Note: Descriptions are shown in the official language in which they were submitted.
~3~ 9
-- 1 --
A VOICE MAIL SYSTEM WIrr~ IMPROVED
DETECTION AND CANCELLATION
BACKGROUND OF THE INVENTION
This invention generally relates to voice mail
systems and more particularly to system improvements in
detection of messages and valid DTMF signals or tones
and in cancellation of a previously recorded audio
message from a message to be recorded.
As e~plained in depth in other patents such as
U S. Patent No. 4,371,752 to Matthews and U.S. Patent
No. 4,747,126 to Hood et al., owned by the present
assignee, a voice mail system records and plays back
telephone messages intended for one of a plurality of
system users. The caller is typically greeted and
instructed by prompt signals, to which he responds by
entering touch tone (DTMF) commands. The caller can
thereby record a message, review it, or perform other
system operations. The user similarly can use DTMF
commands for system control, such as to retrieve or
delete messages for him.
A recurring problem in voice mail systems is
triggering of the system by noncommand or invalid DTMF
signals. These signals can occur both in a message
received for recording, as a result of a caller's voice,
and in the playback of a previously recorded message, as
a result of voice or a DTMF command entered by the
caller. A tone detector within the system is adapted to
respond to DTMF commands during both record and playback
and may inadvertently respond to an invalid signal.
Prior attempts to solve the playback problem
have only been partially effective. The method
described in U.S. Patent No. 4,747,126 limits recorded
tones to a duration shorter than is required for the
tone detector on playback to respond. But these and
other methods do not effectively handle signals produced
on playback by the caller's voice, a voice prompt in the
:
:
~L304~49
-- 2
system, signal noise, etc. r that contain the same tones
as DTMF signals.
Another problem present in prior voice mail
systems is line interference with the caller-generated
DTMF commands. Different telephone systems have
different impedances in their central office lines which
affect the transmission and echoing of audio signals to
a voice mail system. Matching the impedance of all
possible systems to the voice mail system is not
practical. Prior voice mail systems have generally
incorporated a single characteristic impedance that will
work passively with whatever telephone system is
ultimately connected to the mail system. However,
telephone lines that differ in impedance from the
characteristic impedance can interfere with valid DTMF
tone commands generated by the caller, causing the voice
mail system to ignore the command in the presence of a
voice prompt or other previously recorded audio
message. For example, the system may not hang up
beca~se a DTMF command indicating the end of a message
was obscured. Poor matching of the telephone system and
voice mail system impedance can also affect the ability
of the voice mail system to cancel a previously recorded
message from a message to be recorded.
Yet another problem found in prior voice mail
systems is their inability to hang up once a caller has
finished his message. Most systems ask the caller for a
DTMF command to indicate the message has ended and that
the system may hang up. Many callers fail to enter that
command. Moreover, the command may not successfully
pass through the interference described above. Other
systems include an audio detection circuit as a backup
detection, causing the system to hang up after a
predetermined length of silence. But noise or central
office qenerated call progress tones can often produce
sufficient audio energy to fool the audio detection
L3~4849
circuit. A fail-safe method is sometimes emplo~ed, such
as terminating the call after a predetermined time
unless the caller takes further action such as pressing
a -telephone key. Fail-sae methods are effective but
annoying to a caller and inefficient because it wastes
limited system storage.
Prior systems also include means for indicating
to each user at his telephone the status of the system
and whether he has a message waiting. ~ drawback of
such indicators is that they are limited to the
telephone location and do not reach the person who is
frequently out of his office in a laboratory or common
work area~
SUMMARY OF THE ~NVENTION
An object of the invention, therefore, is to
provide an improved voice mail system.
Another object of the invention is to improve
the abilit~ of a voice mail system to differentiate on
playback between valid DTMF commands entered b~ the user
and previously recorded DTMF signals in the message.
A third object of the invention is to improve
the ability of a voice mail system to detect DTMF
commands generated by a caller and call progress tones
during a telephone connection.
Yet another object of the invention is to
improve the ability of a voice mail system to detect the
end of a message during a recording session.
Still another object of the invention is to
provide a voice mail system that indicates at any
location the status of the users' messages.
The foregoing and other objects, features, and
advantages of the invention will become more apparent
from the following detailed description of a preferred
embodiment which proceeds with reference to the
accompanying drawings.
` ,'
~3~41 !3~9
, ~
- 4
BRIEF DESCRIPTION OF THE DRAWINGS
FIG. 1 is a block Aiagram of a voice mail
system according to the invention.
FIG. 2 is a schematic diagram of the CO
interface circuit in FIG. 1~
FIG. 3 is a schematic diagram of the CO
simulator circuit in FIG. 1.
FIG. 4 is a schematic diagram of the audio
distribution circuit in FIG. 1.
FIG. 5 is a schematic diagram of the audio
processing circuit in FIG. 1.
FIG. 6 is a flowchart of a method of
differentiating between previously recorded DTMF signals
and valid DTMF commands.
FIG. 7 is a flowchart of a method of matching
the voice mail system input impedance to the impedance
of the connected telephone system.
FIG. 8 is a flowchart of a method of detecting
call progress tones to indicate the end o the message.
FIG. g is a schematic cliagram of a message
board circuit according to the invention.
FIG. 10 is a perspective view of the message
board.
FIG. 11 is a flowchart of a method of assigning
one of a number of system applications to incoming calls
to the system.
DETAILED DESCRIPTION
System Overview
With reference now to the drawings, FIG. 1 is a
block diagram of one embodiment of a voice mail system
10 according to the invention. The system shown is a
disk-based digital voice mail system. It should be
understood, however, that the invention as described and
claimed herein is not limited to disk-based systems and
can be applied to magnetic tape-based and other voice
mail systems as well. Several features o this
embodiment are adequately explained in U.S. Patent No.
~30~8~L9
4,747,126.
The common features will be described with re~erence to
that application.
At the left of FIG. 1, a CO interface 12 is
shown that interfaces the voice mail system 10 with one
or more telephone lines 14 to a central telephone
of~ice. Lines 14 from the interface 12 may connect
directly to a central office or, more likely, connect to
a PBX, EKTS, or other system 15 that handles
communication ~etween the central office and internal
lines of a business. Shown immediately above interface
12 is a CO simulator 16 that provides an interface to
the system 10 for one or more telephone lines 18 or
other devices connected to the system 15. Simulator 16,
in effect, mimics an outside telephone line so that
internal access may be provided to the voice mail system
without requiring the local user to dial out and tie up
an outside line.
The interface 12 and simulator 16 each provide
two output signals: an incoming audio signal that is
routed to an audio distribution circuit 20 and a ring
detect/off hook detect signal that is routed to a logic
cell array 22. In the present embodiment, interface 12
and si~ulator 16 each have two audio signal paths so
that a total of four audio paths connect to the audio
distribution circuit 20. The number of signal paths may
be increased as desired~ The circuit 20 is a switching
point for connecting up to two active audio paths, i.e.,
a ;oath with an incoming audio signal present, to the
rest of the system 10. Similarly, four detect signal
paths connect to array 22, which is a software
programmable logic cell array such as a XILINK LCA 2064
that comprises system interrupt and status registers for
computer 24. A detect signal on one of the signal paths
alerts the computer 24 via array 22 that an audio signal
is present on the corresponding audio path to circuit
~0~49
20. The computer 24, in turn, communicates with the
audio distribution circuit 20 via a data bus interface
26 to select the active audio path. The elements o~ the
computer 24 are conventional and may include a
microprocessor such as an 8088, random access memory
(RAM), read only memory (ROM), and a disk drive for
digital data storage of messages, voice prompts, test
information, and other information for a caller or user.
The audio distribution circuit 20 transmits
received audio signals via one of the audio channels to
an audio processing circuit 28. ~oice messages, DTMF
signals, and call progress tones such as a busy signal
and a dial tone can be detected by this circuit. As
described in U.S. Patent No. 4,747,126, the DTMF signals
are decoded for system control purposes such as
recording and playing back a message. The audio
processing circuit 28 also includes means for converting
the analog audio signal into a digital signal for
storage as data on a disk and for reconverting a stored
digital signal into an analog signal for transmission to
a caller. To reduce the disk space required for data
storage, the digital audio signals are compressed via a
conventional speech compression circuit 30 before
transfer via a data bus 32 and interface 26 to disk
storage in the computer 24.
Previously recorded audio signals read from the
disk include voice prompts as well as audio messages and
test information. A previously recorded signal is
reconverted from a digital to audio signal by the
processing circuit 28 and sent to the audio distribution
circuit 20. There, the previously recorded audio signal
is combined with a received audio signal via a 2-to-4
wire converter so that the caller can hear a previously
recorded message and respond by DTMF commands. The
2-to-4 wire converter ideally cancels the previously
recorded audio signal so that the converted output is
~3C~8~9
- 7
just the received audio signal, communicated to the
audio processing circuit 28 ~or recording as a new
message. In system 10, as other voice mail systems, the
caller can end the message by entering a DTMF command.
Each of the above-described circuits of the
system 10 are controlled by the computer 24 through a
conventional bus interface address and control circuit
34. The control circuit 34 communicates computer
commands directly to the affected circuit. For some of
these circuits, such as the data bus 26, the responses
are relayed back to the computer 24 by the control
circuit 34. For other circuits, such as CO simulator 16
or CO interface 12, communication with the computer 24
is effected via array 22 or data bus 26 Communications
among the circuits and the computer 24 which are
relevant to the present invention are described in more
detail in the following section. For a more complete
description of the data bus interface and data bus
requirements, reference should be made to the
manufacturer's specification sheet for the selected
microprocessor.
Circuit Descriptioll
Referring now to FIG. ~, a schematic view of
the CO interface 12 is shown. Two identical interfaces
are illustrated ~or accepting two calls concurrently.
Ring level is detected by the cooperation of DC blocking
capacitor 40, current limiting resistor 42, and ring
level threshold diodes 44. If the AC ring voltage
exceeds the threshold set by diodes 44, current flows
through the opto-isolator 46. The output transistor of
opto-isolator 46 switches on, causing a voltage to be
applied across the resistor 48 and capacitor 50, which
are coupled to the inverting input of a ring/loop
comparator 52. Capacitor 50 charges until it reaches
the threshold voltage on the comparator's noninverting
input and causes the comparator to change state and
, . .... .
~3~48~9
assert the detect signal, active low. A resistor 54
provides positive feedback to the noninverting input for
sharp state transitions. When the ringing ceases,
opto-isolator 46 switches off and ca~acitor 50
discharges to resistor S6 to remove the detect signal.
Resistor 48 and resistor 56 are chosen so that the delay
before asserting the detect signal is shorter than the
delay in removing the ringing signal. As described, the
detect signal is passed to the array 22 to communicate
the presence of a call to the computer 24.
Loop current is detected through an
opto-isolator 58. Detection is similar to that for ring
voltage, except that the voltage for recharging
capacitor 50 is applied across a resistor 60 of much
greater resistance than resistor 48. Consequently, the
applied voltage divides more substantially across
resistors 56 and 60 to increase the delay in asserting
the detect signal and decrease the ultimate voltage on
the capacitor. When the ringing ceases, less time is
thus required to lower the capacitor voltage to below
threshold and remove the detect signal.
The off hook switching function for answering a
ring and hanging up is carried out by the computer 24
via signals from the control circuit 34 through an
optically isolated ~OS switch 62. If ringing is
detected, the off hook signal is asserted as the input
to an inverter 64. The inverter output goes low,
allowing the switch 62 to conduct the incoming audio
signal into the circuit 12. The resulting signal
current switches on opto-isolator 58 as the audio signal
is transmitted across an isolation transformer 66 to an
impedance matching pad comprising resistors 68, 70, 72
coupled to the transformer's secondary winding. The
impedance pad reduces the range of impedance values seen
by the 2-to-~ wire converter within the audio
distribution circuit 20 to improve the signal's transfer.
~131[J~849
~ . ~
g
FIG. 3 is a schematic view of the C0 simulator
16. In order to simulate an outside line, simulator 16
includes a -48 voltage source applied across an isolator
circuit 74 that simulates a large inductor and isolates
a -48 V voltage source. Isolation circuit 74 allows for
DC current flow without affecting the AC impedance at
audio frequencies. The audio signal, powered by the -48
voltage source, is transmitted across a transformer 76
to an impedance matching pad comprising resistors 78,
80, and 82. This impedance pad, similar to the pad in
circuit 12, is coupled to the secondary winding of
transformer 76 to reduce the impedance range seen by the
2-to-4 wire converter in the distribution circuit 20
Current produced hy an incoming call is
detected by an opto-isolator 79 coupled to circuit 74.
The detect signal at the output transistor of
opto-isolator 79 is asserted low and routed to an array
22 for communication to the computer 24.
Received audio signals are transmitted along
signal paths from circuits 12 and 16 to distribution
circuit 20 as shown in FIG. 4. At the left of the
figure, these signal paths encounter a cross point
switch 84 such as a Silicon Systems SSI 78093A/B that
connects any I/0 point on its left side to any I/0 point
on its right side. Conne~tion of the left and right
side points is controlled by computer 24 through a
conventional address register in the data bus 26. As
explained in ths specification for the SSI 78093A/B, the
connection is made by the computer 24 by loading the
appropriate address into the register and then strobing
the switch 84 to latch the address therein. Of the left
side inputs include four cancellation networks 86, 88,
90, and 92 covering a range of impedances. Coupled to
the I/0 points on the right side of switch 84 are a
2-to-4 wire converter 94 and a dialer signal path for
each of the two audio channels. Also coupled thereto is
~3
.
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"J.'"i
~3~4849
-- 10 --
a 500 hertz tone for a busy signal that appears on an
audio signal path ~hen both audio channels are in use.
The dialer signal is generated by the system 10 to
connect the caller with the receptionist and take the
caller out of the voice mail system. It should be noted
that expansion audio signal paths are also connectable
to the left side of the switch 84 if additional circuits
12 and 16 are added to the system 10.
The switch 84 then has three functions. One
function is to provide means for routing active signal
paths from the interface circuit 12 and simulator
circuit 16 to the two audio channels at the output of
distribution circuit 20. This means comprises computer
24 communicating with the switch 84 to make connections
in response to detection of active audio signal paths in
circuits 12 and 16. A second function is to provide
means for passing DTMF signals from the dialer signal
path and the busy signal from the 500 hertz tone to the
audio signal paths. Again, this means comprises
computer 24 communicating with the switch 84 to make
these connections in response to active paths in
circuits 12 and 16. The third function is to provide
means for matching the impedance of the central office
lines by switching cancellation networks 86, 88, 90, and
92 into and out of the 2-to-4 wire converter 94 to
improve its cancellation ability. The means ~or
matching the impedance will be described hereinafter.
As mentioned in the background of this
; invention, one problem with prior voice messaging
systems has been the inability to differentiate between
previously recorded DTMF signals produced by a
previously recorded message and valid DTMF commands
generated by the user. Ideally, the 2-to-4 wire
converter should eliminate the previously recorded audio
signal from the audio signal detected by the tone
detector. However, this often is not the caseO In the
~0484g
present embodiment, the potential for the problem can be
seen with reference to FIGS. 4 and 5, which have a DTMF
transceiver 96 such as a MI~EL ~T 8880 tha-t detects and
transmits DTMF signals. The operation of the
transceiver is more fully discussed hereinafter, but for
the present it should be understood that signals
generated by the DTMF transceiver 96 and transmitted via
data bus 32 will cause the computer 2~ to respond as the
command requires. Such responses may include erasing the
message, jumping over the message, etc. To improve the
differentiation, means are providèd for muting the
previously recorded signal so that computer 24 does not
respond to the signal as a command. An element of that
means is an interrupt request IRQ signal generated by
transceiver 96. The IRQ signal is sent to the computer
24 via the array 22 whenever a DTMF signal is detected
for a predetermined time in the received audio signal
from the distribution circuit 20. This time is chosen
to be sufficiently long so that a transient does not
prompt an IRQ signal in response but is short enough
that a valid DTMF tone produced by a key would not go
undetected. A second element o~ the means is an analog
mute switch 98 through which the transmitted audio
signal passes before crossing the signal level switch
25 100 and appearing at the 2-to-4 wire converter 94. The
switch 98 is controlled by a mute signal generated by a
third element of the meansr the computer 24. The mute
signal is triggered by the computer in response to the
IRQ signal from the transceiver 96.
FI~. 6 is a flowchart of the computerls
response to a detected DTMF signal of sufficient length
to prompt an IRQ signal from the transceiver 96.
Whenever a tone is detected by the computer via the IRQ
signal (step A), the mute signal is asserted (step B) to
open the mute switch 98 so that the transceiver is no
longer receiving the previously recorded audio signal.
~a304 !349
A 45 millisecond delay is counted out (step C) so that
the tone within the transceiver may dissipate. With the
mute switch 98 still open, the computer again checks for
a DTMF signal (step D). If it is still present, then
the computer recognizes the tone as valid (step E)
because the previously recorded audio signal is not
present to produce the DTMF signal. ~ence, the DTMF
signal must be in the received audio signal, generated
by the user. If the DTMF signal is absent, the computer
24 recognizes the source of the signal as a previously
recorded audio signal, ignores it as invalid, and closes
the switch 98 (step F). The computer's response then
reverts to step A until another DTMF signal is then
detected.
Referring now to the cross points switch 84 and
cancellation networks 86-92, these components comprise a
means for compensating for a mismatch of impedances
between the 2-to-4 wire converter 94 and the outside
telephone system represented by the connected telephone
lines 14. As described heretofore, this mismatch
hinders the efectiveness of the converter 94 in
cancelling Previously recorded audio signals that
interfere with valid DTMF commands from the user and
call ~rogress tones. To optimize cancellation, the four
cancellation networks 86-92 (which preferably cover a
preselected range of impedance) are inserted into the
converter circuit to test it over a range and strength
of DTMF signal and other tones. The flowchart of FIG. 7
illustrates the means and method. On power up of system
10, the computer 24 takes a central office line
connected to inferface 12 off hook to generate a dial
tone as the received audio signal (step AA). The
computer then checks its memory to see if all
cancellation networks have been tested ~step BB). With
the method just underway, each network is to be tested
and the computer selects a irst network (step CC). If
~3Q~L~
all tones have not been tested against a selected
network ~step DD), then a test tone is selected from
memory to be generated as a previously recorded audio
signal (step EE). Each tone is applied at a number of
amplitudes (step FF, GG, ~H). Whether an audio signal
apart from the dial tone is detected is then logged
~step II, J~). All test tones are applied at their
different amplitudes to each of the four cancellation
networks/ and the logged results evaluated for the
lowest aggregate audio interference detected (step KK).
The optimum cancellation network is then selected by the
computer 24 and connected to the converter 94 via the
cross point switch 84 (step LL).
With reEerence again to FIG. 5, it can be seen
that the audio processing circuit 28 comprises other
components in addition to the DTMF transceiver 96. The
components include a call progress detector 102, an
audio detection circuit 104, an ATC amplifier 106, and a
bidirectional PCM coder decoder (CODEC) 108.
Transceiver 96 itself requires an input gain
stage to compensate for signal losses in circuits 12 and
16 and in the matching pads. This stage also is
configured to shift the DC bias point from zero volts
provided by the converter 94 to 2.5 volts required by
the transceiver 96. This stage includes an internal
operational amplifier with inputs IN- and IN~ and Tin as
the output, with the amplifier internally connected to
tone detectors. The capacitor 110 is required to block
DC to the amplifier inputs. A resistor 112 is an input
resistor to the amplifier, with resistor 114 of a
greater value selected as a feedback resistor for gain.
DC bias for the amplifier is set by Vref on the
transceiver. Capacitors 116 and 118 act to filter high
frequency noise that may be present at Vref.
On the right side of the transceiver 96 are its
outputs. An RC circuit comprising a resistor 120 and
~L3~ 3~9
capacitor 122 set the received timing to DTMF tones,
i.e., the delay before a tone is recognized by the
transceiver. DTMF signals produced by the transceiver
appear at the TONE output. A resistor 124 is the
required pull down resistor for the output and the
capacitor 126 filters internally generated noise from
the TONE output. An isolation amplifier 128 sets the
gain of the TONE output and removes its 2.5 volt DC
component. The amplifier 128 includes a DC blocking
capacitor 130 and resistors 132 and 134 to set the
desired gain. The amplifier output is the dialer signal
heretofore described.
The internal registers of the transceiver 96
may be read and written to directly by the computer 24
via the data bus 26 and da~a bus 32. Data transfers,
which include the tones the transceiver is to detect,
occur over four data lines D0-D3. The transceiver 96 is
enabled for access whenever the CS line is asserted
low. Actual transfer occurs when the 02 line is
asserted high. The R/W input determines whether the
registers are to be read or written to as derived from a
control register bit via control circuit 34, rather than
Erom the signal on the data bus 32, to improve timing.
The RSO input determines which set o~ internal registers
is to be accessed. The IRQ output, as previously
mentioned, is an interrupt signal to the computer 24.
An interrupt is generated in the computer 24 by the
negative transition of the IRQ signal. `For further
description of how to configure a particular transceiver
96, such as the MITEL MT 8880, reference should be made
to the manufacturer's specification sheet.
The call progress detector 102 such as a
Silicon Systems SSI 980 detects the presence of audio
energy in the call progress bands, i.e., dial tones,
busy signals, and off hook "squeal." The detector can
also be triggered by voice band signals. The capacitor
~3(~ 9
-- 15 --
136 is provided to block the DC component of the
received audio signal, and the resistor 13~3 coupled to a
Vref biases the signal input to the proper DC level.
The presence of call progress energy is signalled by a
5 tone detect signal being asserted high. The tone detect
signal is routed through the array ~2 to the computer
24. Computer interrupts are generated by the positive
and negative transitions oE the tone detect signal.
An AGC amplifier 106 such as a LM 1818 provides
10 automatic gain control and audio amplification to the
received audio signal. The signal is passed through a
resistor 140 that forms a voltage divider with an
internal transistor at pin 5 of the amplifier 106 to set
the proper voltage level. The audio signal continues
15 through a DC blocking capacitor 142 and input resistor
144 to an internal amplifier. Gain is set by a feedbac~
resistor 146. At the output of the internal amplifier,
the audio signal splits between coupling capacitors 148
and 150. The signal through capacitor 148 passes into
20 the amplifier 106 and determines the attenuation of the
internal AGC transistor. The attack and decay times of
the amplifier 106 are set b~ a capacitor 152 and a
resistor 154 connected to the amplifier 106. A
capacitor 156 is coupled to the internal transistor
25 input for filtering high ~requencies. For further
description of how to configure an LM 1818 as amplifier
106, reference should be made to the manufacturer's
specification sheet.
The audio signal that passes through capacitor
30 150 is routed as input to the PCM CODEC 108 such as a
Motorola~C 145503. CODEC 108 is adapted to amplify and
then convert the analog audio signal into pulse code
modulated (P(~M) data for storage on the data disk drive
via the compression circuit 30. Additionally, the CODEC
35 108 is adapted to reconvert PCM data back into an analog
audio signal to be transmitted as a previously recorded
~3~ 9
audio message. Resistors 158 and 160 are input and
feedback resistors, respectively, and capacitor 162
filters high frequency noise which otherwise may cause
aliasing inside the CODEC. The converted digital data
appears serially as the TDD. A sample is converted at
each rising edge of an 8 kHz clock on the TDE input.
The serial digital data is clocked out of the,converter
by a 2.048 MHz clock signal on the TDC input. PCM data
that is reconverted to an analog audio signal is read
1~ into the CODEC 108 at the RDD input from the disk drive
via the compression circuit 30. The serial data is
clocked in by the negative transitions of the 2.04~
megahertz clock signal on the RDC input. The entry of
the PCM data is begun by the 8 kHz clock signal of the
15 RCE input. The reconverted analog signal is then
transmitted through the RxO output to the 2-to-4 wire
converter 94 as shown in FIG. 4.
The amplified audio signal appearing at the
output end of the feedback resistor 160 is also applied
to the audio detection circuit 104 for detection of
audio energy. The amplified audio signal passes through
a hot carrier low threshold rectifier 164 and a audio
detector attack time resistor 166 to charge a capacitor
168. If the capacitor 168 charges to the threshold
voltage set by a resistive network attached to an audio
detect comparator 170, the comparator output goes low to
signal detection of audio energy~ Resistor 172 controls
the discharge rate of capacitor 168 to provide a signal
of sufficient duration. Hysteresis is supplied by
resistor 174 to limit the frequency of audio detect
occurrences. The audio detect signal is passed to the
array 22 as a computer interrupt. The computer 24 is
interrupted by the audio detect signal on its positive
and negative transitions.
One function that transceiver 96, call progress
detector 102, and audio detection circuit 104 provide is
L8~
- 17 -
end of message detection so that the system 10 will hang
up after a caller completes his messaqe. Each circuit
is controlled separately by the computer 24 to detect
different evidence that a call has been terminated. For
example, the transceiver 96 may be employed to detect a
DTMF command that is generated by a caller to indicate a
message is ended. The audio detection circuit 104 may
be employed to detect a predetermined length of an audio
silence to signal that a message is over. In some
circumstances, however, call progress tones or noise
produce enough audio energy that the system 10 continues
to record even though the caller is no longer speaking.
To improve end of message detection, the computer 24 is
programmed to monitor call progress tones detected by
the detector 102. The flowchart of FIG. 8 illustrates
the computer's method. The system is in the record mode
recording a message. Concurrently, the computer 24 is
monitoring the interrupt generated by the detector 102
(step A'~. If no call progress tone is present, the
system continues to record. Once a progress tone is
detected, however, an internal counter is set to a
predetermined count, say 10 seconds (step B'). The
progress tone is then monitored for the length of the
co~nt (steps C', D'). If the tone is 10 seconds in
duration, the computer 24 determines that the message
has ended and directs interface 12 or simulator 16 to
hang up (step E'). If, on the other hand, the tone
disappears before 10 seconds pass, then the tone is
determined to be a transient signal and is ignored.
30 Recording then continues until a call progress tone of
sufficient duration is detected or one of the other
means for detecting the end of a message terminates the
call.
As explained in U.S. Patent No. 4,747,126, a
voice mail system such as system 10 can include a
message waiting indicator for indicating the status of
~L3~8~
- 18 -
the system and whether a user has a waitlng message.
Typically, these indicators include LEDs that are
mounted to the user's telephone. For those users who
are frequently out of the ofice, however, the LED is
5 not visible and the message then may not be heard Eor
~uite some time. An alternative for notifyinq these
persons is a message board 176 shown schematically in
FIG. 9 and in perspective in FIG. 10. The board 176
contains a field of 24 red LED lights, one for each
10 individual mailbox in the system. An amber in-use LED
provides the system status information. Labels 173 for
each user are provided. The board 176 is sized to be
readily visible from a distance of 50 feet. Several
boards 176 can be linked together to provide message
15 notification throughout a company. The board can be
installed at a message center near workbenches, lamps,
etc., or centrally located wall. The operation of the
components shown in FIG. 9 is amply described in U.S.
Patent No. 4,747,126 where the components are used for
20 driving the individual LEDs located on the users'
telephones.
For the system 10, the message board 176 is
coupled to AUX ports 180 of the data bus interface 126
in FIG. 1. This connection allows the computer 24 to
25 control the message board 176.
The system 10 is not limited by its structure
to only a voice mail system. Because of software
control over the circuits therein, the system 10 may
also be adapted for further applications such as
30 providing specific information (weather, stock values,
etc.) or as an auto attendant (answering incoming calls
with a greeting and instructions to the caller for
reaching a desired person or department in the
company). In the present embodiment, the application is
35 determined by the number called. FIG. 11 is a flowchart
of the computer's method for determining the
4~3~9
-- 19 -
application. Initially, the computer waits for a call
(step AA'). An incoming call is dekected at the
interface 12 or simulator 16 (step BB'). The computer
responds by answering the call (step CC'). ~ach port of
the interface 12 and simulator 16 has an identifying
number which is read by the computer when it answers the
port (step DD'). With this number as the address to
look up table in memory, the computer determines the
application assigned to the port tstep EE'). The
application, such as auto attendant, is then e~ecuted
for that port (step FF') and the computer turns again to
monitoring incoming calls (step GG'). As an alternative
to dedicated ports for applicationsr the system 10 could
be adapted to query the caller for the desired
application and then assign that application to the
caller's port.
Having illustrated and described the Principles
of the invention in a preferred embodiment, it should be
apparent to those skilled in the art that the invention
can be modified in arrangement and detail without
departing from such principles. We claim all
modifications coming within the spirit and scope of the
following claims.
.
