supported by any network node can be transparently supported across the network to station users configured behind other network nodes. Qsig is structured and organized to adapt to service levels offered by different PBX systems, and it allows each network node to provide only the required level of service. There is an exchange of high-level services between any two nodes, via transit nodes with lower service levels: transit nodes pass communications and control signals between systems.

Innovation. Qsig does not restrict individual PBX manufacturers from developing customized, unique features. A special mechanism within Qsig, generic functional procedures (QSIG GF), provides a standardized method for transporting nonstandard Qsig features. As defined in Qsig GF, the basic rules related to feature transparency allow end-to-end communication through the network, regardless of network structure. Qsig does not prevent the use of innovative, proprietary system features across the customer network and allows for customized new feature development negotiation between PBX suppliers and customers.

Multiapplication domain. Qsig is not restricted to PBX systems and can support applications requiring other peripheral communications equipment, such as VMS, fax servers, data processing equipment, and multipoint conferencing systems.

Evolution. Qsig has an evolutionary path to support communications features, functions, and applications that are developed in the future.

Qsig Architecture

Qsig standards specify a signaling system at the ITU-T ISDN “Q” reference point, which is intended primarily for use on a common channel, although Qsig can be implemented over any suitable inter-PBX connection platform. The “Q” reference point, the logical signaling point between two PBXs, was defined explicitly for the Qsig. The physical connection point to the PBX system is made at the “C” reference (also a new ISDN reference point). There are three sublayer protocols at Layer 3, including the Qsif GF procedures. Qsig GF protocol provides a standardized mechanism to exchange signaling information for the control of supplementary services and additional network features (ANFs) over customer networks.

Qsig basic call (BC) message sequence is an intermediary transit node linking two endpoint PBX systems. Qsig BC is a symmetrical protocol designed for peer-to-peer networks, and it includes transit node capability.

ECMA also has been working on enhancements to its Qsig specifications to support broadband PBX networks. B-Qsig is an extension of Qsig, using many standards as possible available from the ITU-T and ATM forums.

Qsig Supplementary Services and ANFs

The following is a listing of the Qsig supplementary services and ANFs:

  • Advice of charge

  • Call completion

  • Call forwarding and diversion

  • Call interception

  • Call intrusion

  • Call offer

  • Call transfer

  • Call waiting

  • Direct dialing in

  • Do not disturb

  • Identification services

  • CLID presentation

  • Connected line identification presentation

  • Calling/connected line identification restriction

  • Calling name identification presentation

  • Calling/connected name identification restriction

  • Mobile

  • Multiple subscriber number

  • Operator services

  • Path replacement

  • Recall

  • Subaddressing

  • User-to-user signaling


IFTN Features and Functions

There is no standard level of IFTN feature/function transparency within the PBX industry. Some PBXs support a very high percentage of features and functions across multiple networked systems, up to 90 percent of the total generic software program, and some support less than 50 percent. Almost all PBX system IFTN options support the following basic feature/functions:

  • Basic calling with the use of a flexible dialing plan (typically four or five digits)

  • Voice terminal display information (calling party/called party name and number, call redirection information)

  • Call forwarding services

  • Call transfer

  • Call conferencing

  • Automatic callback

  • Bridged call appearance

  • Message waiting indication

  • Trunk release

  • Network-wide attendant services

  • Network-wide CDR

An important category of features supported by only a few IFTN solutions is ACD. For example, all 55 of the identifiable NEAX 2400 IPX ACD features are available with Fusion CCS. The NEC ACD Agent Anywhere option is an intelligent network of multiple ACD systems using Fusion CCS links. ACD nodes can communicate with each other and pass and interpret signaling, caller ID, call prompt, and database information across the network. Intelligent interflow routing of callers between nodes improves customer service levels, balances traffic load, and optimizes agent productivity. Fusion CCS also supports centralized management reporting and supervisor workstation data screens. A multiple system ACD network has built-in redundancy to reduce system down time and increase customer satisfaction.

The Agent Anywhere option supports distributed ACD agents behind switch nodes remote from a centralized ACD processor node. ACD agents can be deployed anywhere within a Fusion CCS network, with the only restriction being that the remote switch node be directly linked, (no pass through signaling), to the control switch node. Agent Anywhere can be implemented when using internal or external ACD processing and software options. The Fusion CCS solution supports decentralized agents assigned to the same call split across multiple nodes. Incoming calls to the central control node can be routed to available agents in remote nodes if all agents are busy at the central location. For configu- rations with local incoming trunking at remote node locations, calls can be queued at the central control node location when no remote agents are available.

Many customers’ ACD-based call center systems include a CTI application server. A centralized CTI application server capable of supporting more than one PBX/ACD switch node is less costly and easier to manage and maintain than application servers at each customer location. The NEC Fusion CCS option supports a centralized CTI application server for ACD systems. Another good example of a centralized application server solution used within an IFTN configuration is the Siemens HiPath Allserve 150. A centralized Windows NT application server can support a network of one to four Siemens Hicom 150H systems networked with the Siemens CorNet option implemented over an IP LAN/WAN. The applications, run on a single server for all networked PBXs, include messaging, call center, personal call manager, and call accounting.

Perhaps the most important transparent system operations are management and control from a centralized application server. A central database for all Hicom 150H switch nodes resides in the application server. The centralized server provides one access point to administer and maintain each system. Station move/add/change transactions are implemented as if there were a single system, not multiple switch nodes. A single management or maintenance command can be applied to all switch nodes across the network, instead of inputting individual changes to individual systems. Centralized management system capabilities for all move, add, and change procedures is an important IFTN capability that is not commonly supported by most traditional circuit switched PBX IFTN options but is supported by more of the newer client/server IP-PBX system designs, such as Cisco’s AVVID IP telephony system.

Shared applications resources for call center, messaging, and management operations are an important IFTN cost savings benefit. The first IFTN offerings were limited to shared VMS applications. One VMS supported station users across a network of PBX systems. One of the cost savings components is attributable to the lower price for a single, very large messaging system as opposed to the collective cost of several smaller systems with equal voice mailbox and storage capabilities. Another cost savings component is ongoing management and service. Maintaining one messaging system is less costly and more efficient than maintaining several systems. The same cost savings criteria can be attributed to other shared application resources in an IFTN configuration.


Intelligent Feature Transparent Network (IFTN)

AT&T’s original ETN offering evolved into DCS in 1982 when the U.S. Navy required a single communications system for its San Diego naval base operations, with a port capacity far greater than the parameter limitations of any single PBX system model available at the time. The AT&T solution was not to design an extremely large PBX system but to intelligently network multiple systems to provide the appearance of one system for most common internal station-to-station user operations. Originally known as the Defense Metropolitan Area Telecommunications System (DMATS), the AT&T Dimension PBX option was renamed the Distributed Communications System (DCS). The AT&T Dimension DCS option became very popular in a short period and forced competitors to develop IFTN offerings of their own. Among the other wellknown IFTN brand name options developed almost 20 years ago and still marketed today are Siemens CorNet, NEC CCIS (since enhanced to Fusion CCS), and Fujitsu FIPN.
An IFTN has the property of transparency with respect to all on-net calling, and many feature operations. Transparency is the ability of the system, from a station user’s perspective, to operate across multiple network nodes in the same way it does at the local node. This allows for a limited digit dialing plan for all on-net calls, thereby eliminating the need for PBX location codes and network extension numbers. All intercom calls are dialed with extension numbers corresponding directly to station user directory numbers.

An IFTN design is based on the ETN model, a hierarchical layer of switching systems interconnected using tie trunks. Direct links between each PBX network node are not required, but there are limitations on the number of transit nodes that can pass intelligent control signals between the originating and terminating nodes. The passing of call handling signals between network nodes is what distinguishes an IFTN from an ETN. Out-of-band common channel signaling techniques are used. Each manufacturer’s IFTN offering was based on a proprietary signaling and messaging scheme, thereby limiting the flexibility of the customer network design, although the competing suppliers have worked together during the past decade to develop industry standards for open inter-PBX networking solutions 

The idea behind an IFTN is to have the common control complexes of multiple PBX systems communicating with each other to transmit data, customer network information, and command messages for a single system image. Specific details concerning how each PBX system implements its proprietary IFTN service offering are not available. The first implementation of AT&T’s DCS offering was based on analog tie trunks and channel-associated private line data circuits for nodal transmission links. When T1-carrier circuit services became available, clear channel signaling techniques were used instead of dedicated data circuits, and analog tie trunks were replaced with voice communications transmission. Twenty-three 64-Kbps channels of the T1-carrier circuit were used as bearer voice communications channels, and one 64-Kbps channel was used for inter-PBX signaling and messaging.

When ISDN PRI services became available in the late 1980s, the B-channels were used for voice communications and the D-channel was used to transport the control information. The most common signaling method used for IFTN networks, based on ISDN PRI service circuit links, is a temporary signaling connection (TSC). A TSC provides a temporary signaling path through ISDN switches for exchanging information between users. There are two TSC types: call associated CA-TSC and non–call associated NCA-TSC.

A CA-TSC refers to a service for exchanging user information messages that are associated with an ISDN B-channel connection by the call reference value of the control data packets. An NCA-TSC is a connection not related to any ISDN B-channel connections. It is an administered virtual connection established for exchanging user information messages on the ISDN D-channel. Once the NCA-TSC has been administered and enabled, it will be active for an extended period. There are two NCA-TSC types: permanent and as needed. A permanent NCA-TSC will remain established while the system is operating. If the connection is lost, an attempt will be made to re-establish it. An as-needed NCA-TSC is established based on user request and the available of TSC facilities. The connection is dropped after a preset period of inactivity.

ISDN PRI transmission services are currently the most commonly used communications and signaling transport links for IFTNs. Some IFTN offerings, such as Siemens CorNet, can be supported only when using ISDN PRI service circuits for circuit switched connections. Most PBX IFTN options also can be implemented with virtual networking services supporting a TSC, such as AT&T’s service offering. Other network transmission solutions that support inter-PBX message signaling are ATM trunk carrier services and TCP/IP over a LAN/WAN infrastructure. ATM networking options include T1-carrier CES and TDM/PCM conversion to ATM cell format for transmission over an ATM WAN. An important advantage of the TCP/IP networking option is that dedicated point-to-point signaling links between PBX network nodes are not required because point-to-multipoint signaling is supported by TCP/IP. Tandem switch nodes, the basic network element of circuit switched IFTNs, are not required if IP trunk circuits are used to pass communications and control/message signaling between switch nodes. This eliminates the transit node (hop-though) limitation for control signaling between originating and terminating switch nodes. There are several other advantages to using a LAN/WAN infrastructure as the IFTN network backbone:
  1. Reduced PSTN trunk carrier services in support of IFTN networking result in potentially significant cost savings. Using existing IP trunk circuits to carry IFTN voice traffic and signaling between switch nodes eliminates the need for dedicated private line and/or ISDN PRI digital trunk circuits. Fewer physical T1-carrier trunk circuits are needed to carry voice traffic over an IP network because VoIP transport uses voice encoding and compression algorithm standards.

  2. PBX system hardware costs decrease because fewer port circuit cards and port cabinet carriers are required. A single IP trunk circuit card can support a greater number of physical IP-based trunk circuit equivalents at a lower cost than traditional TDM/PCM station/trunk port circuit cards.
  3. Signaling support is to and from a single LAN-connected applications server across multiple PBX systems, instead of dedicated servers at each location.

  4. Network management and maintenance operations are simpler because a single converged voice/data network, instead of dedicated networks, is used to transport voice and data communications. An added benefit is a reduction in human and equipment management support resources.


Fundamentals of PBX Private Networks

The basic elements of a PBX private network are PBX switch nodes and tie trunks. Tie trunks are telecommunications channels that directly connect two PBXs. The first analog transmission tie trunks were known as E&M interface signaling trunks. The term E&M originally comes from the works earth (earth grounding) and magnet (electromagnetic generated tones). The letters E&M have also become known as “ear” and “mouth” or “receive” and “transmit.” E&M supervision signaling (on-hook/off-hook signaling) is used for a variety of networking operations, but the most basic function is to pass address signaling (called party number) between two endpoint PBXs in the private network. The most basic private network consists of two PBXs and at least one E&M tie trunk.

During the mid-1980s, E&M signaling was incorporated into T1 circuit digital private line services. In the early 1990s, ISDN PRI services were the primary trunking services used for private network tie trunk operations. By the end of the decade, IP-based trunk services could support traditional E&M signaling capabilities for inter-PBX networking requirements.

The first large private networks consisting or three or more PBX systems were known as tandem tie trunk networks (TTTNs). Each PBX was assigned a location code, and each station was assigned a private network extension number. TTTNs were based on a nonhierarchical network of tie trunks. A station user at the originating switch would dial the location code of the destination switch and wait to receive another dial tone signal before dialing the desired extension number. The request for dial tone was carried from switch to switch over E&M tie trunks. The private lines also were used to provide the dial tone signal back to the calling station user.

The first smart ETNs were designed in the late 1970s. An ETN was an enhanced version of a TTTN. It was based on a hierarchy of tie trunks and PBX switching nodes, with multiple call routes between network endpoints. A major innovation was an automatic alternate routing (AAR) program that selected a call route based on the number dialed and the most economic (or preferred) route available at the time the call was placed. The tandem switch node in the early ETNs was equipped with routing tables for determining the best route for on-net calls, had the capability of modifying outpulsed digits (for rerouting and directing calls), and could allow or deny call routing privileges to certain station users. Switch nodes in the ETN one level below the tandem switches are known as main switches. Main switches have trunk circuits for local CO switching access, but all network calls must be routed through a directly linked tandem node. Switches working behind the main nodes are known as satellites and tributaries. These switches are equipped with tie trunks to the main node only. They lack local CO trunk services and attendant operator positions. CO and private network access is through the main switch for all off-premises incoming and outgoing calls.

PBXs with ETN options support the important features described in the next sections.

Uniform Dial Plan (UDP)

A UDP provides for a common multidigit (usually four or five) dial plan that can be shared across a group of networked PBXs for interswitch and intraswitch dialing. The UDP includes a network location code, comparable to a CO code, and a multidigit extension number. The UDP extension number is not necessarily the same number as the station directory number. The network location code is often designated as the RNX; the extension number as XXXX. The network location code determines how the call is routed. Whenever a UDP is used to route a call, the number it outputs is in the form of RNX-XXXX. This must be taken into account so that the correct digit deletion or insertion can be specified within the routing pattern, so that the receiving switch receives digits in the format it expects. The UDP software program automatically translates the RNX- XXXX number to the directory number associated with the called station for digit analysis and routing by the destination PBX system.

Automatic Alternate Routing (AAR)

AAR provides alternate routing choices for on-net calls carried over the private network. Based on a routing table designed by the customer, the AAR software automatically selects the most preferred (usually the least expensive) route over the hierarchical tie trunk network. If the first route in the program table is not available, another route may be selected if the station user calling privileges warrant a more costly route based on the individual’s FRL. Large PBX systems can support several hundred different routing patterns (originating and terminating nodal pairs), with more than a dozen different call routes per routing pattern. The AAR routing patterns are shared with the ARS feature.

AAR also provides for digit modification to allow on-net calls to route over nonprivate PSTN trunk facilities when an on-net call route is not available. Calls rerouted off the network require digit modification to translate an RNX-XXXX number to a direct distance dialing number for national or international calling. On-net calls that are routed off-net are then controlled by the ARS feature of the PBX system.

FRLs and Network Class of Service (NCOS)

FRL and NCOS are features that provide for multiple levels of restriction for users of the AAR and ARS features. FRLs and NCOS allow a certain call type to a specific station user and deny the call to another user. For example, some station users may be allowed to place calls only to private network nodes and not off-net long distance toll calls. A call type that is highly restrictive is international direct distance dialing. FRLs and NCOS are transparent to the station user and are assigned and programmed by the system administrator. Each system facility, station and trunk, is assigned an FRL. Whether or not a call is allowed is based on the originating caller’s FRL and the availability of idle trunk circuits within an assigned trunk group required to route the call. If the station user’s FRL is equal to or greater than the trunk group FRL, calls may be routed using a trunk circuit in the trunk group. If the station user’s FRL is less than the trunk group FRL, the call is denied. Most intermediate/large PBX systems support at least eight FRLs (0 to 7).

The NCOS level of the originating station user determines which call routes can be used to complete the call for a specific routing pattern while the call route is being established across the network. NCOS is also known as traveling class mark (TCM) because the assigned restriction level of the originating station user is embedded within the voice communications signal as the call is routed across the network. If trunk facilities at one tandem switch node are busy and an alternate route (more expensive) is available, the NCOS/TCM level determines whether or not the station user is allowed access to a different call route. A call can be blocked or denied anywhere in the network. Another private network feature, advanced routing (look ahead routing) can be used to avoid possible blocked calls at tandem switch nodes.

Automatic Circuit Assurance (ACA)

ACA is a feature that helps customers identify trunk errors and problems, particularly for private tie trunk circuits. The PBX system maintains a record of individual trunk performance relative to short and long holding-time calls. Holding time is the period from trunk access to trunk release. A system administrator defines short and long holding-time limits. When a possible trunk circuit failure is detected, the system automatically initiates a referral call to an attendant, station user with a display-based voice terminal, or system manager with a CRT monitor. Based on system measurements of holding times per call, referral calls may be placed to attendant, station, or system manager positions. The display information identifies the call as an ACA call, identifies the trunk group access code and the trunk group member number, and shows the reason for the referral (short or long holding time). The ACA feature provides better telecommunications service through the early detection of faulty trunk circuits and possibly reduces out-of-service time.

Virtual Private Networks (VPNs)

A VPN is based on a custom switched telecommunications service tariffed by an interexchange carrier, which permits a customer to establish a communications path between two stations using a UDP. PBX systems are linked to the carrier’s CO facilities using private tie trunks or through dial-up facilities using special access codes. The network facili- ties are shared as part of the PSTN. The key benefit of a VPN is a significant reduction in private line services between networked PBX systems. The first voice communications VPN service was AT&T’s SDN. SDN was designed to expand the scope of private network solutions to customers who could not justify dedicated private line services in support of their private networking requirements. Other similar services were soon available from AT&T’s competitors. VPN telecommunications services, from exchange carriers such as AT&T, MCI Worldcom, and Sprint, simplify private networking applications for business customers because:

  1. The backbone private network carrier facilities are maintained and managed by the service provider.

  2. The UDP and AAR/ARS databases are centralized in the service provider service control points.

  3. Less training and specialized communications equipment is required to design, implement, operate, and maintain a private network.

  4. Call accounting records and billing information are provided by the service provider for all on-net and off-net calls.

Most customers with private network requirements use a combination of traditional ETN tie trunk links (analog, mostly digital) and virtual network services for their on-net and off-net calling requirements. Private networks using VPN services can enjoy the following benefits:

  • Flexible system and station user port capacity

  • Integrated voice/data communications

  • System compatibility across different PBX platforms

  • Cost effective usage- and distance-sensitive pricing

  • Porting of private network numbering plans

  • National/international service transparency

  • PSTN reliability and QoS standards

  • Customized report options

  • Centralized network management capabilities

  • Secure access via screening provisions and enforced use of authorization codes

There are several options available for PBX access to the carrier network, including local exchange service access and special services access. Local exchange access is usually reserved for very low-traffic volume locations that cannot justify a special access service. Local exchange access is provided through a switched connection from a local exchange carrier’s equal access end office. A customer using this access method may presubscribe to the VPN carrier code, or individual station users can dial the carrier code.

Special access arrangements provide direct access to the VPN provider network facilities:

  • PSTN analog trunk circuit facilities

  • PSTN digital trunk circuit facilities

  • Customer-provided access (local bypass)

Digital trunk and customer-provided access may support multiple exchange carrier services, not only VPN service.

The VPN service functions similarly to private line ETNs for on- and off-net calls. Basic call processing features include:

  • Seven-digit dialing (national and international calls)

  • Advanced numbering plan (flexible multidigit numbering plans)

  • Private network interface to ETNs

  • Flexible routing when conditions warrant

  • Network intercept announcements for call completion procedures

  • Network remote access from off-net locations

VPN call management features include:

  • Authorization codes

  • Originating call screening—Grouping of callers with same calling privileges

  • Feature screening—Specifies calling privileges for each screened caller group

  • Access line grouping call screening—Call restriction by specific off-net telephone numbers

  • Partitioned database management—Partitioning of VPN locations into multiple autonomous network groups with direct distance dialing and private networks

Exchange carrier VPNs originated as a voice-focused service. During the past few years VPN services have evolved to support voice and data communications networking applications. Most of today’s VPN offerings are focused on packet switched services to support customer WAN data communications requirements, but can be used for voice networking needs. As VPN has evolved, PBX networking has evolved with the emergence of ToIP technology and the trend toward VoIP trunking.

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