Phonetic voice activated dialing

A telephone communications system Advanced Intelligent Network (AIN) platform provides a voice activated call dialing functionality through speaker independent phoneme speech recognition having a minimum volume of storage without requiring user template training. Speaker independent phoneme recognition identifies phoneme strings of caller spoken utterances which are then compared to phoneme string representations that previously have been stored in respective caller processing records (CPRs) for those subscribers listed in the ISCP database, or stored in an equivalent peripheral database with which the ISCP can communicate. Each stored phoneme string representation is associated in the CPR with a destination telephone number that may then be extracted to route a call.

TECHNICAL FIELD 
The present invention relates generally to communications networks that 
provide voice activated dialing and more particularly to the use of 
speaker independent phoneme recognition to determine call routing. 
BACKGROUND ART 
The relatively recent development of new and expanded telecommunication 
services has provided subscribers increased flexibility in the selection 
and use of the various features that have become available. These services 
are amenable to being tailored to specific requirements of the subscriber. 
So-called "flash hook services," such as Call Waiting, Tone Block, 3-Way 
Calling, Call Transfer and Consultation Hold, are implemented using 
appropriate switch hook depression by the user. Other services, such as 
Return Call, Answer Call, Repeat Call, Priority Call, Call Trace, Per Call 
Blocking, Intercom Extra, Home Intercom, Speed Calling and Call Block, 
require various combinations of keyed DTMF inputs by the subscriber. Such 
inputs, for example, may be codes including special keys such as the * key 
in combination with a preset number sequence or dialing a dialed system 
telephone followed by further keyed input for purposes of identification 
or choice of options. 
Flash hook services, while offering a wide range of communication 
customization, are particularly complex. Traditionally, depression of the 
switch hook disconnects a call. However, momentary switch hook depression 
with the newer flash hook services effects different results, such as 
connecting new callers. Without prompts and feedback from the system, 
inexperienced users tend to lose their ability to track their location 
while adding, transferring or dropping a caller in a multiple call, flash 
hook operation. Confidence is low in the ability to complete the service 
and the user often expects to lose the connection to the other party. 
In addition to such disadvantages, the user of flash hook operations must 
be cognizant of the correct DTMF key combinations for each of the various 
services, the appropriate sequences for inputting key combinations in 
various complex services, and the appropriate responses from the 
communications network that either signal the next step in the process or 
verify completion of the process. A burden is placed on the subscriber to 
remember the appropriate activation and deactivation codes for the various 
subscribed services. Flash hook operations can be not only complex but 
time consuming. 
These drawbacks extend to preprogrammable functions included in a 
subscriber's telephone equipment as well as those provided by the 
telecommunications system. For example, speed dial features that the user 
may set up when first obtained may be abandoned later when instructions 
are not at hand due to the complexity of the entry process. As a result, 
speed dial keys may not be fully populated or may include obsolete 
entries. 
More recently, Common Channel Signaling has been utilized advantageously by 
the Advanced Intelligent Network (AIN) of the public switched telephone 
system to predefine services according to the subscriber's requirements 
and to implement such services for applicable calls. A description of an 
Advanced Intelligent Network (AIN) implementation may be found, for 
example, in U.S. Pat. No. 5,247,571 to Kay et al. Each central office of a 
network of interconnected central offices is connected to a number of 
local telephone lines constituting a specified group. Call routing is 
carried out in accord with data stored in the AIN database and with 
customer specified parameters, such as calling/called party number, 
time-of-day, day of the week, authorization codes, etc. After the central 
office switching system detects an off-hook, it determines whether or not 
the call originates from a subscribing line. If not, the system receives 
dialed digits and executes normal call processing routines. If the call is 
from a subscriber line, the originating office receives dialed digits, 
suspends the call and sends a query message to the Integrated Service 
Control Point (ISCP) through the Signaling Transfer Points (STP's). This 
query message, in Transaction Capabilities Applications Protocol (TCAP) 
format, identifies the calling station and the digits dialed as well as 
other pertinent information. Based on the identity of calling party's 
address, the ISCP retrieves from its database a table of trunk group 
routing information. The ISCP formulates a response message, again in TCAP 
format, including the routing information, and transmits the response 
message back to the originating central office via the STP(s). The system 
then executes normal call processing routines for completing the call 
using the received routing information provided by the ISCP. 
The use of AIN reduces the number of DTMF entries that a subscriber must 
input as much of the information needed for providing the service has been 
stored in the AIN database. For those services that require a significant 
amount of caller input, interactive voice menus are used to prompt callers 
in a user friendly manner. Nevertheless, inherent drawbacks exist in 
situations in which the caller must provide DTMF input. The subscriber 
often finds it difficult to remember the proper DTMF representations of 
the required input, such as a multiplicity of telephone numbers and codes, 
and may be inconvenienced by the time and steps necessary to follow a menu 
driven procedure in order to complete the desired service. 
The use of speech recognition is an attractive approach to alleviate such 
annoyances. As the development of commercially available speech 
recognition systems has progressed, voice responsive features have been 
provided in telephone services. Prior examples of telephone devices that 
are responsive to caller voice input to dial a call to a corresponding 
destination are U.S. Pat. No. 4,928,302, issued to Kaneuchi et al., and 
U.S. Pat. No. 4,961,211, issued to Marui et al. The Marui et al. device is 
a mobile telephone apparatus that makes an outgoing call in response to 
the caller speaking a number that corresponds to the destination telephone 
number. The telephone number is read out from stored telephone numbers and 
is then dialed. When a number has been identified, it is synthesized and 
displayed so that the user can determine if it is the correct number. In 
the Kaneuchi et al. device, standard patterns are associated with 
registered telephone numbers. 
U.S. Pat. No. 5,165,095, issued to Borcherding, and U.S. Pat. No. 
5,369,685, issued to Kero, disclose voice activated dialing systems in 
which remote databases are referenced. In the Borcherding arrangement, a 
local database contains speaker independent voice recognition templates 
for various command functions and a remote database in which speaker 
dependent templates are stored. The latter templates represent phrases 
associated with destination telephone numbers. If a dial command is spoken 
by a caller, a local database containing speaker independent speech 
recognition templates is accessed. The templates of this local database 
are compared to a dial command so that dialing instructions can be 
recognized and executed. The caller is identified and speaker dependent 
templates for the identified caller are downloaded from the remote 
database. The speaker dependent templates are then accessed. A spoken 
destination identifier is compared with the speaker dependent templates 
and when a match is found, the destination telephone number is dialed. 
In the Kero arrangement, a voice activated telephone directory and call 
placement system accessible over a telecommunications network allows a 
caller to store a personalized telephone directory and to retrieve 
selected directory listings therefrom by speaking a series of voice 
entries. A plurality of subdirectories are formed to complete the 
listings. A call-spoken entry received over the network is compared with a 
previously stored voice template of the caller speaking the name of a 
subdirectory that is included as part of the caller's personalized 
telephone directory. If a match with a subdirectory name template is made, 
a subsequent caller-spoken entry received over the network is compared to 
a voice template of listings in the subdirectory. The system retrieves the 
destination telephone number associated with the directory listing if a 
match is found and the call may then be completed. Each subdirectory may 
include subordinate levels of subsidiary directories, each having a 
plurality of listings. 
Speech responsive dialing systems such as those of the prior art 
exemplified above have inherent limitations. The large storage required 
for templates of either speaker dependent recognition or speaker 
independent recognition vocabularies is a restrictive factor as the number 
of users and the vocabulary size increase. 
Development of speaker independent templates involves, for each vocabulary 
word, the input from many diverse speakers in order to provide reliably 
accurate recognition. Such templates occupy a large volume of storage. As 
recognition must accommodate speakers of different accents, inflections, 
and pronunciation, the size of the word vocabulary must be limited to 
avoid confusion among similar words. A small number of words may be 
recognized with confidence, while a large number would give an 
unacceptably erratic response. In addition, provision must be made in the 
system to distinguish between use by different callers of the same word, 
for example "mom," for different destinations. 
Speaker dependent recognition requires developing templates for each user. 
While these templates individually would occupy less storage volume than 
speaker independent templates for corresponding words, templates must be 
trained and stored for each word to be used by each user. Users in the 
same household who would use the same vocabulary word for the same 
destination number nevertheless would be required to go through a template 
training process. Moreover, in order to access the appropriate templates, 
provision must be made in the system for identifying the particular user. 
DISCLOSURE OF THE INVENTION 
Accordingly, an advantage of the present invention is a voice activated 
dialing service that provides to a subscriber a user friendly environment 
requiring little user input in establishment of the service. 
Other advantages of the invention are that speaker independent phoneme 
speech recognition, as used in the invention, requires a minimum volume of 
storage while eliminating the need for user template training. As the 
amount of storage is substantially reduced compared to prior art 
arrangements, the number of subscribers need not be limited. 
These and other advantages of the invention are satisfied, at least in 
part, through the use of the Advanced Intelligent Network (AIN) Platform. 
A detailed description of an AIN system, suitable for implementation in 
connection with the present invention, is provided in the aforementioned 
U.S. Pat. No. 5,247,571, the disclosure of which is incorporated herein by 
reference. 
The AIN conventionally provides services based on feature logic and data 
located at a centralized node in the network known as a Service Control 
Point (SCP) or as an Integrated Service Control Point (ISCP). Network 
switches appropriately equipped, known as Service Switching Points 
(SSP's), communicate with the ISCP and together provide various AIN 
services. The SSP determines which calls require AIN service based on 
characteristics of the call, such as the line it originated from or the 
digits that were dialed. 
The process of identifying calls that require AIN processing is known as 
"triggering," since a particular characteristic of the call "triggers" a 
switch to provide AIN treatment. Once a trigger occurs, a query message is 
sent to the ISCP asking for instructions. Based on information contained 
in the query message, the ISCP determines which service is being requested 
and provides appropriate information such as routing and billing 
instructions that the SSP then executes to complete the call. ISCP, 
through a lookup in its database, determines which service is being 
performed on a particular call. The SSP simply identifies calls that 
require AIN processing and executes instructions provided by the ISCP. 
Current program controlled switches such as the AT&T 5ESS and 1AESS and 
comparable switches from other manufacturers are provided with an Advanced 
Services Platform (ASP) that provides SSP and Network Access Point (NAP) 
capabilities. ASP provides services independent triggering and call 
processing capabilities and also supports Operations, Administration and 
Maintenance (OA&M). These capabilities interact with many existing switch 
based features. SSP capabilities enable end offices and access tandem 
offices to interface with SCP databases using Common Channel Signaling 7 
(CCS7) Transaction Capabilities Application Part (TCAP) protocol to 
implement services. These services include standard equal access 
multi-frequency (EAMF) and CCS7-ISDN user part (ISUP) interfaces to a 
network access point (NAP) switch, standard CCS7-TCAP interfaces to an SCP 
database, call processing triggers, non-call processing triggers such as 
test queries, customized announcements under the control of an ISCP, such 
as terminating announcement or play announcement and collect digits, 
connection control under control of the ISCP, business and residence 
custom services (BRCS) interworking, new terminating restrictions, ISDN 
interworking, notification of call termination (returned to ISCP), 
enhancements for OA&M, and billing under control of the SCP. Signaling in 
LATA switching systems is described in detail in Bell Communications 
Research Technical Reference TR-TSY-000506, July 1987. Further details are 
provided in AT&T 235-190-125 October, 1990. 
The present invention employs speaker independent phoneme recognition to 
identify phoneme strings in caller spoken utterances and compares the 
recognized strings to those that previously have been stored in respective 
caller processing records (CPRs) for subscribers listed in the ISCP 
database, or stored in an equivalent peripheral database with which the 
ISCP can communicate. Each stored phoneme string representation is 
associated in the CPR with a destination telephone number that may then be 
extracted to route a call. Alternatively, the phoneme stored in the 
peripheral database may be converted to a destination telephone number 
that is then transmitted to the ISCP to route the call. 
Phonemes are the distinctive elements that a language can combine to form 
different words. Any language utilizes a comparatively small set, often 
less than fifty, of phonemes from which words can be built. Any two 
phonemes, by definition, contrast with each other in the sense that they 
can be distinguished from each other, and thereby used to distinguish 
words. To represent phonemes unambiguously, standard systems of phonemic 
transcription have been established. While it is possible to represent 
each phoneme with a number, or an invented symbol of unique shape, 
standard phoneme tables adopt familiar letter shapes, adding a few more 
characters where necessary. Reference is made to "Electronic Speech 
Recognition, Techniques, Technology & Applications," by Bristow, 
McGraw-Hill Book Company, 1986, more particularly at pages 26 and 27, for 
exemplification of a standard phoneme table. 
The present invention takes advantage of commercially available automatic 
speech recognizers (ASR's), which are essentially acoustic pattern 
recognizers, to use speaker independent representations of each of the 
relatively few phonemes in a feasible manner. This use is in contrast to a 
speaker dependent device that will only recognize correctly the utterances 
of one individual at a time. With speaker dependent technology, such 
individual will have `trained` the device by previously supplying it with 
acoustic reference patterns in his/her own voice. 
In the present invention, the speaker independent phoneme patterns will 
have been trained by the manufacturer, with data representing a composite 
reference voice derived from a number of different voices. Phonetic models 
are thus provided without the need for each subscriber to go through a 
process of training the recognition device. The subscriber's voice is used 
only for recognition. In use, the ASR will recognize the phonemes of which 
the speaker utterances are comprised and translate the recognized phonemes 
into character representations such as those of the standard phoneme 
table. Each character is in the order of eight bytes of storage. As 
phonemes are distinguishable from each other, an acceptable rate of 
recognition accuracy is provided while minimizing required storage volume. 
Storage of phoneme strings for voice activated dialing, with associated 
destination numbers, may take place at the outset of the service or 
thereafter. A list of several entries can be developed by one or more 
callers at a subscriber line in a single session. Thereafter, the 
associated central switching office is set to recognize, upon an off-hook 
condition of the subscriber line, that outgoing calls can be processed by 
voice activation or in the standard manner per DTMF or pulse input. If a 
voiced utterance does not match the stored phoneme strings, the caller may 
be prompted for a destination telephone number and be given an option to 
add the phoneme string with associated telephone number to the CPR or 
peripheral database in addition to completing the call. 
When a call is to be made from the subscriber line, an off-hook condition 
can trigger the central office to switch the call over a voice line or a 
T1 line to a platform that will recognize dialed digits or a spoken phrase 
and, in the latter instance, determine the corresponding telephone number. 
The destination telephone number is transmitted through the ISCP back to 
the switch, which then will route the call accordingly. 
In a preferred embodiment, the central office switching facility includes 
speech independent recognition capability as well as the standard 
universal tone receiver (UTR). Outgoing calls dialed in typical manner, 
such as by DTMF or pulse input, are processed conventionally. Voice 
dialing is recognized by an automatic speech recognizer in the central 
office or on an AIN/IP platform. This event triggers the AIN network 
wherein the recognized phonemes of the spoken utterance is digitized and 
transmitted in a message over a common channel signaling path by the 
central office to the ISCP. The ISCP identifies the calling subscriber 
line from information contained in the message, such as calling party 
Automatic Number Identification (ANI), to access the appropriate CPR. The 
call destination information is thereby obtained and transmitted to the 
central office through the common channel signaling network to route the 
call. The ISCP generates any necessary instructions for interactivity with 
the caller if the recognized phonemes do not match the phoneme strings 
stored in the CPR. In the peripheral database embodiment, recognized 
phonemes of the spoken utterance are processed and translated into a 
destination telephone number which is transmitted to the ISCP for call 
routing. 
Additional advantages of the present invention will become readily apparent 
to those skilled in this art from the following detailed description, 
wherein only the preferred embodiment of the invention is shown and 
described, simply by way of illustration of the best mode contemplated of 
carrying out the invention. As will be realized, the invention is capable 
of other and different embodiments, and its several details are capable of 
modifications in various obvious respects, all without departing from the 
invention. Accordingly, the drawings and description are to be regarded as 
illustrative in nature, and not as restrictive.

BEST MODE FOR CARRYING OUT THE INVENTION 
One system for providing a Common Channel Signaling Network (CCSN) utilizes 
Signaling System 7 (SS7) protocol in a Packet Switched Data Network (PSDN) 
connecting Network Elements (NE) via packet switched 56 Kb digital data 
circuits. In addition to providing call set signaling functions, the SS7 
network also provides access to switching control points (SCP) used to 
permit line identification database (LIDB) look-up for 800 services. Class 
services also use the SS7 network to provide custom call features. The 
latest services using the SS7 network comprise Advanced Intelligent 
Network (AIN) services. AIN services use the SS7 network to access an 
Integrated Switching Control Point (ISCP) where AIN service functions are 
performed. 
FIG. 1 is a diagram of a common channeling signaling network using SS7 
protocol. Common channel signaling uses an out of band signaling path that 
is separate from the path used for voice transmission. This signalling 
technology provides for faster call set-up times and a more efficient use 
of the voice network than prior manual signaling, dial pulse signaling or 
multi-frequency signaling schemes wherein the trunk connecting the calling 
and the called subscribers required both signaling and voice transmission 
over the same circuitry. When a call is placed, the voice communication is 
suspended while signaling is transmitted through the common channel 
signaling network to check whether the line at the destination switch is 
busy and to determine the voice connection path. 
Local telephone lines are connected by individual telephone stations 10 in 
each geographic area to a Service Switching Point (SSP) which may be 
included in the closest CO. Each CO connects via trunk circuits to one or 
more of the other COs, and each CO has a CCIS data link to a Switching 
Transfer Point (STP). Redundant STPs are provided for backup reliability. 
The trunk circuits carry large numbers of telephone calls between the 
CO's. 
Control logic and feature data are located at a centralized node in the 
network called a Service Control Point (SCP). SSPs communicate with the 
SCP through the associated STP. B-link or D-link lines interconnect STPs, 
while A-link lines interconnect the STPs with either SCPs or SSPs. 
If a call requires a feature service such as call redirection, an SSP is 
triggered to communicate with an SCP on the basis of the call 
characteristics, such as originating line or dialed digits. If a trigger 
occurs, a query message is sent to the SCP to obtain instructions. The 
SCP, if provided with appropriate database storage and processing 
capability, can determine the nature of the service and information 
appropriate to routing of the call. Redirection of the call can be 
signaled through the STP(s) to seize a trunk circuit between the 
originating CO and the redirected destination CO. 
All of the CO's 11, 13, 15 and 17 in the illustrated embodiment are 
equipped and programmed to serve as SSPs. Such central office switching 
systems typically consist of a programmable digital switch with CCIS 
communications capabilities. One example of such a switch is a 5ESS type 
switch manufactured by AT&T. Other vendors, such as Northern Telecom and 
Siemens, manufacture comparable digital switches. SSPs are appropriately 
equipped programmable switches present in the telephone network, which 
recognize AIN type calls, launch queries to the ISCP and receive commands 
and data from the ISCP to further process the AIN calls. In instances in 
which the SSP functionality is not present in the CO, end offices without 
such functionality forward calls to an SSP at its prescribed point in the 
network. 
The SSPs 11 and 13 connect to a first local area STP 23, and the SSPs 15 
and 17 connect to a second local area STP 25. The connections to the STPs 
are for signalling purposes. As indicated by the circles between SSPs, 
within the local areas of STPs 23 and 25, each local area STP can connect 
to a large number of SSPs. The central offices or SSPs are interconnected 
to each other by trunk circuits for carrying telephone services. 
The local area STPs 23 and 25, and any number of other such local area 
STPs, communicate with an STP 31 associated directly with an ISCP 40 to 
serve the entire area. The STP hierarchy can be expanded or contracted to 
as many levels as needed to serve appropriately subscriber demand. The 
links between the COs and the local area STPs are dedicated CCIS links, 
typically SS7 type interoffice data communication channels. The local area 
STPs are in turn connected to each other and to the regional STP 31 via a 
packet switched network. The regional STP 31 also communicates with the 
ISCP 40 via a packet switched network. 
The ISCP 40 is an integrated system. Among other system components, the 
ISCP 40 includes a Service Management System (SMS) 41, a Data and 
Reporting System (DRS) 45 and the actual database or Service Control Point 
(SCP) 43. The ISCP also typically includes a terminal subsystem referred 
to as a Service Creation Environment or SCE (not shown) for programming 
the data base in the SCP 43 for the services subscribed to by each 
individual business customer. Separate communication lines 26 and 28, 
respectively, are connected between SMS 41 and an external processing 
system 22 and between SMS 41 and an SSP shown, for example, as SSP 11. 
The messages transmitted between the SSPs and the ISCP are all formatted in 
accord with the Transaction Capabilities Applications Protocol (TCAP). The 
TCAP protocol provides standardized formats for various query and response 
messages. Each query and response includes data fields for a variety of 
different pieces of information relating to the current call. An initial 
TCAP query from an SSP includes, among other data, a "Service Key" which 
is the calling party's address and digits representing the called party 
address. The TCAP specifies a number of additional message formats, for 
example a format for a subsequent query from the SSP, and formats for 
"INVOKE" responses for instructing the SSP to play an announcement or to 
play an announcement and collect digits. For a detailed description of 
signaling in LATA switching systems reference is made to the 
aforementioned Bell Communications Research Technical Reference 
TR-TSY-000506. 
Each central office switching system normally responds to a service request 
on a local communication line connected thereto to selectively connect the 
requesting line to another selected local communication line. The 
connection can be made locally through only the connected central office 
switching system if the originating location and the destination location 
are served by the same central office switch. When the called line 
connects to a distant station, the connection is made, for example, 
through the connected central office switching system SSP 11 and at least 
one other central office switching system SSP 13 through the telephone 
trunks interconnecting the two COs. 
The network includes interactive voice response (IVR) capability provided 
either within a central office or externally thereto, the latter 
illustratively shown as IVR 29. Additionally, external adjunct processing 
capability may be provided for a central office as illustrated by the 
connection 27 between SSP 17 and processor 24. This processor is known as 
an intelligent peripheral (IP) device. Direct communication between the IP 
and the ISCP can be provided. 
FIG. 2 is a simplified block diagram of a central office facility such as 
SSP 11 in accordance with a preferred embodiment of the invention wherein 
speech recognition functionality is included within the SSP. Subscriber 
stations 10 are connected by subscriber lines 8 to line interface circuit 
52. This circuit connects all subscriber lines to switching network 54 
whereby calls are completed through appropriate communication paths 
between local subscriber lines or between local lines and trunks through 
trunk circuit interface 56. Control circuit 58 provides various service 
and supervisory functions that are required for normal operation. Broadly 
included therein are conventional elements, for example, dial tone 
generators, busy tone generators, ringback tone generators, and various 
announcement platforms. 
Connected in parallel to the line circuit interface are automatic speech 
recognizer (ASR) 60 and universal tone receiver (UTR) 62. ASR may comprise 
any of the commercially available recognizers, such as described in the 
aforementioned Bristow publication, at pages 216-233, or at pages 236-242 
of Volume 17 of McGraw-Hill Encyclopedia of Science and Technology, 
seventh edition. UTR 62 is a conventional tone recognizer that registers 
dialed digits. 
Operation of the voice activated dialing arrangement embodied in FIGS. 1 
and 2 is described with respect to the flow charts shown in FIGS. 3A and 
3B. Upon going off hook, the subscriber line condition is sensed at the 
central office at step 100. The calling subscriber line is identified at 
step 102. At step 104, determination is made whether input has been 
received from the subscriber line at the central office. If so, step 106 
determines whether voice has been detected by the ASR. If voice has not 
been detected the UTR collects received digits at step 108 and the call is 
then routed to the dialed destination at step 110 in conventional manner. 
If voice detection occurs in step 106, the ASR recognizes and digitizes the 
phonemes of the spoken utterance at step 112. The SSP then formulates a 
digital message that includes calling line identification as well as the 
digitized phoneme string, at step 114, and transmits the message over a 
common channel signaling path to the ISCP. The ISCP accesses the CPR that 
corresponds to the identified calling line at step 116 and compares the 
received digitized phoneme string with the phoneme strings stored in the 
accessed CPR, at step 118. If a match is determined in step 120, the 
destination telephone number is retrieved from the CPR at step 122 and 
transmitted in an SS7 message back to the SSP at step 124. The call is 
then routed to the destination. 
If there has been no phoneme string match in step 120, an announcement is 
transmitted from an IVR platform to the caller at step 126. The 
announcement indicates that the voiced phrase is not currently in the CPR 
list and prompts the caller to input the call destination number. Either 
DTMF, speech or pulse signals can be handled. The caller is also prompted 
to indicate whether the entry is to be added to the CPR list and whether 
the call is to be routed to the destination. 
Dialed digits for the call are collected by the UTR register at step 128. 
If it is determined at step 130 that the caller does not want to add the 
phrase and number as an additional entry to the CPR listing, the call is 
then routed to completion. If the entry is to be added, the ISCP updates 
the CPR accordingly at step 132. At step 134 it is then determined whether 
the call is to be completed to the identified destination, in addition to 
the CPR update function, or whether the current call was only for the 
purpose of adding the entry to the CPR. If the call is not to be 
completed, it is terminated in step 136. Otherwise, the call is routed to 
the destination. 
As can be seen from the above described operation, the arrangement provides 
the calling subscriber with flexibility and simplicity of use. With any 
call, the subscriber has the options of simply voice dialing to a 
destination that currently exists in the CPR list, or adding a new entry 
to the list, with or without completing the call. The caller merely utters 
a destination phrase and, only in the case of a new entry, enter by DTMF 
keys the destination number and option responses. As a further 
convenience, the prompt announcement may give the option of adding 
additional entries so that a single call can be used as a session for 
populating the CPR list. 
In an alternative embodiment, the database may be stored in an intelligent 
peripheral device (IP) at a location remote from the ISCP and SSPs. In 
operation, the ISCP would access the IP either directly through a common 
channel signaling path or through the SSP. 
FIG. 4 is a simplified block diagram of another alternative embodiment. A 
remote platform 65, linked to SSP 11 and other network SSPs by a voice 
line or T1 line, includes ASR 60, UTR 62 and processor 66. The processor 
comprises storage that accommodates listing of phoneme strings with 
associated destination telephone numbers for each subscriber line. For 
each position 10 that subscribes to the voice activated dialing service, 
an off-hook trigger is set at the associated central office SSP 11. Upon 
sensing off-hook initiation on the line, the SSP issues a TCAP query 
message, including calling line identification, to the ISCP. Through 
access to its database, the ISCP determines that the call is a voice 
activated dialing subscriber call and returns a message to the SSP to 
switch the line to connect with platform 65. As previously described, UTR 
62 will receive and register dialed digits and ASR 60 will recognize a 
spoken utterance and produce a digitized string. Under control of 
processor 66, if dialed digits are received, the destination number 
information will be transmitted to the SSP for switching the calling line 
and routing to the identified destination. If, instead, a digitized 
phoneme string is produced by ASR 60, the string is compared with phoneme 
strings stored in the database of processor 66 for the calling subscriber 
line. If a match occurs, the destination number is identified and signaled 
to the SSP for routing the call to completion. If a match does not occur, 
similar options as those of the preferred embodiment are provided to the 
caller in an announcement message handled by an IVR at the platform. 
In this disclosure there is shown and described the preferred embodiment of 
the invention and but a few examples of its versatility. The following 
advantages result from the described invention. The end user can control 
speaker independent programming of his or her own list. Names added in 
this manner can be utilized by all users of the telephone since the 
technology is speaker independent. Recognition for an individual line can 
be enhanced through prompting a caller to add a new destination number or 
create a synonym name for an existing destination number. Overall system 
performance is improved compared to existing systems that utilize 
combinations of speaker dependent and speaker independent technologies. 
Such systems are prone to difficulty resulting from complexities necessary 
to determine which recognition technology to use at particular stages of a 
call. In contrast, the invention uses a single speaker independent 
recognition technology that acts upon phoneme units rather word or phrase 
vocabularies. It is to be understood that the invention is capable of use 
in various other combinations and environments and is capable of changes 
or modifications within the scope of the inventive concept as expressed 
herein.