U.S. patent application number 14/028274 was filed with the patent office on 2014-01-30 for telecommunications system and method.
Invention is credited to Harold E.A. Hansen, II, Eric G. Suder.
Application Number | 20140029736 14/028274 |
Document ID | / |
Family ID | 38227161 |
Filed Date | 2014-01-30 |
United States Patent
Application |
20140029736 |
Kind Code |
A1 |
Hansen, II; Harold E.A. ; et
al. |
January 30, 2014 |
Telecommunications System and Method
Abstract
A telecommunications system includes switching circuitry for
coupling a call to a extension coupled to the system, voice
processing circuitry for automatically interacting with the call, a
microprocessor, a first data bus connected, between the
microprocessor and the switching circuitry, and a second data bus
connected between the microprocessor and the voice processing
circuitry. In the telecommunications system, a method includes
coupling, with the switching circuitry, a call to a
telecommunications extension coupled to the system, and the voice
processing circuitry automatically interacting with the call when
the telecommunications extension does not answer the call.
Inventors: |
Hansen, II; Harold E.A.;
(Plano, TX) ; Suder; Eric G.; (Plano, TX) |
Family ID: |
38227161 |
Appl. No.: |
14/028274 |
Filed: |
September 16, 2013 |
Related U.S. Patent Documents
|
|
|
|
|
|
Application
Number |
Filing Date |
Patent Number |
|
|
13487762 |
Jun 4, 2012 |
8538002 |
|
|
14028274 |
|
|
|
|
12200330 |
Aug 28, 2008 |
8194845 |
|
|
13487762 |
|
|
|
|
09805395 |
Mar 13, 2001 |
7421066 |
|
|
12200330 |
|
|
|
|
08873215 |
Jun 11, 1997 |
6252944 |
|
|
09805395 |
|
|
|
|
60023749 |
Jun 12, 1996 |
|
|
|
Current U.S.
Class: |
379/88.22 |
Current CPC
Class: |
H04M 3/533 20130101 |
Class at
Publication: |
379/88.22 |
International
Class: |
H04M 3/533 20060101
H04M003/533 |
Claims
1. In a telecommunications system comprising a microprocessor, a
first data bus connected between the microprocessor and switching
circuitry, and a second data bus connected between the
microprocessor and voice processing circuitry, a method comprising:
coupling, with the switching circuitry, a call to a
telecommunications extension coupled to the system; and the voice
processing circuitry automatically interacting with the call when
the telecommunications extension does not answer the call.
2. The method as recited in claim 1, wherein the first and second
data busses share at least one data path.
3. The method as recited in claim 1, wherein the switching
circuitry and the voice processing circuitry are controlled by the
microprocessor.
4. The method as recited in claim 1, wherein the microprocessor is
implemented in a single Integrated chip.
5. The method as recited in claim 1, wherein the voice processing
circuitry couples the call to a voice mail box when the
telecommunications extension does not answer the call.
6. The method as recited in claim 1, wherein the voice processing
circuitry automatically interacts with the call when the
telecommunications extension does not go off hook to answer the
call.
7. The method as recited in claim 1, wherein the voice processing
circuitry further comprises a conference bridge operable for
coupling the call to one or more internal or external
telecommunications devices.
8. The method as recited in claim 1, wherein the system further
comprises circuitry operable for recording all or a portion of the
call after the telecommunications extension is connected to the
calf wherein the circuitry operable for recording operates in
response to an activating signal activating tactilely initiated on
the telecommunications extension.
9. The method as recited in claim 1, wherein the voice processing
circuitry automatically interacting with the call further
comprises: playing a message to the call; receiving a signal sent
from the call; and connecting the call to the telephone extension
in response to the signal sent from the call.
10. In a system comprising a microprocessor, a first data bus
connecting the microprocessor to switching circuitry, and a second
data bus connecting the microprocessor to voice processing
circuitry, a method comprising: the switching circuitry coupling a
call to an extension coupled to the system; and the voice
processing circuitry automatically interacting with the call.
11. The method as recited in claim 10, wherein the first and second
data busses share at least one data path.
12. The method as recited in claim 10, wherein the switching
circuitry and the voice processing circuitry are directly
controlled by the microprocessor.
13. The method as recited in claim 10, wherein the microprocessor
is implemented in a single integrated chip.
14. The method as recited in claim 10, wherein the voice processing
circuitry automatically interacting with the call comprises
coupling the call to a voice mail box when the telephone extension
does not answer the call.
15. The method as recited in claim 10, wherein the extension is
operable for establishing a voice channel between the extension and
the call.
16. The method as recited in claim 10, further comprising coupling
the call to one or more internal or external telecommunications
devices.
17. The method as recited In claim 10, further comprising recording
at least a portion of the call after the extension is connected to
the call.
18. The method as recited in claim 17, wherein the recording of at
least a portion of the call is initiated in response to an
activating signal tactically initiated on the telephone
extension.
19. The method as recited in claim 10, wherein the voice processing
circuitry automatically interacting with the call further
comprises: playing a pre-recorded message to the call; receiving a
signal sent from the call; and connecting the call to the extension
in response to the signal sent from the call tones.
20. The method as recited in claim 10, wherein the voice processing
circuitry automatically interacting with the call further comprises
playing a pre-recorded message to the call when the extension does
not answer the call.
Description
CROSS-REFERENCE TO RELATED APPLICATIONS
[0001] This application is a continuation of U.S. application Ser.
No. 13/487,762, which issued as U.S. Pat. No. 8,538,002 on Sep. 17,
2013. which is a continuation of U.S. application Ser. No.
12/200,330, which issued as U.S. Pat. No. 8,194,845, which is a
continuation of U.S. application Ser. No. 09/805,395, which issued
as U.S. Pat. No. 7,421,066, which is a continuation of U.S.
application Ser. No. 08/873,215, which issued as U.S. Pat. No.
6,252,944, and which was reissued as RE39,722, which claims benefit
of priority to U.S. Provisional Application No. 60/023,749, filed
Jun. 12, 1996.
TECHNICAL FIELD
[0002] The present invention relates in general to a
telecommunications system.
BACKGROUND INFORMATION
[0003] Referring to FIGS. 2 and 13, there is illustrated a prior
art technique for combining telephone and voice mail systems. The
dilemma is how to provide communication between the switching
system ("PBX") 200 and the voice mail ("VM") system 201.
Communication with the PBX 200 is typically done through either the
CO lines or on the station (extension) side. Since CO lines are
more of a precious resource than the station connections, the prior
art system shown in FIG. 2 communicates between the voice mail
system 201 and the PBX 200 on the station side using connection
202. Connection 202 may be an analog telephone line or via an EKT
(electronic key telephone) integrated connection. Alternatively, a
proprietary EKT line 204 may be coupled to an analog telephone
adapter 205, which uses analog line 203 to couple to voice mail
system 201.
[0004] Such systems are typically configured by programming the PBX
200 to perform a transfer to an extension that is connected to the
voice mail system 201 upon one or more occurrences, such as when
the outside call received by the PBX 200 intended for a particular
extension receives a busy signal or the extension rings a certain
number of times. At this point in time, the call resides within the
PBX 200 (step 1301). Next, the PBX 200 transfers the call using a
flash-hook and then dialing the extension number (step 1302)
pertaining to the voice mail system 201 in order to transfer the
call to the voice mail system 201. At this point in time, the call
cow resides within the voice mail system 201, which may play a
greeting to the incoming call (step 1303). In response to the
greeting played by the voice mail system 201, the call may send a
signal, which is detected by the voice mail system 201 (step 1304).
Thereafter, die voice mail system may record a message received
from the call (the incoming call resides in the voice mail system
201; step 1306), or the voice mail system 201 may transfer the call
to a desired destination, such as a station extension (the incoming
call is now resident within the telephone system 200; step 1305).
In-band signaling, a serial connection, etc. may he added to
farther improve the system, but it is still configured as two
separate systems--a PBX 200 coupled to a separate voice mail system
201.
[0005] Another prior art system not shown herein is the use of a
personal computer with a voice adapter card inserted therein for
interconnecting to a switching system. Again, the same problems
arise, since there is a separate voice mail system coupled to a
telephone system where software in the personal computer operates
the voice mail portion.
BRIEF DESCRIPTION OF THE DRAWINGS
[0006] FIG. 1 illustrates, in block diagram form, components of
embodiments of the present invention;
[0007] FIG. 2 illustrates a prior art system coupling a switching
system and a voice mail system;
[0008] FIG. 3 illustrates, in block diagram form, components of a
port card implemented within embodiments of the present
invention;
[0009] FIG. 4 illustrates a flow diagram of a process for recording
an incoming call;
[0010] FIG. 5 illustrates a flow diagram of a process for
implementing a beep timer;
[0011] FIG. 6 illustrates a flow diagram for terminating a
recording;
[0012] FIGS. 7A-7D illustrate a flow diagram implementing
interactive help;
[0013] FIG. 8 illustrates a flow diagram for implementing context
sensitive help;
[0014] FIG. 9 illustrates a flow diagram implementing real-time
call screening;
[0015] FIG. 10 illustrates functions implemented within a signal
processing circuit within embodiments of the present invention;
[0016] FIG. 11 illustrates an electronic key telephone
interface;
[0017] FIG. 12 illustrates a loop start CO interface;
[0018] FIG. 13 illustrates a prior an process for call
processing;
[0019] FIG. 14 illustrates an EKT;
[0020] FIG. 15 illustrates a process for implementing an auto
attendant within embodiments of the present invention; and
[0021] FIGS. 16-17 illustrate processes for implementing Quick
Groups.
DETAILED DESCRIPTION
[0022] In the following description, numerous technical details are
set forth such as specific word length and specific hardware
interfaces, etc. to provide a thorough understanding of embodiments
of the present invention. However, it will be obvious to those
skilled in the art that embodiments of the present invention may be
practiced without such specific details. In other instances,
well-known circuits have been shown m block diagram form in order
not to obscure embodiments of the present invention in unnecessary
detail For the most part, details concerning timing considerations
and the like have been omitted inasmuch as such details are not
necessary to obtain a complete understanding of embodiments of the
present invention and are within the skills of persons of ordinary
skill in the relevant art.
[0023] The use of the word "a" or "air" when used in conjunction
with the term "comprising" in the claims and/or the specification
may mean "one," but it is also consistent with the meaning of "one
or more, ""at least one," and "one or more than one." The use of
the term "or" in the claims is used to mean "and/or" unless
explicitly indicated to refer to alternatives only or the
alternatives are mutually exclusive, although the disclosure
supports a definition that refers to only alternatives and
"and/or." Throughout this application, the terms "about" or
"approximately" are used to indicate that a value includes the
inherent variation of error for the device, the method being
employed to determine the value, or the variation that exists among
the study subjects.
[0024] As used in this specification and claim(s), the words
"comprising" (and any form of comprising, such as "comprise" and
"comprises"), "having" (and any form of having, such as "have" and
"has"), "including" (and any form of including, such as "includes"
and "include"), and "containing" (and any form of containing, such
as "contains" and "contain") are inclusive or open-ended and do not
exclude additional, unrecited elements or method steps.
[0025] The term "or combinations thereof" as used, herein refers to
all permutations and combinations of the listed items preceding the
term. For example, "A, B, C, or combinations thereof" is intended
to include at least one of: A, B, C, AB, AC, BC, or ABC, and if
order is important in a particular context, also BA, CA, CB, CBA,
BCA, ACB, BAC, or CAB. Continuing with this example, expressly
included are combinations that contain repeats of one or more item
or term, such as BB, AAA, MB, BBC, AAABCCCC, CBBAAA, CABABB, and so
forth. The skilled artisan will understand that typically there is
no limit on the number of items or terms in any combination, unless
otherwise apparent from the context.
[0026] While the hardware and methods of this invention have been
described in terms of described embodiments, it will be apparent to
those of skill in the art that variations may be applied to the
hardware and/or methods and in the steps or in the sequence of
steps of the methods described herein without departing from the
concept, spirit, and scope of the invention.
[0027] Refer now to the drawings wherein depicted elements are not
necessarily shown to scale and wherein like or similar elements are
designated by the same reference numeral through the several
views.
[0028] Referring to FIG. 1, there is illustrated, in block diagram
form, system 100 for integrating call processing and voice
processing using a single processing means, which in this example
is a microprocessor 101. Microprocessor 101, which may be a
Motorola 68000 class microprocessor, communicates with hard disk
107 using driver circuitry 108. Hard disk 107 may store program
data, voice prompts, voice mail messages, and all other types of
speech used within system 100.
[0029] Microprocessor 101 may also include a watchdog timer 109 and
real-time clock source 110.
[0030] Microprocessor 101 may be coupled via bus 105 to flash
memory 111 and dynamic random access memory ("DRAM") 112. Flash
memory 111 may be used to store bootstrap data for use during power
up of system 100. DRAM 112 may store a program accessed by
microprocessor 101 during operation of system 100.
[0031] Bus 105 also couples microprocessor 101 to signal processing
circuitry, which may be a digital signal processor ("DSP") 102.
Digital signal processor ("DSP") 102 may implement a number of
functions traditionally implemented by discrete analog
components.
[0032] Referring next to FIG. 10, there are illustrated some
functions that may be
[0033] implemented in DSP 102. DTMF receivers 1001 may be
implemented using frequency domain filtering techniques. DTMF
receivers 1001 may detect standard DTMF (touch-tone) signals or
digits.
[0034] Automatic gain control ("AGC") 1002 may be a closed-loop
gain control system, which normalizes received audio levels during
recording.
[0035] Recording buffers 1003, which are coupled to AGC 1002, may
receive and store speech samples after they have passed through AGC
block 1002. These speech samples may be converted to .mu.-law PCM
(Pulse Code Modulation) and double buffered (several samples per
buffer). Microprocessor 101 may copy the record data out of DSP
buffers 1003 into RAM buffers (not shown), which may be located in
the microprocessor 101 data RAM area.
[0036] Fax tone detector 1004 may be implemented using frequency
domain filtering techniques. Fax tone detector 1004 may detect a
standard 1100 Hz FAX CNG tone (also referred to as the Calling
Tone).
[0037] Caller ID modems 1005 may be 1200 band FSK modems similar to
Bell 202-type modems. Caller ID modems 1005 may be implemented as a
frequency discriminator where a time delayed (e.g., quadrature)
signal is multiplied by the original signal, low pass filtered,
then sliced, which produce the square wave caller ID data
stream.
[0038] Call processing tone generators 3007 may be free running
oscillators, which generate the appropriate tones (and tone pairs)
that make up industry standard call processing tones. These tones
may include dial tone, busy/reorder tone, ring back tone, single
frequency (440 Hz) tone, and DTMF dialer tones.
[0039] Play buffers 1008 may replay data from hard disk 107 through
microprocessor 101 and place this play data in buffers 1008. This
data may be converted from an 8-bit .mu.-law PCM signal to 14-bit
linear data.
[0040] Conference bridges 1006 may allow multiple conference
bridges to mix together conferees into a multi-party conference.
These conferees may be a mixture of inside and outside parties. A
combination of "loudest speaker" and "summing" may be utilized.
[0041] DSP 102 may communicate with microprocessor 101 via a host
interface port ("HIP") via bus 105. The HIP link may support a
command-based protocol which may be used to directly read or write
DSP memory locations. DSP 102 may be a RAM-based part and may have
its program downloaded from microprocessor 101. Once downloaded and
running, microprocessor 101 (the host) may poll, for events or
receive interrupts indicating that data is available. DSP 102
speech connections may be made over an industry standard 32-time
slot, 2.048 megabits per second. (Mb/s) digital serial link 124.
Link 124 may occupy one of the digital highways implemented by
digital cross-point matrix 103. Each service of DSP 102 may occupy
a single time slot. For example, DTMF receiver 1001 may occupy time
slot 0 while conference bridge circuit 12 may occupy time slot
31.
[0042] Digital cross-point matrix 103 may also be coupled to bus
105 and operate to connect any voice path to any other voice path.
Digital cross-point matrix 103 may be a VLSI (Very Large Scale
Integration) integrated circuit. An example of digital cross-point
matrix 103 is manufactured by MITEL Semiconductor Corporation as
part No. 8980. Digital cross-point matrix 103 may communicate with
microprocessor 101 via a memory mapped input/output ("I/O") scheme.
A command/control protocol may be used for communication between
microprocessor 101 and digital cross-point matrix 103 via bus 105.
Cross-point matrix 103 may be coupled by highway 124 to DSP 102.
Cross-point matrix 103 may be coupled by connection 325 to highway
121. Cross-point matrix 103 may also be coupled to peripheral cards
by highways 122 and 123. The peripheral cards are described in
further detail below with respect to FIG. 3.
[0043] Connections 121-125 are also referred to herein as
"highways," which may be transmission links using time-division
multiplexing ("TDM") as a means for transmitting and receiving
data.
[0044] Digital cross-point matrix 103 may be capable of making 256
simultaneous fully non-blocking connections within system 100.
However, system 100 may be upgraded by adding additional DSPs
and/or cross-point matrices.
[0045] Cross-point matrix 103 may make connections using the TDM
highway by receiving instructions from microprocessor 1.01 to
interconnect channels within the frames of the TDM bit stream. This
results in the non-blocking capability of cross-point matrix 103,
and also allows for a single voice resource, caller, or voice
message to be simultaneously coupled to multiple other voice
resources, station or CO originated callers, and/or voice
messages.
[0046] Gate array 104 may be an SRAM (Static Random Access Memory)
based device. An example of gate array 104 is manufactured by
XILINX. Gate array 104 may be responsible for generating system
timing. A master clock signal may be provided by microprocessor 101
at 16.384 MHz. This clock signal may be divided down to provide a
number of phase coherent system clocks such as 4.096 MHz, 2.048
MHz, and 8 KHz (frame sync). In addition, a 5-bit time slot counter
may be implemented, which allows all the system CODECs to detect
the appropriate time slot to use (e.g., 0-31). An additional
divider chain may be included to divide the system clock down to 20
Hz, which may be used by the ringing generator power supply (not
shown).
[0047] Gate array 104 may be downloaded at boot-up by system
software. Gate array 104 may be based on a SRAM architecture. That
is, the internal fusible links commonly found in programmable logic
may be actually stored in volatile SRAM. Because of this
architecture, gate array 104 may be downloaded after power-up.
Also, note the added flexibility of being able to modify the logic
by simply loading new system software.
[0048] Bus 105 may also be coupled to modem 106, which may provide
a capability of calling into system 100 on a remote basis to load
additional programs, voice prompts, etc., or updates thereto, into
hard disk 107. Modem 106 may be coupled to coder/decoder ("CODEC")
113, which may be coupled to highway 121. This connection may allow
coupling of modem 106 through cross-point matrix 103 to CO lines
through highway 122 and the p-card described below with respect to
FIG. 3.
[0049] Highway 121 may also be coupled to a dual subscriber line
access chip 114, which is well-known in the art, and which may be
coupled to analog ports 115 and 116, which may provide an ability
for system 100 to communicate to analog-type connections such as
cordless telephones and fax machines.
[0050] Highway 121 may also be coupled to CODEC 117, which may be
coupled to transformer 118 to a music source 119, which may provide
an ability to couple an external music source to a received call
through cross-point matrix 103 for such things as providing the
received call with music on hold.
[0051] Power to system 100 may be provided through switching power
supply 120, which may convert AC to the various DC supply voltages
needed by circuitry within system 100.
[0052] Referring next to FIG. 3, there is illustrated
peripheral-card ("p-card") 300, which may be coupled to main board
190 of system 100. Main board 190 may communicate with p-card 300
via a multi-drop async serial link 307. This connection 307 may be
made directly to microprocessor 101 (via butlers not shown). P-card
300 may provide interconnections between CO lines and extension
lines to system 100.
[0053] Microcontroller 301 may be an 8-bit microcontroller, an
example of which is manufactured by Hitachi as Part No. H8, which
may control all the real-time functions associated with p-card 300.
Microcontroller 301 may be responsible for all low-level
communication with the EKTs 1400 (electronic key telephones) (see
FIG. 14) and CO lines. A low-level event is an event which is
specific to the hardware and is required to be handled in
real-time. These events may be unique to the EKT or CO trunk
protocol. In contrast, high-level events can be abstracted to have
no correlation to actual hardware. An example of a high-level event
might be "Turn the SPKR LED On." The corresponding low-level event
would be "Send HEX Code 21 to EKT Address 4." This level of
abstraction helps stabilize the complex system software. Another
example would be that system software can send a command to seize a
CO trunk without being concerned with the low-level differences
between a ground start or DID trunk. Some of the low-level tasks
may include updating EKT LEDs and LCD displays, decoding key press
messages from the EKTs 1400, scanning the CO status bits, and
filtering RING and CO seizure events.
[0054] Microcontroller 301 may convert these low-level real-time
events to high-level events, which may form a protocol referred to
as the ESi Command Language ("ECL"). This ECL protocol may be
implemented on multi-drop async serial channel 307 between main
board 190 and all p-cards 300 in system 100. Microcontroller 301
may contain 2 async serial ports. One of these serial ports may be
connected to main board 190, and the other port drives data
transceiver and multiplexer 302.
[0055] When p-card 300 is plugged into main board 190 (e.g., via
ribbon cable (not shown)) a card address is assigned to p-card 300.
This card address may be read by microcontroller 301 and used to
filter commands over communication link 307. When main board 190
software wants to communicate with the specific p-card 300, the
address may be sent in a message packet, which all p-cards 300
receive. P-cards 300 may match the address in the message to the
hard wired address on the ribbon cable. If a match is made, that
p-card 300 responds to the command set.
[0056] Microcontroller 301 may contain an internal program memory
(not shown), and may be connected to an external SRAM 303. The
internal program memory may contain a bootstrap program, which upon
reset or power-up, may request a fresh firmware load from main
board 190. This firmware load may be transferred to SRAM 303. Upon
download completion, the program may be run from within SRAM 303.
This scheme may allow for microcontroller 301 -firmware to be
updated and loaded at any time.
[0057] Main board 190 may source all system timing through block
304. Timing signals to p-card 300 may include a 2.048 MHz clock
signal, an 8 KHz frame sync, which may signify the first time slot
of a 32 time slot highway, and 5 time slot counter bits, which may
represent a binary count from 0 to 31.
[0058] As mentioned above, p-card 300 may be assigned a card slot
address when it is connected to main board 190. This card slot
address may be used to calculate which time slots p-card 300 should
be using. The time slots used for the CO CODECS 1204 (see FIG. 12)
may be generated by time slot assignment circuitry contained in the
DSLAC chip. There may be two separate 2.048 MHz (32 time slot)
highways 122 and 123 that run between main board 190 and p-card
300. One (123) may be for the EKTs 1400 and the other (122) may be
for the COs.
[0059] Referring to FIGS. 3 and 11, EKT interface 306 describes a
connection between system 100 and electronic key telephone ("EKT")
1400. This interlace may include two physical pairs of wires
running between system 100 (often referred to as a Key System Unit
("KSU")) and EKT 1400. One of these pairs may support an analog
bi-directional audio path, and the other may support a
bi-directional digital control channel.
[0060] EKT 1400 may be connected to the KSU via transformers 1101
and 1102, providing a high degree of isolation as well as
longitudinal balance. Transformer 1101 may be for the audio path,
and transformer 1102 may be for the data path on each end of the
connection. Power may be supplied to EKT 1400 by phantoming the
power through the center taps of transformers 1101 and 1102. The
KSU may supply a nominal voltage of 36 volts DC, which may pass
through a positive temperature co-efficient varistor ("PTC") 1103.
PTC 1103 may act as a resettable fuse, which may become very
resistive during excessive current flow (such as when a short in
the station wiring occurs). EKT 1400 may regulate down to +12 and
+5 volts.
[0061] The audio path may be a dry analog bi-directional path
including a traditional hybrid (2:4 wire converters) on each end.
The audio path on p-card 300 may be converted to a 4-wire path by
the hybrid circuit in interface 306. The separate transmit and
receive paths may be gain adjusted and connected to CODEC 1104.
CODEC 1104 may convert the analog signals to digital and may
present these voice signals to EKT highway 123. EKT highway 123 may
include a 2.048 Mb/s serial stream, which may be divided into 32 64
Kb/s time slots. Each CODEC 1104 may occupy one time slot on
highway 123. System 100 may reserve two time slots per EKT 1400 for
future migration to a fully digital 2B+D EKT where two 64 Kb/s
digital channels may be available to each station instrument.
[0062] Timing for CODECs 1104 may be supplied by time slot
generation block 304, which may be coupled to the time slot counter
output from system timing block 104 (see FIG. 1).
[0063] The EKT data may be produced by a UART (Universal
Asynchronous Receiver/Transmitter) in microcontroller 301. This NRZ
transmit and receive data may be presented to data transceiver and
multiplexer 302. A single data transceiver may be used for all 8
EKT circuits and may be multiplexed through an 8-channel analog mux
to each EKT data transformer 1102 in a round-robin fashion.
[0064] Messages to EKT 1400 may include commands such as POLL,
TURN_ON_LED, WRITE_LCD_CHARACTER, RING PHONE, etc. Response
messages from EKT 1400 may include a lower level key command in the
first 5 bits and a single hook switch bit in the 8th bit. If the
7th bit of the response message is set, a high level response
command such as FIRMWARE_VERSION or TERMINAL_TYPE may be present in
the first 5 bits.
[0065] Referring next to FIGS. 3 and 12, the loop start central
office ("CO") lines may be supplied by a local telecommunications
company and may include a wet balanced differential audio pair. The
term "wet" refers to the fact that a voltage (e.g., -48 volts) is
present on the pair. System 100 may request dial tone from the CO
by providing a nominal 200 ohm loop across the TIP and RING
conductors and may release the connection by opening the loop.
[0066] The CO may ring system 100 by placing a 90 vrms AC, 20 Hz
sine wave on the TIP and RING conductors. System 100 may seize the
line by going off hook.
[0067] P-card 300 may incorporate a unique circuit, which may
monitor the voltage present across TIP and RING of each CO. This
line voltage monitor circuit 1202 may serve to detect the ring
voltage present during ringing (ring detection) and monitoring the
CO line status for conditions, such as whether the CO is plugged in
or if someone is off hook in front of system 100. The latter may be
used to detect theft of sendee or allow a credit card verification
terminal to be used without interfering with normal system
operation.
[0068] Voltage monitor 1202 may include a balanced differential
op-amp connected across TIP and RING of the CO lines through a very
high impedance (e.g., >10 Mohms). The output, of the four
voltage monitor op-amps may be fed to an analog-to-digital
converter with a built-in analog multiplexer (not shown).
Microcontroller 301 firmware may monitor the line voltages.
[0069] There may also be a balanced differential AC coupled op amp
across the CO TIP and RING to monitor the low level audio tones
present during caller ID. The output of these op-amps may be
selected via an analog switch during the idle period and may be
connected to the CO line CODEC 1204.
[0070] To correctly terminate the CO line (seizure) care may be
taken to satisfy the DC loop requirements (e.g., -200 ohms) and the
AC Impedance requirements (e.g., -600 ohms). The classic approach
has been to terminate TIP and RING with an inductor (called a
holding coil), which has a large inductance (e.g., >1 Hy) and a
DC resistance of (e.g., -200 ohms). The inductor separates the AC
and DC components to give the desired effect. The problem is that
the Inductor must be large enough not to saturate with currents as
high as 100 milliamps. An inductor that satisfies these
requirements is physically cumbersome.
[0071] P-card 300 may incorporate a solid state inductor circuit
called a gyrator (not shown) to implement the holding coil
function. This single transistor may emulate an inductor with the
above requirements while taking up very little PCS space.
[0072] A small solid state relay (not. shown) may be used as the
hook switch. When energized, the gyrator holding coil may be placed
across TIP and RING closing the loop. The audio present on TIP and
RING maybe AC coupled to a small dry transformer 1203. The
secondary of this transformer 1203 may be connected to the AC
termination impedance and to the CODEC 1204, which may be
implemented on a dual subscriber line access chip ("DSLAC").
[0073] High voltage protection may be provided for all paths on the
TIP and RING connections. These paths may include TIP to RING, TIP
to GROUND, RING to GROUND, and TIP and RING to GROUND. This high
voltage protection is accomplished by first passing the TIP and
RING conductors through positive temperature coefficient varistors
(not shown). These varistors may act as resectable fuses. When
excessive current flows through these varistors, they become
resistive thus limiting the current flow. When the excessive
current is stopped, the original resistance is restored.
[0074] DSLAC 1204 may include two identical circuits, which may
contain the CODEC, DSP-based echo canceller, gain control, and time
slot assignment circuit. DSLAC 1204 may be controlled by
microcontroller 301 to set parameters such as echo canceller
coefficients, gain coefficients, and time slots.
[0075] Referring next to FIG. 15, the following is an example of
how a. call may be processed by system 100. A call may come in on
one of the available central office ("CO") lines (step 1501),
wherein a speech path, for the CO line may be coupled through
digital cross-point matrix 103 to an available "play" channel
(e.g., play buffer 1008) in DSP 102 (step 1502). Also during
set-up, a connection may be made to an available DTMF receiver
1001. A connection may also be made to one of the available fax
tone detector channels 1004 in case the incoming call is a
facsimile transmission. In step 1503, microprocessor 101 may access
hard disk 107 and transfer, speech data to play buffers 1008. Next,
in step 1504, a determination may be made whether or not FAX tones
have been detected by FAX tone detector 1004. If FAX tones are
detected, then in step 1505, microprocessor 101 may instruct
digital cross-point matrix 103 to connect the incoming call to one
of analog ports 115 or 116 coupled to DSLAC 114. If FAX tones are
not detected, then the process may determine whether or not DTMF
tones have been detected in step 1506. If yes, then in step 1507,
digital cross-point matrix 103 may be instructed to connect the
incoming call to an extension coupled to p-card 300. If DTMF tones
are not detected, then the process may determine whether nor not a
predetermined amount of time has passed in step 1508. If yes, then
the call may be terminated by freeing resources in step 1509 and
tearing down the call in step 1510 to place the system in an idle
state (step 1511).
[0076] If the incoming caller dialed an extension and that
extension has answered, a speech path connection may be made
between the extension and the incoming CO line.
[0077] Referring next to FIG. 14, there is illustrated EKT 1400,
which may include many of the well-known features of a typical
telephone, such as LCD display 1401, soft feature keys 1402 for
such features as Station, Speed Dial, Line Keys, etc,
speaker/handset volume control 1404, and message and speaker LEDs
1403. Further described in detail below are the program/help key
1407, the record/monitor key 1406, and the voice mail key 1405,
which may be part of the fixed feature keys on EKT 1400.
[0078] Referring next to FIG. 4, there is illustrated a process for
recording all or a portion of an incoming call after it has been
connected to an extension (e.g., EKT 1400) (e.g., by digital
cross-point matrix 103 and p-card 300). Such a recording can occur
while a user is speaking with an incoming call over the extension
(e.g., EKT 1400), However, such a recording of the incoming call
may occur using any telecommunications device coupled to system 100
if it is supplied with some type of mechanism for initiating the
recording process to be discussed. In step 401, the user presses
record key 1406 on the extension (e.g., EKT 1400). One skilled in
the art will, surely appreciate that any means for activating a
record signal may be utilized, such as the depression of a physical
button, the touching of a touch screen (display 1401 could utilize
such a touch screen), or even voice activation of the record
sequence. Such a record sequence activation signal may be
transmitted from the extension (e.g., EKT 1400) to interface 306
via transformer 1102 (see FIG. 11), which may pass the signal to
microprocessor 101 through data transceiver and multiplexer 302 and
microcontroller 301. Next, in step 402, a determination may be made
whether or not the extension (e.g., EKT 1400) is connected to a
valid call. A valid call may be defined as energy being detected on
the line, and the energy is not dial tone. If not, the process may
proceed to step 403 to ignore the record activation signal.
However, if the extension (e.g., EKT 1400) is connected to a valid
call, the process may proceed to step 404 to determine whether or
not a record resource is available. This may be accomplished by
determining whether or not a recording buffer 1003 is available in
DSP 102. If not, the process may proceed to step 405 to display an
error message to the telephone extension user. This may be
accomplished by some type of visual (e.g., on display 1401 or via
an LED on EKT 1400) or audible indication provided by the extension
(e.g., EKT 1400). This may be implemented by sending from
microprocessor 101 to microcontroller 301 such an error message,
which is then transmitted to the extension (e.g., EKT 1400) through
data transceiver and multiplexer block 302 and transformer 1102
within interface 306. Next, in step 406, a counter may be
incremented to record that a record resource was not available.
[0079] If in step 404, a recording buffer 1003 is available (e.g.,
in DSP 102), such a recording buffer 1003 may be assigned to the
record sequence in step 407. Thereafter, in step 408, the recording
buffer 1003 may be connected, to the appropriate speech path via
highway 124 and digital cross-point matrix 103. As noted above in
the discussion regarding digital cross-point matrix 103, digital
cross-point matrix 103 may have the ability to couple multiple
resources to each other. Therefore, digital cross-point matrix 103
may be able to couple recording buffer 1003 (along with automatic
gain control function 1002) to the incoming call, which has
previously been connected (and remains connected) to the pertinent
extension (e.g., EKT 1400).
[0080] Thereafter, in step 409, the recording process begins. In
addition to recording the ongoing phone conservation, system 100
may also store the called extension number, the incoming calling
telephone number, and the date and time of the call. These data may
be stored in a recording record, which may be associated with the
actual recording. The recording data record may be written to hard
disk 107, and may be available for display when the recording is
accessed. At the time the recording begins, a timer may be started
to accumulate the duration of the recording. When the call is
completed, the duration may be added to the recording data record,
and may be written to hard disk 107. When the recording is played
back, the incoming caller phone number, the date and time of the
recording, and the duration may be displayed (e.g., on display
1401). Next, in step 410, a determination may be made whether or
not a beep tone feature has been enabled. If not the process may
proceed to step 601 in FIG. 6, However, if a beep tone feature has
been enabled, the process may proceed to step 411 to start a beep
timer, which may be required by law in certain jurisdictions.
Implementation of step 411 is further described below with respect
to FIG. 5.
[0081] The recording sequence illustrated in FIG. 4 may be
implemented as a software program stored within hard disk 107,
which may be up-loaded to DRAM 112 for operation by microprocessor
101.
[0082] Referring next to FIG. 6, there is illustrated a process for
terminating a recording sequence. In step 601, one of the parties
(e.g., the incoming call or EKT 1400) may hang up, or the
termination of the recording sequence may be initiated by again
pressing record key 1406 (deactivation of the recording sequence).
In response to one of these signals, in step 602. the recording
process may be stopped. Thereafter, in step 603, the recording,
which may have been temporarily stored within recording buffer
1003, may be recorded in the mailbox assigned to the extension
(e.g., EKT 1400) that initiated the recording sequence. The other
data, such as the called and caller telephone numbers, time and
date information, and recording duration may also be stored within
the extension telephone's mailbox. Such a mailbox may be stored
within hard disk 107. Thereafter, in step 604, the beep timer may
be terminated. Thereafter, in step 605, recording buffer 1003,
which may have been assigned to this record sequence, may be freed.
Then, in step 606, digital cross-point matrix 103 may disconnect
the speech path from recording buffer 1003, and the process may end
at step 607.
[0083] Referring next to FIG. 5, there is a flow diagram
illustrating a process for implementing a beep timer. A beep timer
may be provided so that an audible beep is heard by both parties
during a conservation that is being recorded. The beep may be heard
every 15 seconds, and may have a configurable duration between 40
and 500 milliseconds. In step 501, the beep timer has been enabled.
In step 502, a determination may be made whether or not recording
buffer 1003 is still connected to the pertinent speech path. If
not, the process may proceed to step 503 to terminate the beep
timer. However, if recording buffer 1003 is still connected to the
speech path (e.g., by digital cross-point matrix 103), the process
may proceed to step 504 where the speech path transmit is opened.
Next, in step 505, the speech path may be connected to a tone
produced by call processing tone generator 1007 by DSP 102. As a
result, the tone may be heard by one or both of the incoming call
and/or extension user. In step 506, a delay period (e.g., 200
milliseconds) may be allowed to pass. After passage of the delay
period, the tone generator 1007 may be disconnected from the speech
path in step 507. The beep timer process may be started again in
step 508. The process may end step 509. An advantage of the
recording sequence is that it can he performed without any
interruption in the connection between the call and the extension
(e.g., EKT 1400), Additionally, it may be performed with merely the
activation of a single signal. However, a sequence of signals may
be utilized to initiate the recording sequence, such as the
entering of a code (e.g., by the user using the touch-tone keypad
on EKT 1400).
[0084] Furthermore, the recording sequence may be initiated while a
user is screening an incoming call or while a voice message is
being placed in the user's mailbox. These recordings may be
accomplished following the process described in FIG. 4 as was
described earlier for recording all or a portion of an incoming
call.
[0085] Referring next to FIGS. 7A-7D, there is illustrated a flow
diagram for implementing an interactive help sequence (e.g., verbal
user guide) whereby "help" messages may be provided to a user of
system 100. This feature can alleviate the need for the user to
possess and access a written help manual. Note, however, that one
skilled in the art will appreciate that the process illustrated in
FIGS. 7A-7D may be utilized to play or display any type of messages
to a user, and not just those associated with a help menu.
[0086] In step 700, the user may press a key or button 1407 on the
extension (e.g., on EKT 1400), wherein the key or button 1407 may
be associated with a help menu (or any other information) stored
within system 100. A. signal activated by the pressing of key 1407
may be sent from the extension (e.g., EKT 1400) through p-card 300
to microprocessor 101. As noted above with respect to the record
sequence, any type of activation signal may be utilized to initiate
the sequence. Next, in step 701, microprocessor 101 may assign a
play channel or butter 1008 (e.g., within DSP 102) to be coupled to
the extension (e.g., EKT 1400) through digital cross-point matrix
103. Next, in step 702, a determination may be made whether or not
such a play resource (e.g., buffer 1008) is available (e.g., in DSP
102). If not, the process may proceed to step 713 to display an
error message (e.g., on display 1401), or some other type of error
indication, such as a tone or LED light 1403, to the user on the
extension (e.g., EKT 1400). This may be done in the same manner as
described above with respect to step 405 in FIG. 4, Additionally, a
reorder tone may be generated (e.g., by call processing tone
generator 1007) to be connected to the extension (e.g., EKT 1400).
Next, in step 714, a lock-out state may be entered. In this lockout
state, all key presses on the extension (e.g., EKT 1400) may be
ignored, so that the only option for the user is to hang up the
extension. Then, in step 715, the process may wait for the
extension (e.g., EKT 1400) to go on hook.
[0087] If in step 702 a play buffer 1008 is available, then in step
703, microprocessor 101 may assign, a play buffer 1008 (e.g.,
within DSP 102) to the extension (e.g., EKT 1400) and digital
cross-point matrix 103 may connect (e.g., via highway 124), play
buffer 1008 and associated filter 1009 to the time slot with, which
the extension (e.g., EKT 1400) is associated. Additionally,
microprocessor 101 may couple a DTMF receiver 1001 (e.g., within
DSP 102) to a time slot in digital cross-point matrix 103
associated with the extension (e.g., EKT 1400) in order to
recognize any DTMF tones actuated on the extension (e.g., EKT 1400)
by the user (see steps 705 and 707-710 described below).
Thereafter, in step 704, a message may be played to the user on the
extension (e.g., EKT 1400). Such a message may be downloaded from
hard disk 107 through, microprocessor 101 to play buffer 1008. Such
a message may be "To access the help menu, press 0." Thereafter, in
step 705, a determination may be made whether or not a key on the
extension (e.g., the "0" key on EKT 1400) has been pressed by the
user. If not, the process may proceed to step 712 which operates a
time out function. If the designated key (e.g., "0") has been
pressed in step 705, the process may proceed to step 706 where
another "Help" prompt may be played by play buffer 1008 to the user
(e.g., via the speaker on EKT 1400). Such a message may be "To
leant how to program your phone, press 1, to learn how to use voice
mail, press 2 . . . . "
[0088] Next, in steps 707-710, determinations may be made whether
the user depresses a digit on the touch-tone keys of the extension
(e.g., EKT 1400) (or voice activation may be utilized in response
to the user stating a number) in response to the introductory
prompt message 706. A DTMF receiver 1001, which may have been
connected to the speech path associated with the extension (e.g.,
EKT 1400) initiating the help sequence (see step 703), recognizes
which of digits 707-710 have been depressed and may play a
corresponding prompt message (see steps 720, 740, 760), in response
thereto as programmed within system 100. For example, in step 707,
if the "1" key on EKT 1400 is pressed by the user, then the process
proceeds to step 720 in FIG. 7B where an introductory prompt is
played to the user regarding use of the EKT 1400, which may include
options presented to the user for pressing certain digits to access
associated help information. For example, if the "1" key is pressed
by the user in step 721, then in step 722 a prompt may be played to
the user regarding how to answer the extension using the features
of EKT 1400. If the "2" key is pressed on EKT 1400 by the user
(step 723), then, in step 724, a prompt may be played to the user
on how to place a call using EKT 1400, If the "3" key is pressed by
the user (step 725), then, in step 726, a prompt may be played to
the user on how to transfer a call. Similarly, if the "4" key is
pressed by the user (step 727), then a prompt may be played to the
user on how to conference a call (step 728). If system 100
determines that the "5" key has been pressed on EKT 1400 by the
user in step 729, their a prompt on how to answer a call under Call
Waiting may be played to the user in step 730.
[0089] After each of prompts 722, 724, 726, 728, and 730 have been
played to the user, the process may return to step 720 to repeat
the process. However, if the user presses another key (e.g., the
key) in step 731, then the process may return to step 706 in FIG.
7A. A time-out feature may be implemented in step 732. Thus, if the
user does not press any key after step 720 after a predetermined
amount of time, the process may proceed to step 716 in FIG. 7A.
[0090] If the user has pressed the "2" key in step 708, the process
may proceed to step 740 (FIG. 7C) to play an Introductory prompt
regarding the use of voice mail. Such a message may provide the
option to listen to further messages upon the pressing of selected
keys on the extension (e.g., EKT 1400). For example, if in step
741, it is determined that the user has pressed the "1" key, then
the process, in step 742, may play a message on how to leave a
message. Likewise, steps 743 and 744 may implement a process for
playing a message on how to transfer a call to voice mail. Steps
745 and 746 may implement a process for informing the user on how
to pick up an internal voice mail. Additionally, steps 747 and 748
may implement a process for informing the user on how to pick up an
external voice mail. Likewise, steps 749 and 750 may implement a
process for informing the user on how to record a personal greeting
for their voice mail box.
[0091] Again, in a manner similar to that described previously with
respect to step 731, in step 751, a process may be Implemented for
permitting the user to return to step 706. The time out function
may be implemented in step 752 in manner similar to that described
previously with respect to step 732.
[0092] Returning to FIG. 7A, if, in step 709, the user presses
another key (e.g., the "3" key), then the process may proceed to
step 760 in FIG. 7D to play an introductory prompt regarding help
information on various phone features. In the process illustrated
in FIG. 7D, the user may then press any key on the extension (e.g.,
EKT 1400) and receive help information corresponding to the pressed
key. In step 761, the process may determine whether a key has been
hit or pressed by the user. If yes, in step 762, whichever key has
been pressed by the user, the signal generated by the extension
(e.g., EKT 1400) corresponding to the pressed key is analyzed by
DTMF receiver 1001, so that microprocessor 101 can access an
appropriate help message to play to the user using play buffer
1008. This may be performed in step 763, where the key code may be
used as an index into a prompt, or message, array stored within
system 100. The playing of the corresponding message may be
performed in step 764. The process then may return to step 761 to
determine whether or not another key has been pressed by the user.
Step 765 may determine whether or not. at this time the key has
been pressed by the user. If yes, the process may then returns to
step 706.
[0093] The time-out feature implemented in step 766 may provide for
an exit to step 716 in FIG. 7A if no key is pressed by the user
within a predetermined amount of time.
[0094] Returning to FIG. 7A, if in step 712, a predetermined amount
of time does pass before the "0" key has been pressed by the user
as determined by step 705, the process may proceed to step 716 to
free any accessed resources. Teasing down of the call may be then
performed in step 717 so that the phone is placed in an idle state
(step 718). Steps 716-718 may also be entered if a predetermined
amount of time passes in steps 711, 732, 752, or 766.
[0095] Referring next to FIG. 8, there is illustrated a flow
diagram for implementing a context sensitive help menu. Such a help
sequence can be implemented so that the help messages sent to the
user relate to the particular function the user is currently
implementing. Note that. FIG. 8 may apply when the user is already
connected to an outside call (a call which is not an intercom
call). In step 801, the user may press help key 1407. or performs
some other type of activation of the help sequence, such as
described, above with respect, to step 701 in FIG. 7A. Thereafter,
in step 802, the user may press the specific key that help is
required on. For example, if the user is connected to an outside
call, and needs to transfer the call, the user may press the help
key 1407, followed by the transfer key 1408 to receive
instructions. While the user is receiving instructions, the other
party may hear music provided via music source 119. Thereafter, in
step 804, a determination may be made whether or not a play
resource is available in DSP 102. In other words, a determination
may be made whether or not a play buffer 1008 is available. If not,
an error message may be provided to the user in step 805 in a
manner similar to the one described above with respect to step 712
in FIG. 7A.
[0096] If a play resource is available in step 804, the process may
proceed to step 806 to assign a play resource. Thereafter, in step
807, the play resource (i.e., play buffer 1008 and associated
filter 1009) may be connected to the speech path to which EKT 1400
is connected. Thereafter, in step 808, the appropriate help message
may be played to the user (e.g., by first downloading the help
message from hard disk 107 via microprocessor .101 to the play
buffer 1008). Thereafter, in step 809, alter the help message has
been played to the user, the user may be reconnected to the other
party in the call. The process may end at step 810.
[0097] Referring next to FIG. 9, there is illustrated a flow
diagram for implementing real-time call screening. It allows a
station user to listen to calls being left in their mailbox. If
desired, the user can for example lilt the handset of the extension
(e.g., EKT 1400) to intercept the call at any time. The process may
begin in step 901, where an incoming call has been routed to an
extension and the extension has not answered the call After a
configurable number of rings, the call is transferred to the
extension's mailbox. Thereafter, in step 902, a determination may
be made whether or not the call screening application has been
enabled. Each extension may enable call screening mode by pressing
key 1406 while the extension is idle. If not, system 100 may
process the Incoming call normally in step 907 where the incoming
call may leave a message in the station's mailbox.
[0098] However, if in step 902, call, screening has been enabled,
the process may proceed to step 903 where the extension (e.g., EKT
1400) associated with the station that was called continues to ring
while the mailbox plays a message to the incoming call. In step
904. the incoming caller may hear a beep, which indicates it is the
proper time to leave a message. Next, in step 905, a speech path is
set up to the extension (e.g., EKT 1400) that was called, so that
the message can be heard on the speaker as it is being left.
Thereafter, In step 906, the user is permitted to monitor the
message while it is being left by the incoming call. Next, in step
908. the user may for example lift their handset. If so, in step
909, the voice mail messaging may be terminated. In step 910, the
incoming call may be connected to the station user by (e.g.,
digital cross-point matrix 103). If in step 908 the user does not
for example lift their handset, then the process may proceed to
step 911 when the user presses the record/monitor key 1406 (or a
similar action), the process may proceed to step 912 to turn the
monitor speaker off and then in step 913 the call may proceed into
voice mail in a normal fashion.
[0099] Referring next to FIG. 16, there is illustrated a flow
diagram for implementing Quick Groups, which allows a user to leave
or copy a voice mail message to multiple destination mailboxes by
merely pressing the desired Direct Station Select ("DSS") key or
dialing the number of the extension. There may be 16 DSS keys at
1402, on each EKT 1400. These DSS keys can be programmed to provide
one button access to extensions and outside lines. In step 1601,
the user of the extension (e.g., EKT 1400) is listening to a voice
mail message, which may be a "new" or "old " voice mail message
previously recorded into the user's mail box, or the user may be
listening to the voice mail message while it is being recorded by
another person into the user's mail box. Thereafter, in step 1602,
a determination may be made whether or not a specified key (for
example, the "6" key) has been pressed by the user. If not, a
time-out operation may be implemented with step 1613. If the
specified key has been pressed by the user In step 1602, the
process may proceed to step 1603 to play a message to the user,
which may prompt to the user to enter a destination. Thereafter, in
step 1604, the system may receive a code for the entered
destination, which may be a pressed DSS key or digits dialed by the
user on the extension (e.g., EKT 1400).
[0100] Next, in step 1605, another message may be played to the
user requesting that the user enter another destination for the
voice mail message to be sent. In step 1606, a determination may be
made whether or not the user has pressed another DSS key or has
entered an extension. If yes, the process may return to step 1604
where the signals associated with the pressed digits or DSS key are
received. However, if the user has not entered another destination,
a determination may be made in step 1607, whether or not a
specified key (for example, the "1" key) has been pressed by the
user. If yes, in step 1608, the system may play a message to the
user requesting if the user wishes to record a message to be
appended to the beginning of the voice mail message sent by the
user. Any message entered by the user may be recorded by a record
buffer 1003, which may have been coupled to EKT 1400 by digital
cross-point matrix 103. Next, in step 1610, the introductory
message left by the user and the voice mail message noted in step
1601 may be copied to all mailbox destinations entered by the user
in steps 1604-1606. The process may then end at step 1611.
[0101] Step 1609 implements a method by which the user can. press a
specified key (for example, the "6" key) in order to copy the voice
mail message noted in step 1601 to destinations entered by the user
in step 1610, in the instance where the user has not decided to
record an introductory message (see steps 1607-1608).
[0102] Step 1612 permits a return to step 1606 for a specified
amount of time. If such a specified amount of time has passed
without any buttons pressed by the user, then the process may
proceed to step 1614 to tear down the call and enter an Idle
state.
[0103] Referring next to FIG. 17, there is illustrated another
process for implementing Quick Groups. This process enables a user
to leave a new message entered by the user for recording into a
number of destination mailboxes specified by the user. In step
1701, the user picks up an idle extension (e.g., EKT 1400) for the
purpose of leaving a new message in a number of specified
destination mailboxes. In response, in step 1702, the extension
(e.g., EKT 1400) goes off hook. Thereafter, in step 1703, the user
presses voice mail key 1405. Thereafter, in step 1704, the process
may provide for the user to press a DSS key or digits associated
with a destination extension. In step 1705, the user may press the
DSS key or dial the digits associated with each extension which is
to be added to the group to receive the message. In step 1706, once
all the extensions to receive the message have been selected, the
process may proceed to step 1707, where upon hearing the voice mail
beep of the destination, the user leaves the desired message. In
step 1708. the user may then hang up the extension (e.g., EKT 1400)
by going on hook.
[0104] Thereafter, in step 1709, the user may copy the message to
all other desired destination mailboxes. The process may then end
at step 1710.
[0105] Although the present invention and its advantages have been
described in detail, it should be understood that, various changes,
substitutions and alterations can be made herein without departing
from the spirit and scope of the disclosed embodiments of the
invention.
* * * * *