Computer Telephony Integration (CTI) | Features/Function Enhancements

CTI had its origins in the early 1980s when AT&T was the first PBX manufacturer to use an adjunct applications processor to provide a message center and directory system function behind its Dimension PBX. Unanswered calls were forwarded to a message center agent who could do a directory lookup for the person being called and enter a message into a computer terminal. The message was then forwarded to a printer close to the called party. The connection between the PBX and the applications processor was a physical link, and no enhanced PBX voice functions were provided. This was CTI in its most rudimentary form.

The concept of enhancing the PBX system with an adjunct applications processor, with physical and logical connectivities between the two independent systems, was not realized until the late 1980s. The first true CTI options were available almost concurrently on two PBX systems. The NEC NEAX2400 and the Intecom IBX offered an Open Applications Interface (OAI) option for third-party application software developers to design applications running on an adjunct client or server to enhance the performance capabilities of the core PBX system. Both suppliers’ OAI option included an intelligent signaling link between the PBX and the adjunct applications processing system. The intelligent signaling link provided a communications path for the PBX system to send system status and message packets to the applications processor and provided the means to transmit the software application program commands back to the PBX. The applications programming interface included with the OAI software developer’s toolkit was proprietary to each system and required software developers to write different programs for different systems. The PBX manufacturers initiated the first CTI implementations, and customer demand for the NEC and Intecom offerings was limited.

When the first CTI industrywide standards were being developed in the late 1980s and early 1990s, it was the major computer companies, such as DEC and IBM, that took the initiative, not the PBX manufacturers. Most of the early CTI implementations were host-based applications, such as predictive dialing and agent screen pops for outbound calling centers. The PBX manufacturers focused on first-party desktop CTI applications by providing a physical/logical link between their digital telephones and desktop PC clients. The client software applications supported a variety of PC telephony features and functions, including directories, screen pops, PBX feature/function activation, on-screen dialing, and call logs/notes. Client/server CTI applications were initially driven by Novell, which promoted its Telephony Services Application Programming Interface (TSAPI) standard. The CTI standard in Europe was developed by the European Computer Manufacturers Association (ECMA) and was known as Computer Services Telephony Applications (CSTA). Microsoft developed its own desktop CTI standard, Telephony Applications Programming Interface (TAPI), that was later enhanced to support client/server applications, and supplanted Novell’s TSAPI as the most popular CTI platform (Figure 1).

Figure 1: CTI system designs

CTI is used behind a PBX system to provide features and functions not available in the generic communications software package. Many of the CTI applications have limited market potential for which PBX developers cannot afford to expend resources. Desktop and client/server CTI was envisioned by many within the industry has a means to replace the proprietary telephone instrument. Instead of buying an expensive voice terminal with limited feature button and display capabilities, the PC client and monitor would serve as the station user interface to the PBX system for all system operations, including dialing, call screening, call answering, and feature implementation. Despite substantial marketing efforts, customers resisted using CTI PC telephony as a substitute for high-performance telephone instruments. Station users were reluctant to learn new programming tools, the ergonomics were poor, the cost was too high, and the PC reliability factor was unacceptable. Desktop CTI shipments behind PBX systems have been negligible, with annual shipment levels less than 2 percent of total station shipments. The only PBX market segment in which CTI gained a foothold was call centers because the evolving call center process became heavily dependent on computer technology, and ACD agents were already using PC clients to handle the typical caller transaction. About 25 percent of call centers are currently implementing CTI.

In the late 1990s several recent PBX market entrants attempted to design and market enterprise communications systems based on CTI client/server architecture principles: an applications processor served as the call processing manager, and station user desktops were analog telephones logically working with PC telephony clients. The desktop PC telephony applications software was included in the system price, and CTI links were used to provide third-party applications not included in the generic software package. These systems did not support multiple-line appearance digital telephones and found limited market appeal. One manufacturer who first attempted to market a pure client/server design, Vertical Networks, recognized the value of digital, multiline, display telephones, and downplayed the CTI attributes of its system when it belatedly marketed a traditional PBX-like digital telephone.

The new IP softphones, available from most longtime and new PBX suppliers, is a proprietary version of desktop CTI, although it is not marketed as such. The PBX system’s call processing manager, be it a traditional proprietary common control complex or a third-party server, functions as a server to the PC client desktop. It is forecasted that IP softphones will become more popular throughout the remainder of this decade, although desktop telephone instruments will remain the dominant voice terminal type.


ACD-based Call Centers | Features/Function Enhancements

The fundamental function of a call center is to direct calls to a group of answering positions, equitably distribute the calls among the group members, and minimize the caller’s time in queue waiting for an agent. ACD is the general term used to describe this telephony function and was introduced as a PBX feature in the early 1980s. The early PBX-based ACD software options had limited flexibility in screening, routing, and queuing calls, and the MIS reporting function was minimal. ACD was developed as an enhanced version of the Uniform Call Distribution (UCD) feature, which itself was an enhanced version of Hunting.

ACD systems in the early 1980s were used only in very formal incoming call center environments, and annual shipments were limited to several hundred systems. In the early 1980s the only PBX system with an ACD capability that was competitive with stand-alone ACD systems was the Rockwell Wescom 580. Rockwell was also the leading manufacturer of stand-alone ACD systems at the time; its Galaxy ACD was the leading system in the market in terms of features and functions. The Rolm CBX had a basic ACD package, but most PBXs could offer little more than UCD for distributing calls. By the mid-1980s the market for ACD systems was growing almost 50 percent annually, and many PBX manufacturers looked at the call center application market for potential high profit margins. The first PBX manufacturer to offer a sophisticated ACD option that could compete with the Rockwell Galaxy ACD was AT&T. When AT&T introduced its latest version of System 85 in 1987, the ACD call screening, routing, and queuing functions were based on a new customer programming tool called Call Vectoring. Call Vectoring consisted of a series of programming commands that defined call coverage and treatment operations for each incoming call with flow charting methods. A new advanced MIS reporting system, based on an adjunct computer working behind the System 85, offered customers dozens of reports for call center management and monitoring. Supervisor positions, linked to the MIS reporting processing system consisted of GUI workstations displaying real-time call center information. The AT&T announcement marked a major change in the direction of PBX-based call centers.

After the AT&T announcement, other PBX manufacturers began development of similar ACD and MIS reporting options for their communications systems. By the mid-1990s the typical PBX/ACD offering was based on an adjunct application server required to run sophisticated call routing and treatment software, provide advanced MIS reports, and support supervisor PC client positions. Customers with complex call center application requirements would likely link the PBX/ACD system to a CTI application server to provide features such as screen pops, an coordinated voice/data screen transfer. An IVR would likely serve as a front-end system to screen calls and pass call-prompted data to the PBX/ACD program for analysis. New ACD features that improved call center efficiency and customer satisfaction levels included skills-based routing and agent skill profiling and mapping (Figure 2-4).

By 2000 PBX/ACDs dominated the call center market, accounting for more than 80 percent of total agent shipments. During the next few years the traditional voice-centric call center will migrate toward a mixed-media model that integrates the traditional ACD system with email distribution systems and Internet-based Web sites. The emerging IP-PBX system will be able to support the new-generation mixed-media agent position, which will handle incoming and outgoing voice calls, respond to e-mails, and chat with Web surfers on-line. Despite the changes in technology and the mix of voice calls, e-mail, and on-line chat, the fundamental functions of the new contact center system will be the same as those of the original ACD systems: equitable distribution of calls to agents and minimized response times for customer inquiries.


System Features

Account Codes

This feature allows a station user to input a multiple digit code for certain types of outgoing trunk calls. Each code is associated with a unique file record, usually used for account billing purposes.

Answer Detection

This feature detects the state of outgoing trunk calls that do not receive network answer supervision to improve the accuracy of the call duration field in CDRs.

Authorization Codes

This feature allows a station user to input a personal identification code as a means for extending the control of system users’ calling privileges and security for remote access callers. Authorization codes may be used for any or all of the following reasons: allow a calling user to override the FRL assigned to the originating station or trunk, restrict individual incoming tie trunks and remote access trunks from accessing an outgoing trunk, identify certain calls on CDRs for cost-allocation purposes, and provide additional security control for the system.

Automated Attendant

This feature allows the system to answer incoming trunk calls with no intervention of an attendant position. The system will provide the caller with a message or dial tone and allow the caller to directly dial an internal extension number.

Automatic Alternate Routing

This feature provides alternative routing choices for private on-network calls. When implemented, the system automatically selects the most desirable (normally the least expensive) route among multiple trunking facilities for private network calls. AAR also provides digit modification to allow on-network calls to route through the public network when an on-network route is not available.

Automatic Call Distribution

This feature provides the automatic connection of incoming calls to specific splits (hunt groups) of station users (agents). Calls to a specific split are automatically distributed among the agents assigned to that split. If agents are not available, the call can queue to the split to wait for an agent to become available.

Automatic Camp-on

When a DID call has been terminated at a busy station, the call is “camped-on” to the called station. When busy station becomes idle, it is automatically connected to the camped-on incoming trunk call.

Automatic Circuit Assurance

This feature assists users in identifying possible trunk malfunctions. The system maintains a record of the performance of individual trunks relative to short and long holding-time calls. The system automatically initiates a referral call to an attendant or display-equipped voice terminal user when a possible failure is detected.

Automatic Number Identification

This feature allows the system to receive an incoming caller’s local telephone company trunk billing number and display the number on a station user’s voice terminal with a display. The ANI is transmitted over an incoming digital trunk circuit with the use of in-band or out-of-band signaling techniques.

Automatic Recall

This feature alerts a voice terminal after a fixed interval that a call it transferred has been placed on hold, is camped-on, or continues ringing with no answer.

Automatic Route Selection

This feature routes calls over the public network based on the preferred (normally the least expensive) route available at the time the call is placed. ARS provides a choice of routes for any given public network call. The following types of trunk groups can be accessed by ARS: local CO, FX, WATS, tie trunk, T1/E1, ISDN PRI, and IP WAN. The system selects the most preferred (normally least expensive) route for the call. Interexchange carrier code dialing is not required on routes selected by the system. Interexchange carrier codes are assigned in translations to best benefit the customer on any given call. These codes are inserted as needed to guarantee automatic carrier selection.

Automatic Transmission Measurement System

This feature provides for trunk facilities to be measured for satisfactory transmission performance. The performance of the trunks are evaluated according to measurements produced by a series of analog tests and are compared against user-defined threshold values.

Call Coverage (Multiple Call Forwarding, Split Call Forwarding)

This feature provides automatic redirection of calls that meet specified criteria to alternate answering positions in a call coverage path. Lead coverage paths can be administered to apply to all calls all the time, internal or external calls, or to apply to a specific day of the week or a specific time of the day. Different coverage paths are administered based on incoming call origination, type, or time.

Call-by-Call Service Selection

This feature allows a single ISDN-PRI trunk group to carry calls to many services or facilities or to carry calls using different interexchange carriers. The feature typically uses the same routing tables and routing preferences that are used by AAR and ARS. The service or facility used on an outgoing CBCSS call is determined by information assigned in the AAR/ARS routing patterns. Without CBCSS each trunk group must be dedicated to a specific service or facility. CBCSS eliminates this requirement by allowing a variety of services to use a single trunk group. These services are specified on a call-by-call basis.

Call Detail Recording

This feature records detailed call information on all incoming and outgoing calls on specified trunk groups and extensions administered for intraswitch recording and sends this information to a CDR output device. The CDR output device provides a detailed printout that can be used by the system administrator to compute call costs, allocate charges, analyze calling patterns, detect unauthorized calls, and keep track of unnecessary calls.

Call Log

This feature stores dialed station numbers and incoming identification numbers (internal CLID, CLASS CLID, ANI) at a multiple line voice terminal with a display. The numbers that are stored are those of the most recently dialed and incoming calls. There is a limited amount of stored and displayed numbers that varies by system. Pressing a call log button brings up the display. Calls to numbers appearing in the call log display field can be dialed automatically through menu control keys.

Centralized Attendant Service

This feature allows services performed by attendants in a private network of switching systems to be concentrated at a central, or main, location. Although all incoming calls to the network are routed to a main PBX system, each branch in the centralized attendant service configuration has its own listed directory number or other type of access from the public network. Incoming trunk calls to the branch and attendant-seeking voice terminal calls are routed to the centralized attendants over a release link trunk. The centralized attendants are located at the main location.

Class of Restriction

This feature defines different classes of call origination and termination privileges. Systems may have one restriction class, one with no restrictions, or as many restriction classes as necessary to effect the desired restrictions.

Class of Service

This feature determines whether or not voice terminal users can access any or all station or system features and functions. There are many COS levels that can be programmed by the system administrator; each level is associated with a defined feature/function set that may contain one or many features and functions. Each COS level allows or denies access to the defined feature/function set. Every system user is assigned a COS level by the system administrator.

Controlled Private Calls

This feature allows the system operator to charge station users for personal outgoing calls. The following items define station users: extension number (virtual or real), personal identification number, and call restriction table for private calls. The user can make a private call according to the following rules: only from his own set, from every authorized set in the subnetwork, or from only a few selected sets.

Delayed Ringing

This feature allows trunks and station lines to ring immediately at the dialed destination station and, after a programmed interval, at a secondary station that shares the same line appearance as the original destination station.

Dial Plan

The dial plan is the system’s guide to digit translation. When a digit is dialed, the system must know what to expect based on that digit. The dial plan, or first-digit and second-digit tables, established during administration for each system provides information to the switch on what to do with dialed digits. The tables define the intended use of a code beginning with a specific first digit or specific pair of digits. These digits tell the system how many digits to collect before processing the full digit string.

Dialed Number Identification Service

This feature provides a display of the listed directory number of an incoming trunk call to the attendant position. The display can be the actual digits of the number, or an alphanumeric name or identifier. This screening feature allows an attendant to better handle the incoming call and provides a higher level of customer service.

Direct Department Calling

This feature allows direct inward access to an answering group other than the attendant even if the system does not use the DID feature. A direct department answering group can consist of voice terminals and individual attendants. One extension number is assigned to all voice terminals and individual attendants. Incoming calls to a direct department group can be internal or external. With this feature, an incoming call rings the first available voice terminal or individual attendant in the administered sequence. If the first group member in the sequence is active on a call (busy) or has had calls temporarily redirected, the call routes to the next group member, and so on. Incoming calls always try to complete at the first group member in the administered sequence. Calls are not evenly distributed among the group members.

Direct Inward Dialing

DID connects calls from the public network directly to a dialed extension number without attendant assistance. Specialized DID trunk circuits are required to implement this feature. DID reduces attendant workload and facilitates connections between an external calling and an internal called party.

DID Call Waiting

This feature allows an incoming call on a DID trunk circuit to be automatically camped-on to the destination station if the destination station is busy.

Direct Inward System Access

DISA allows system users who are off-premise to dial into the system, input a special access code, and use the system facilities even though the caller is not using an internal voice terminal. It allows access to the system’s optimally priced trunk network facilities and other cost savings features.

Direct Inward Termination

This feature automatically routes incoming network exchange calls to a preselected station without attendant assistance. The called party can process the call in a manner similar to any normal trunk call.

Direct Outward Dialing

This feature allows voice terminal users to access the public network without attendant assistance. Station users dial a defined trunk access code (such as 9) for public network connection and dialing.

Extended Trunk Access

This software feature provides a means for routing calls that are not defined in the first- or second-digit tables or the feature/trunk access code tables. This feature makes use of an extended trunk access routing pattern or node number for determining how to route an unidentified call.

Facility Restriction Levels

FRLs provide multiple levels of restriction for users of the AAR or ARS features. FRLs provide a method of allowing certain calls to specific users and denying the same calls to other users. For example, certain users may be allowed to use CO trunks to other corporate locations, whereas other users may be restricted to the less expensive private network lines. The FRLs are defined and programmed into the system by the administrator and are transparent to the station user. Regular dialing procedures are unaffected.

Facility Test Calls

This feature allows a voice terminal user to make test calls to access specific trunks, DTMF receivers, time slots, and system tones. The test call makes sure the facility is operating properly. The feature is implemented by dialing an access code.

Forced Account Code

This feature forces a station user to enter an account code for all outgoing trunk calls. The account code must be entered before dialing the outgoing number. Calls are processed only after the account code is entered and verified. Some systems allow calls to be classified into multiple groups, such as: call with a controlled project number, call with an uncontrolled project number, and call without a project number. The choice of call depends on a data system configuration based on two parameters: with/without a project number or with/without a controlled project number.


This feature allows users whose stations are translated to their own preferences and permissions to associate those preferences and permissions with any compatible terminal. These include the definitions of terminal buttons, abbreviated dial lists, and COS and restriction class permissions assigned to the user’s station.

House Phone

This feature allows station users to use selected voice terminals to reach an attendant by simply going off-hook.


This feature routes calls to a station within a predefined ordered group after checking for station idle or busy status. Calls are routed to another group when all stations are busy. Hunting is accomplished through the ACD, direct department calling, and UCD features. The order of hunting is defined under each feature. Under direct department calling, call distribution is not uniform across hunt group members.

Integrated Directory

This feature allows internal system users with display-equipped terminals to access the system database, use the touch-tone buttons to key in a name, and retrieve an extension number from the system directory. The directory contains an alphanumeric listing of the names and extension numbers assigned to all voice terminals administered in the system.

Modem Pooling

This feature allows switched connections between digital data endpoints (data modules) and analog data endpoints and data modems. The analog data endpoint can be a trunk or a line circuit.

Multiple Listed Directory Numbers

This feature allows a publicly published number for each incoming and two-way (incoming side) FX and local CO trunk group assigned to the system. This feature also allows DID numbers to be treated as listed directory numbers.

Music on Hold

This feature plays music for a caller who is on hold, waiting in a queue, or on a trunk call in the process of being transferred. The feature provides a means to let callers know they are still connected to the system.

Night Service

This feature directs all calls for the primary and daytime attendant consoles to a night console. It is typically activated when an attendant presses the night button on the principal attendant console and deactivated by pressing the night button again. Night service also can be activated and deactivated from one station in the system by use of a night service button assigned to that station.

Off-Hook Alarm

This feature allows a station user to call an attendant or any preselected programmed station by simply staying off-hook for a preprogrammed period. The calling number is automatically displayed at the attendant console or the preselected station.

Off-Premises Station

The feature allows a voice terminal outside the switching system location to be connected to the system via specialized CO trunk circuits. The voice terminal must be analog and must be registered with the Federal Communications Commission.

Open System Speed Dial

A station user can select a system speed number by dialing a system speed code or name and then dialing the relevant final sequence of numbers to select the external party.

Power Failure Transfer

This feature provides service to and from the local telephony company to a designated station during power failures affecting the PBX system. During PFT mode, no other system features can be activated.

Property Management System Interface

PMS interface provides a communications link between the system and a customer-owned PMS. The PMS allows a customer to control certain features used in hospital-type and hotel/motel-type environments. The communications link allows the PMS to interrogate the system and allows information to be passed between the system and the PMS.

Recent Change History

This feature allows system administrators to view or print a history report of the most recent administration and maintenance changes. The history report also lists each time a user logs in or off the system. This report may be used for diagnostic, information, or security purposes.

Restricted Incoming Station

When a station is configured to receive only incoming calls, the station user receives a busy signal as soon as the handset is picked up.

Restriction—Controlled, Inward/Outward, Toll/Code, Trunk, Voice Terminal

A series of features that allows an administrator or attendant to activate or deactivate defined trunk access and I/O calling privileges for a station or group of stations.

Route Advance

This feature automatically routes outgoing trunk calls over alternate facilities when the first choice trunk group is busy. This feature is implemented only if a station user selects the first choice trunk group with a dial access code. The system advances through a series of alternate trunk groups only if the first-choice trunk group is busy.

Shared Tenant Service

A system manager can partition the PBX to provide telecommunications services to multiple tenant groups. The tenant groups can have independent dial plans, CDR, ARS and call routing tables, attendant groups, and COS/class restriction levels. Each group is logically partitioned from the others for all premises telecommunications services. The number of partitioned tenants depends on the system.

System Speed Dial

A station user can call a number by dialing system speed codes or names. The list of system speed codes can be common to all system users or split into different lists. With splits, the users can access different lists according to their COSs. Each outside system speed code corresponds to the access feature code of a trunk group or external trunk (public or private).

Timed Reminder

This feature allows the system to be programmed to automatically call stations at specified times. When the called party answers, the station is connected to a recorded announcement or music source.

Trunk Answer Any Station

This feature allows any station to answer an incoming call trunk when the system is in night service mode. A common alert signal is sent to all stations, and any station can answer the call. The answering station can extend the call to any other station by using call transfer.

Trunk Callback Queuing

This feature places outgoing calls in an ordered queue (first in, first out) when all trunks are busy. The voice terminal user is automatically called back when a trunk becomes available. The voice terminal receives a distinctive three-burst alerting signal when called back.

Uniform Call Distribution

This feature allows direct inward access to an answering group other than the attendant. A UCD answering group can consist of voice terminals and individual attendants. One extension number is assigned to all voice terminals and individual attendants. Incoming calls to a UCD group can be internal or external. With UCD, an incoming call rings the member of the group that has not received a UCD group call for the longest period (the most idle member). Incoming calls to a UCD group extension number are distributed evenly across the group members.

Uniform Dial Plan

A UDP may be established during administration as part of the dial plan. This plan provides a common extension number plan that can be shared across a group of switches. If a UDP is to be established, all extension numbers (in the UDP numbering plan) must be the same length.

Virtual Extensions

This feature permits the assignment of circuits that do not physically exist, to be used for secondary extensions on multiple line voice terminals.

Voice Message System Interface

This feature provides a signaling interface between a PBX and an external VMS. The interface allows the VMS to activate message waiting indicators on PBX voice terminals.

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