Basic Centrex Features

Call Transfer transfers calls to another line either inside or outside your Centrex system.

Direct Inward and Outward Dialing allows you to control whether incoming or outgoing calls are routed directly to an employee’s desk or to an attendant.

Hunting automatically sends an incoming call from a busy line to the next designated line.

Line Restriction limits phone access on selected lines, so that only authorized numbers or regions can be called.

Station Line Identification provides a record of calls made by each extension.

Station-to-Station Dialing allows station users to intercom between stations by using abbreviated dialing, usually four digits.

Three-Way Calling allows you to add a third person to a regular two-way call.

Optional Centrex Features(Available at an additional cost per line)

Call Forwarding-Busy Line automatically routes calls to another extension or to your voice mail if your line is busy.

Call Forwarding-Don’t Answer automatically reroutes calls to another phone if you haven’t responded within a preset number of rings.

Call Forwarding-Variable sends your calls immediately to another number and is set by the user by entering a three digit code and the destination number.

Call Hold puts a call on hold and returns dial tone.

Call Park lets you hold a call on one Centrex line and pick up a call on another.

Call Pickup lets you use your extension to answer any ringing phone in your designated group.

Call Return lets you return your last incoming call with a code or a single button.

Call Screen allows up to 10 predetermined numbers to be routed directly to a prerecorded announcement.

Call Waiting-Incoming signals you with a tone when another call is coming through on your line.

Caller ID displays the caller’s phone number (on compatible equipment).

Directed Call Pickup use your extension to answer a specific ringing phone in your designated group.

Direct Inward Dial to Direct Outward Dial Transfer allows you to transfer an incoming direct-dialed call to a location, either within, or outside the Centrex system. The caller doesn’t have to hang up and dial the new number.

Distinctive Ringing lets you know if a call is originating inside your office. A single ring means an inside call, a double ring means the call came from outside.

Executive Busy Override permits an interruption on a station that has a call in progress. Parties engaged in conversation hear a warning tone before new caller joins the conversation.

Flexible Route Selection automatically routes calls over designated carriers. You can cut costs if your business does a high volume of statewide or nationwide calling by using your preselected carrier.

Meet Me Conference allows you to meet with a maximum of either 6 or 30 people on a conference call.

Message Waiting Indicator alerts you with a stutter dial tone when someone has left a message for you in your voicemail box.

Music On Hold provides music or an announcement to callers on hold.

Paging provides access to customer-owned paging systems.

Priority Ringing special ringing indicates when any of up to ten numbers you select are calling. If you also subscribe to Call Waiting, a unique call waiting tone alerts you that the caller is one you’ve selected on your Priority Ringing list.

Remote Access to Call Forwarding allows you to control remotely the destination of your forwarded calls.

With Repeat Dialing you can program your phone to keep dialing when your call can’t go through. A special ring lets you know when the line is free.

Sectional Billing permits you to group individual Centrex lines by department, agency or work group.

Select Call Acceptance only callers whose numbers are on your selected list will be allowed to complete the call. Automatically accepts calls from up to 31 telephone numbers you choose.

Select Call Forwarding forwards calls from up to ten numbers you select to any number you designate. Important calls can reach you directly, rather than forwarding to voice mail. Calls not on your special list can be forwarded to yet another number by using Call Forwarding/Variable, Call Forwarding/Busy Line, or Call Forwarding/Don’t Answer.

Speed Calling (also called abbreviated dialing) lets you press fewer digits to make a call.

IP Telephone Design Basics

All manufacturers base their IP telephone on proprietary design schematics and circuitry, but there are common design elements across the unique terminals. IP telephone basics include:

• User interface

• Voice interface

• Network interface

• Processor complex and associated logic

The accompanying diagram illustrates the internal design elements of an IP telephone instrument (Figure 1)

Figure 1: IP telephone design.

The user interface elements provide four classic telephone user function interfaces: keypad for dialing numbers; a variety of keys for line and feature access; a display for user prompts, caller feedback, messages, and other call processing information; serial interface to allow communications to an external device, such as a PDA, to allow synchronization of telephone information; speed dialing; and customer programming. An audible indicator (ringer) is also included to announce incoming calls.

The voice interface converts input analog voice signals into 8-bit digital word bit samples. Speech signals are sampled at an 8-KHz rate to create a 64-Kbps digital bit stream to the processor by using a standard PCM codec. Voice signal compression and IP encoding functions are performed by processor complex elements. The processor complex performs voice processing, call processing, protocol processing, and network management software functions. The complex consists of a DSP for voice-related functions and a MCU for the remaining control and management functions. The DSP and MCU each have associated memory. DSP memory usually includes RAM and ROM elements; MCU memory usually includes RAM and Flash elements. The Flash memory element supports software upgrades.

The network interface allows the transmission and reception of voice packets to and from the telephone terminal based on 10BaseT or 10/100BaseT Ethernet running TCP/IP protocols. Some IP telephones may be equipped with multiple RJ-45 Ethernet connector ports and an integrated Ethernet hub/switch to support connections to the customer premises LAN and desktop PC clients. Newer IP telephones also may be designed with a USB connector port.

Basic IP telephone software modules include a variety of user interface drivers (display, keypad, ringer, user procedures), voice processing modules, telephony signaling gateway modules, network management modules, and system service modules. The voice processing software modules include a PCM interface unit; a tone generator (call progress tones, in-band DTMF signaling digits); a line echo canceler unit (ITU G.168-compliant echo cancellation on sampled, full-duplex voice port signals); an acoustic echo canceler for terminals equipped with a speakerphone; VAD; voice codec unit (compression and packeting of the 64-Kbps digital stream received from the station user based on a variety of algorithms, such as G.711, G.723.1, G.729/A, etc.); packet playout unit (compensation for network delay, jitter, and packet loss); packet protocol encapsulation unit (based on RTP, which runs directly on top of the UDP); voice encryption (to ensure privacy); and a control unit (coordinates the exchange of monitor and control information between the voice processing module and the telephony signaling and network management modules).

The telephony signaling gateway subsystem in an IP telephone performs the basic functions for call setup and teardown procedures. Software modules used by this subsystem include call processing, address translation and parsing, and network signaling. The most widely implemented network signaling standard used by currently available IP-PBX systems is H.323 protocol. H.323 is an ITU standard that defines several signaling and protocol specifications for multimedia communications between LAN-based terminals and network equipment. The main H.323 standards used for VoIP in an IP telephone are H.225–Call Signaling Protocol (based on Q.931), H.245–Control Protocol; RAS Protocol; and RTCP. An emerging network signaling standard not currently used by any commercially available IP-PBX, but being planned for by most suppliers, is SIP. SIP is the protocol developed and promoted by the Internet Engineering Task Force (IETF) and is forecasted to be widely implemented in network hosted services, such as IP-Centrex, and may eventually replace H.323 as the primary signaling protocol used by premises communications systems.

Basic Call Processing Functions

The primary call processing responsibilities of the Main System Processor are provisioning of dial tone, digit reception and analysis, number analysis, TDM bus talk slot assignments and switch connections for intercom and trunk calls, routing analysis, feature provisioning, and call monitoring.

There are several fundamental main processor management functions used to process calls:

1. Call sequencing control: management of the call sequence logic that takes a call from one state to another

2. Resource management: management of various system resources, such as DTMF receivers; time/talk slots for call connections; tone generators; and internal software records for call processing (including the system dial plan), messaging, measurements, and call detail records

3. Terminal handling: management of different desktop terminal models, including support of line appearances, feature buttons, display fields, adapter modules, and other functional components

4. Routing and termination selection: controls the selection of the terminating endpoint (station, trunk) of the call, including functions such as hunting, bridging, call coverage, and least cost routing.

Call processing is a series of events that result in the completion of a call. Figure 1 illustrates the call dialing and connection process. The process begins when a port (station or trunk) changes from an idle to an active state. Port seizure occurs when a station goes off-hook, a trunk port circuit receives an incoming call signal from the Central Office or network, or an attendant begins dialing. When a station port has been seized, the main processor seizes a register storage record, instructs a tone sender unit to send dial tone to the caller, and instructs the switch network to establish a connection to the port. When a CO or DID trunk port has been seized, the main processor seizes a register storage record, assigns a tone receiver unit register to the port, and instructs the trunk circuit port to signal that the main processor is ready to receive digits.

Figure 1: Call dialing and connection process.

The second step in the call process is digit reception and analysis. Digits are sent by system users to dial another station or activate a specific feature. A call register stores dialed digits or received digits over a trunk circuit. Digits can be received one by one (manual dialing) or in a group if a dialed number has been preprogrammed—speed dial. For station initiated calls, the first digit received by the register suppresses the dial tone. There is typically a time-out period between going off-hook and dialing the first digit or between the first and second dialed digits; if no digits are dialed within the programmed period, an intercept tone is sent to the caller. In addition to seizing a register, a tone sender and receiver are simultaneously seized, a switch connection is made, and the Class of Service of the caller is checked. A no-progress tone is sent to the caller if a tone receiver cannot be seized.

The third step in the call process is number analysis. Number analysis allows the PBX system to identify the number dialed and properly route the call. Internal system number analysis is performed for all calls, both intercom and trunk calls, within the PBX system. Special number analysis processes are performed for DID trunk calls and dialed feature access codes using * and # keys. For incoming DID trunk calls, the main processor analyzes the digits to determine whether the number is an attendant console position. If the number is an attendant console position, the call is processed and routed to the attendant. If the number is not an attendant console position, the internal number analysis program analyzes digits.

The internal number analysis program determines the correct call processing procedure to be implemented. There are many types of dialed or received numbers that can be analyzed for call routing purposes: station number, including hunt group number; individual, group, or emergency attendant number; abbreviated dialing number (speed dial); paging number; automatic route selection access code; Direct Inward System Access (DISA) code; data station number; modem pool group access code; and external destination codes, such as 911. If a number is not defined in the number analysis program, it is treated as a vacant number, and the appropriate intercept treatment is applied.

Another important main processor function is provisioning of call progress tones and indications. Call progress tones and indications are single- and dual-frequency combinations applied in a variety of cadence patterns, such as continuous tone sending, one repeated sequences (tone, pause, tone pause…), two repeated sequence (tone 1, pause 1, tone 2, pause 2…), or three repeated sequences (tone 1, pause 1, tone 2, pause 2, tone 3, pause 3…).

Call progress tone types include:

• Busy tone

• Call diversion indication (call forward indication)

• Call waiting indication

• Conference tone

• Confirmation tone

• Dial tone

• Expensive route warning tone

• Intercept tone

• Intrusion tone

• Message waiting tone

• No-progress tone (congestion tone)

• Off-hook queue tone

• Recall dial tone

• Ringback tone

• Special dial tone (do not disturb tone)

• Special message waiting tone (message waiting)