Analog terminal internet access

A system and method of effecting transfer of a message such as a voice message from one multipurpose, multimode centralized messaging system in a first switched telephone network to a multipurpose, multimode centralized messaging system in a remote second switched telephone network, wherein each of the telephone networks includes central offices connected to subscriber terminals. The message is inputted in analog or digital, voice or text form by one of the terminals connected to the first telephone network, selectably processed in the messaging system and stored in a selectable digital form in the messaging system in that network. The message is then transferred from that messaging system to the centralized messaging system in the second telephone network where it is selectably processed and stored. The message is delivered by an outgoing call to an addressee terminal in the second telephone network. The transfer between telephone networks and their respective centralized messaging services occurs via the Internet. The transfer occurs through connectionless packet signaling using TCP/IP protocol.

TECHNICAL FIELD
 The present invention relates generally to switched communications networks
 and providing voice mail services and more particularly relates to a
 system and method for providing communication between voice mailboxes in
 multiple mailbox systems using connectionless packet delivery via
 established networking arrangements.
 BACKGROUND ART
 Voice mail has become commonplace not only in business usage but also on an
 individual telephone service subscriber basis through Centrex service from
 a central office. A voice mail system is a specialized computer that
 stores messages in digital form on a fixed disk. The voice is generally
 digitized, usually at a much slower rate than the 64 Kb/s signal the
 central office uses in its switching network. The digitized voice is
 compressed and stored on a hard disk that maintains the voice mail
 operating system, system prompts, and greetings, and the messages
 themselves. A processor controls the compressing, storing, retrieving,
 forwarding and purging of files. A form of early systems is described in
 Matthews et al. U.S. Pat. No. 4,371,752 (hereinafter the Matthews '752
 patent), issued in February, 1983, and several related patents.
 U.S. Pat. No. 4,585,906 (hereinafter the Matthews '906 patent), issued Apr.
 29, 1986 to Gordon H. Matthews et al. The Matthews '906 patent is a
 continuation-in-part of the Matthews '752 patent.
 U.S. Pat. No. 4,602,129 (hereinafter the Matthews '129 patent), issued Jul.
 22, 1986 to Gordon H. Matthews et al. The Matthews '129 patent is a
 continuation-in-part of the '752 Matthews patent.
 The three Matthews patents each describe a voice mailbox type system using
 digital storage and programmed control to offer a wide variety of message
 storage, forwarding and delivery type services. The system architecture is
 essentially the same in each patent disclosure. With reference to FIG. 3
 of the '752 patent, the voice message system (VMS) 10 includes an
 administrative subsystem 60, a number of call processor subsystems shown
 as 62A-62C, and a digital data storage subsystem 64.
 The call processor subsystems each include a microprocessor based single
 board computer 70, a memory 72 having for example 64K of RAM, a
 communication port interface 74, two disk adapters 76, 78 communicating
 with the storage subsystem via two storage buses and a block transfer
 interface 80 which communicates with the administrative subsystem (FIG.
 4). The communication port interface 74 provides communication to and from
 the telephone lines via communication port driver modules 90A-B, each of
 which includes port drivers 92, CODECS 96 and voice connection
 arrangements 98 (FIG. 5). As shown in FIG. 6 of the '752 patent, the
 administrative subsystem 60 includes a microprocessor based single board
 computer 100, a memory 110, a non-volatile memory 112, two disk adapters
 114, 116 communicating with the storage subsystem via two storage buses
 and a block transfer interface 118 which communicates with the call
 processor subsystems.
 In the '752 system, a message router program is informed of each occurrence
 of a new incoming message stored in the system. This program creates a
 message control block on disc for each message, and the message is thereby
 queued to each of the addresses selected by the person sending the message
 (see the '752 patent, Column 29, lines 5-16). As disclosed in the '752
 patent, to deposit a message (FIG. 11), a user calls the VMS. The VMS
 answers the call and transmits an initial prompt message to the caller.
 The caller then inputs a unique authorization code identifying that person
 as a subscriber to the VMS service. Upon receipt of a valid authorization
 code, the VMS transmits a short progress tone and accepts an series of
 dialed digits representing an address input from the subscriber.
 Typically, an address is a single telephone number. The '752 system also
 offers the subscriber the option to select a previously established
 distribution list including a number of such addresses. After entry of all
 necessary address information identifying one or more destinations, the
 user inputs a "1" to initiate voice recording and then transmits a voice
 message. The VMS stores the voice message in one of the digital disc
 storage units 120 within the data storage subsystem 64 (FIG. 7). The user
 is then given the option to deposit another message, inquire about
 messages stored for the subscriber or terminate the session by hanging up.
 To retrieve and replay stored messages, a subscriber initiates a routine
 referred to in the '752 patent as the INQUIRY feature (see FIG. 21). A
 user can enter this routine after completion of message deposit as
 discussed above, or the user can initiate an inquiry by calling the VMS.
 Again, the VMS initially answers the call and transmits the opening prompt
 message to the caller. The caller inputs his or her unique authorization
 code which is verified by the VMS. The caller then enters a special
 function code (SFC) for an INQUIRY. The VMS determines whether or not any
 messages have been recorded for this subscriber. If there are no messages,
 the VMS plays a canned prompt so informing the subscriber. However, if
 there are messages recorded for the current caller, the VMS provides
 another canned message, and the person initiates playback by dialing a "2"
 (Column 26, lines 42-61). The caller can control the replay of the
 messages using additional dialed digit inputs, for example to repeat all
 or a segment of a message or to skip all or a segment of a message (table
 bridging Columns 23 and 24).
 The VMS system disclosed in the '752 patent will also automatically deliver
 messages to the identified addressees. In the DELIVERY routine (FIG. 15),
 the VMS calls the addressed recipient by dialing that person's telephone
 number. If the call is answered, the VMS plays a canned announcement which
 includes a request for the person answering the call to enter her unique
 authorization code. If there is no answer, the line is busy or the
 answering person does not enter the correct authorization code, the VMS
 will attempt to deliver the message again after a specified time period.
 When the answering party has responded by entering a valid authorization
 code, the VMS emits an idle tone, and the person can initiate playback by
 dialing a "2". The person listening to message playback can control the
 replay using additional dialed digit inputs, as discussed above (see
 Column 23, lines 30-65). The user is then given the option to redirect the
 message to another destination, deposit a reply message, save the message,
 or file the message for long term storage.
 In the reply routine, the user records a message for the sender of the
 message just replayed. In the redirect routine, the user enters a new
 address, and if desired, records a new message. The new message is
 appended to the original message already held in digital storage, and both
 messages are delivered to the new addressee using the DELIVERY routine
 discussed above ('752 patent, Column 25, lines 41-59, and FIG. 18).
 The file function disclosed in the '752 patent transfers a message to a
 "verbal file folder" for long term storage and later retrieval (Column 26,
 lines 11-18).
 The '906 and '129 patents include the subject matter of the parent '752
 patent discussed in detail above. The '906 and '129 patents, however, add
 a number of message processing features. For example, these patents add a
 delivery option referred to as "TIME-I.D. VALIDITY", which allows the user
 to specify a recipient and a date and time for delivery of a stored
 message.
 The '906 and '129 patents also expressly describe storing messages for an
 identified subscriber in terms of depositing messages "in the user's
 address". For example, one feature added in these patents is a "Priority
 Hold" feature. As described, if the deposited message meets certain
 priority conditions, the "VMS would automatically dial the user's
 telephone rather than deposit the message in the user's address, thereby
 forcing delivery" ('906 patent, Column 58, lines 4-12). Similarly, these
 patents describe depositing messages "in the user owner's RO message
 address", as will be discussed in more detail below (see, e.g., '906
 patent, Column 70, lines 51-53). Such references to depositing messages in
 a "user's address" indication that the addressing of messages for each
 subscriber in the Matthews et al. system defines "message baskets."
 The '906 and '129 patents also disclose several features which permit
 access by non-subscribers. For example, a subscriber can be assigned a
 receive only (RO) message address. To receive messages from a subscriber
 by using this address, a non-subscriber may call the VMS using a direct
 inward dial line (DID). In the specific example given in the patents, the
 subscriber might activate a call forwarding feature in the TELCO network
 whereby calls to her home telephone number are automatically forwarded to
 the DID/RO number into the VMS. When such a DID call comes in, whether
 forwarded or connected directly in response to dialing the DID number, the
 TELCO network will forward the last three or four digits of the DID number
 to the VMS system. The VMS uses the received digits to identify the RO
 address. If the subscriber prestored any messages in the RO address, the
 VMS will play those messages to the caller, otherwise the VMS will play a
 canned prompt indicating that the subscriber is not in ('906 patent,
 Column 69, lines 27-62).
 U.S. Pat. No. 4,625,081, issued Nov. 25, 1986, to Lawrence A. Lotito, et
 al. (hereinafter referred to as the "Lotito patent"). Referring to FIG. 1,
 the patent describes an automated telephone voice service system 100 which
 provides automatic recording and editing of voice messages as well as
 forwarding of recorded voice messages to other accounts and telephone
 numbers with or without operator assistance.
 The system includes a data store 104 coupled to store and retrieve voice
 messages at each of a plurality of individually addressable message
 baskets 1-N and a control system 102 providing a selective coupling
 between the store and each of a plurality of telephone lines of a
 telephone network 108.
 The data store may be physically implemented as one or more magnetic or
 electronic storage devices and may be distributed throughout a data
 processing system. The data store provides storage for a plurality of
 addressable message baskets, a plurality of individually addressable voice
 message prompts and client greetings, and an audit trail for each client
 accessing the system.
 Each message basket provides storage for a plurality of voice messages and
 is segregated into an inbasket section and an outbasket section. The
 inbasket functions in a manner analogous to a recording mechanism for a
 telephone answering machine and stores voice messages and message
 forwarding notices directed by system users to account owners of the
 associated message basket. The outbasket portion receives voice messages
 for forwarding to selected other message baskets or to telephone network
 108 users at indicated telephone numbers.
 In a fully automatic mode, the control system 102 can operate to call the
 indicated telephone number and upon its being answered, communicate an
 appropriate recorded voice message prompt, communicate the voice message
 being sent, and then terminate the call. As an example, the voice message
 prompt might inform the person answering the telephone at the indicated
 number that the person is about to receive a prerecorded message from the
 account owner. The account owner, when setting up his account, establishes
 predetermined distribution lists and sets of delivery instructions. The
 delivery instructions can cover such features as days of the week and time
 intervals during which delivery may be made, number of retries, and
 whether the forwarding of the message is to be accomplished automatically
 or semiautomatically with operator assistance.
 In a semiautomatic mode, the control system 102 waits for delivery
 conditions to be met, and then obtains ownership of an active operator
 console 106 including a terminal having a keyboard and a video display
 unit and an operator headset. The control unit informs an operator through
 the console 106 that a semiautomated message forwarding operation is to be
 undertaken and displays a prompting message for the operator to read. Upon
 command, the control system generates the Touch Tone signals corresponding
 to the recipient's telephone number and connects the operator console 106
 to the line when it is answered. The operator informs the answering party
 of the call, asks to talk to a particular person at the called telephone
 number if appropriate, and secures the permission of the called party to
 forward the voice message. The operator then commands the control system
 to communicate the voice message stored in the outbasket to the called
 telephone line as indicated by arrow 114.
 For voice messages forwarded to another inbasket rather than to a telephone
 number, the voice message is not actually recorded in duplicate in each of
 the designated inbaskets. Instead, a notification is stored in the
 inbasket which indicates that a forwarded message is stored by the system
 for delivery to the owner of the forwarding message basket. The
 notification indicates the particular outbasket and the particular message
 within the outbasket which is being forwarded. This enables the person
 sending the message to retain ownership of the message in his own
 outbasket and selectively change or delete the message until it has
 actually been delivered. Depending upon the delivery instructions of the
 sender and the preselected instructions of the recipient, a forwarded
 message might simply wait for delivery until the recipient retrieves the
 receipt of a message in his inbasket by a paging signal communicated over
 a paging system (not shown), by the illumination of an indicator light at
 the recipient's telephone, or by a telephone call to the recipient's
 telephone number informing the recipient by a prerecorded message that a
 message has been received in the recipient's inbasket.
 The prompts and client greeting section of the data store 104 stores a
 plurality of individually addressable voice message prompts explaining how
 to operate the voice service system and a client greeting for each
 inbasket. The greeting invites the caller to leave a message but does not
 identify the specific owner of the inbasket which has been accessed by the
 call. Each client may record and change his own personal greeting at will.
 The audit trail portion of the data store 104 stores a record for each
 caller accessing the system of the commands which have been given to the
 system by the caller. This record enables the control system to select
 particular voice message prompts.
 The particular functions executed by the control system depend upon by
 which one of the functionally different types of telephone lines the
 control system is accessed and upon the keyboard commands which are
 entered.
 The control system responds to an incoming call on a client's normal use
 telephone line (secretarial line) by waiting for a predetermined number of
 rings and then answering the telephone. The client greeting is accessed in
 the data store and communicated to the caller. The caller is invited to
 leave a message, which remains in the client's inbasket until retrieved by
 the client. A sophisticated caller who is familiar with the system is free
 to edit the voice message.
 Another type of line upon which a call might come into the system is a
 direct incall line. This line is dedicated to the particular inbasket of
 the client and is not available for general use by the client.
 A general incall line is similar to a direct incall line except that it is
 not associated with any particular message basket or inbasket. Upon
 accessing the system through a general incall line, a caller is prompted
 to enter a message basket number. The caller then is able to leave a
 message in a selected basket.
 A general access line is intended primarily for clients of the VSS and
 affords the broadest range of system functions. Upon calling in on a
 general access line, a client is prompted to enter his personal ID number
 if he desires to have access to the ownership privileges of his own
 account. This provides immediate access to message retrieval and the
 control system informs the caller whether or not there are any messages
 within his inbasket and, if there are, begins communicating the voice
 messages over the connected telephone line. Before each message is
 retrieved, the caller is informed of the age of the message. After
 reviewing the incoming messages, the caller is informed of the status of
 any outgoing messages in the caller's outbasket which are awaiting
 delivery.
 The telephone voice service system is implemented with a data processing
 system. As shown in FIG. 2 a telephone network 108 provides a number of
 physically different types of telephone lines to which connection must be
 made by the service system. Through a concentrator 202 up to 640
 secretarial lines are connected to a telephone room subsystem 206.
 Telephone room subsystems 214 and 216 are connected to special service
 telephone lines such as DID or DX tie lines. The concentrator and
 telephone room subsystems are physically located at one or more telephone
 company central offices or client PABX centers. The telephone room
 subsystems operate as interfaces between the digital portion of the
 service system and the analog telephone lines and trunks. It is possible
 for each telephone room subsystem to connect to up to 1260 voice grade
 circuit terminations. The system can accommodate up to 4 telephone room
 subsystems.
 Up to 4 real time subsystems receive the voice and control data from the 4
 telephone room subsystems. The real time subsystems provide selected
 switching connection between channels and communicate with an information
 processing system 250 for storage and retrieval of voice messages and
 system control.
 An interactive service subsystem 252 provides a communication connection
 between the information processing system 250 and input/output devices for
 the service system. The input/output devices may include keyboard display
 terminals 266, 268 and 270 within operator consoles 106, a printer 262 and
 a card reader 264.
 In addition to the systems described in the foregoing patents networking of
 voice mail systems has also been implemented to permit users in one
 location to use voice mail in other locations. The simplest form of
 networking voice mail is to use guest mailboxes, which are boxes assigned
 to persons outside the system. Another method of networking voice mail has
 been to terminate the voice mail on one switch and connect other switches
 to the central switch with networking software. A third method has been to
 network the voice mail systems themselves. However, generally speaking,
 the networked systems must be of the same manufacture because there are no
 standards for communication between systems. Work is underway to develop a
 set of standards known as Audio Message Interchange Service (AMIS) in the
 hope that when AMIS standards are approved, they will form a common
 language that network voice mail systems can support so voice mail of
 different manufacture can communicate.
 It is accordingly an object of the present invention to provide a system
 and method for effecting mailbox to mailbox communication in an
 expeditious and economical fashion basically utilizing existing equipment
 and network facilities. It is a feature of the invention that it permits
 such communication between mailbox equipment of different manufacturers.
 It is a still further feature of the invention that this is accomplished
 while blocking off the voice trunking network and blocking ringing of the
 telephone station corresponding to the recipient mailbox. This provides a
 significant lightening of the traffic load on the network trunking system,
 among other advantages which will become apparent upon the following
 description of the invention.
 According to one embodiment of the invention the local to remote
 mailbox-to-mailbox transfer is accomplished through the use of existing
 common channel signaling (CCS) packet networks and preferably through CCS
 Advanced Intelligent Networks (AIN). According to another embodiment of
 the invention the local to remote mailbox-to-mailbox transfer is
 accomplished through the use of the internetwork commonly referred to as
 the "Internet." This embodiment of the invention possesses the advantage
 that the Internet is presently operative on a world wide basis whereas
 interconnection of the existing AIN's of telephone operating companies has
 not yet been implemented over the entire United States. This is partially
 due to regulatory constraints and partially due to limitations in the
 common channel signaling systems of some telephone companies. While
 Internet users are presently able to engage in a limited form of voice
 communication using specialized computer programs, these are tailored to
 existing Internet procedures and facilities. Both participants in such
 communication must be Internet literate, have access to the Internet via
 computers meeting the necessary hardware and software requirements, be
 running compatible voice programs in their respective computers, and
 virtually simultaneously effect Internet connection in their respective
 locales. As a result such communications must be prearranged.
 DISCLOSURE OF THE INVENTION
 Architecture of Switched Telephone Networks Using an Advanced Intelligent
 Network (AIN)
 According to the present invention it has been discovered that it is
 possible to implement mailbox to mailbox data communication to transfer
 voice messages using the existing advanced intelligent network (AIN) in
 public switched telecommunications networks in the United States. 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). Appropriately equipped switches in the network, known as Service
 Switching Points (SSPs), communicate with the SCP and together they
 provide various AIN services. The SSP knows 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" to switch into providing
 AIN treatment. Once a trigger occurs, a query message is sent to the SCP
 asking for instructions. Based on information contained in the query
 message, the SCP 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. Only the SCP "knows" which service
 is being performed on a particular call. The SSP simply knows how to
 identify calls that require AIN processing and how to execute instructions
 provided by the SCP. For this reason, two services that are very different
 from the viewpoint of the subscriber and the SCP may appear identical to
 the SSP since it performs the same basic functions for both.
 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) which provides SSP and Network Access Point (NAP)
 capabilities. ASP provides services independent triggering and call
 processing capabilities and also supports OA&M (Operations, Administration
 and Maintenance). These capabilities interwork 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 SCP, such
 as terminating announcement or play announcement and collect digits,
 connection control under control of the SCP, business and residence custom
 services (BRCS) interworking, new terminating restrictions, ISDN
 interworking, notification of call termination (returned to SCP),
 enhancements for OA&M, and billing under control of the SCP. Further
 details are provided in AT&T 235-190-125 October, 1990. As there
 described, voice mail is readily implemented.
 According to the present invention caller to remote mailbox and/or mailbox
 to remote mailbox communication is provided using TCAP and SS7 messaging
 in the AIN while blocking or obviating trunking of voice messages. This is
 advantageously accomplished using existing voice mail equipment because
 that equipment is currently interfaced to the telephone network and is
 fully responsive to TCAP and SS7 protocols.
 According to the invention, a caller desiring to leave a voice message in
 the mailbox of a remote person may use a telephone to access his own voice
 mail system and mailbox and speak the message. The voice processing unit
 of the mailbox may operate its voice menu to direct the caller to depress
 a specified key when satisfied with the message. It may then query the
 caller as to whether he desires to send the message and, if so, to depress
 another specified key. The voice unit then may instruct the caller as to
 the procedure for keying in the identity of the destination and to depress
 a further specified key to send the message. The message is digitized in
 conventional fashion and stored in the mailbox of the sender. The caller
 may go on hook after depressing the designated send key. The depression of
 the send key causes the generation of a tone or other signal which is
 recognized by the acting SSP as a trigger. This local connection ends
 usage of the voice network.
 The trigger causes the SSP to frame a TCAP inquiry message which is
 directed to the SCP for instructions. The TCAP message includes
 information identifying the calling station and mailbox and the called
 station and the fact that the caller is requesting mailbox to mailbox
 message transfer. The SCP consults its database to establish whether the
 caller is authorized to communicate mailbox-to-mailbox and as to the
 existence and identity of a mailbox for the called number. The SCP then
 originates a response to the SSP to dispatch one or more SS7 packets to
 the called directory number and mailbox along with an appropriate routing
 label and handling instructions and carrying as the included information
 in the SS7 packet the digitized voice retrieved from the mailbox of the
 sender. The information may be in T1 protocol which is conventionally the
 output digital signal of mailbox equipment regardless of manufacture.
 Thus any translation which is necessary between the digitized message in
 the mailbox and the T1 or equivalent protocol used in the SS7 packets
 inherently occurs in the equipment furnished by the manufacturer.
 The number of SS7 packets which may be required will be dependent upon the
 length of the message as in conventional packet communication. Each packet
 includes a suitable header which permits reassembly in the original order
 at the destination. The fact that the packets may not arrive at the
 destination in the same order as originated is of no consequence in that
 real time voice communication is not involved in the transfer.
 The dispatched SS7 packet communication proceeds through the common channel
 signaling SS7 network until all of the packets are received at the
 destination. It is a feature of the invention that the redundancy of the
 SS7 network and packet switching techniques may entail individual packets
 traveling different routes to the same destination. This redundancy is
 utilized as a feature of the invention to enable the existing SS7 network
 to handle the digital packet communication involved without overload.
 When the packets reach the destination SSP and end office (EO) the packet
 headers contain the necessary information to direct the packets directly
 into the mailbox without setting up a connection to the associated
 telephone station and without initiating ringing of the telephone. The
 packets arrive in their transmitted form containing T1 protocol digitized
 voice which the recipient mailbox equipment is designed to receive and
 deposit as a digitized voice signal in the mailbox. Again, any necessary
 translation is accomplished by the existing mailbox equipment by virtue of
 the fact that its vendor must assure that it is compatible with the
 switched telephone network. Deposit of the message in the destination
 mailbox is followed by the customary notification of the mailbox
 proprietor that a message is waiting. The proprietor may then access the
 mailbox in conventional fashion and have the message delivered as an audio
 voice message in the usual fashion. The recipient then has the option of
 returning a message in a converse fashion by depressing predetermined keys
 at his telephone station which utilizes the information in the packet
 header to reverse the origination and destination identifications.
 Because current model SCP's include billing modules they can also effect
 billing. The data is sent out through the ISCP so that it can either be
 directed to the revenue accounting office on a direct line or send a TCAP
 message back into the SSP or end office switch to the originating number
 responsible for the origination of the call. Billing can be accomplished
 in any desired fashion, such as in bits per second, call setup, number of
 packets, or any combination or the same. The billing information may go
 into the journal on the switch to be forwarded to the revenue accounting
 office. The system of the invention is particularly suited to delivery of
 the same mailbox message to multiple mailbox destinations.

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's) 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.
 Referring to FIG. 1 there is shown a block diagram of a public switched
 telephone network and the SS7 network that is used to control the
 signaling for the switched network. Thus an analog switched telephone
 network is generally indicated at 10 having a common channel signaling
 network in the form of an SS7 network illustrated generally at 12. The
 switched telephone network consists of a series of central offices which
 are conventionally referred to as signaling points (SPs or SSPs) in
 reference to the SS7 network. Certain of these SPs comprise end offices
 (EOs) illustrated at 14, 16, 18 and 20 as EOs 1-4 in FIG. 1. Each
 signaling point has a point code comprising a 9-digit code assigned to
 every node in the network. In FIG. 1 EO1 has a point code of 246-103-001,
 EO2 has a point code of 246-103-002, EO3 has a point code of 255-201-103,
 and EO4 has a point code of 255-201-104.
 The end offices EO1 and EO2 represent end offices in the region of one
 regional operating company, while end offices EO3 and EO4 represent end
 offices of the region of a different operating company. Each operating
 company has its own network ID, shown here as 246 for the left region and
 255 for the right region in FIG. 1. The number 103 in the designation
 246-103-001, is the number of the cluster. A cluster can hold 32 SPs or
 members, the member being designated by the final 3 numbers. Thus 246 may
 represent C & P of Virginia Regional Operating Company, cluster 103,
 member EO2 for EO2 when viewed from an SS7 standpoint. The broken lines
 connecting the SPs together may be analog trunks or voice or similar
 circuits. The SPs in a given region are connected together by local trunks
 22, 24 and 26 in the left region and 28, 30 and 32 in the right region.
 The SPs in one region are connected to the SPs in other regions via
 inter-exchange carrier network trunks or ICN trunks 34 and 36 in FIG. 1
 connected to Access Tandems (ATs) 38 and 40 (AT1 and AT2). These SPs or
 ATs are shown as having point codes 246-103-003 and 255-201-101
 respectively.
 Referring to FIG. 1, the SS7 network 12 comprises a series of Signal
 Transfer Points (STPs) shown here at 40, 42, 44 and 46 designated STP1,
 STP2, STP3 and STP4. Each STP in a network is connected to the SPs in the
 network by A links indicated at 48, 50, 52 and 54. STP1 and STP2
 constitute a mated pair of STPs connected by C links 56 while STP3 and
 STP4 constitute a mated pair connected by C links 58, each mated pair
 serving its respective transport area. It will be understood that there
 may be multiple mated pairs per region, one for each designated transport
 area. STP1 is connected to STP3 by B link 60 and to STP4 by D link 62.
 STP2 is connected to STP4 by B link 64 and to STP3 by D link 66.
 As will be understood, the A, B, C and D links are physically identical
 with the designation relating to cost in terms of ease of access. The A
 links represent the lowest cost. B and D links have the same route cost
 with respect to SS7 so that the D designation is used only because it
 extends diagonally in the drawing. The C links are used to communicate
 between the two paired STPs for network management information and also
 constitute another route. The STPs in mated pairs have the same
 translations. Thus the translations in STP1 are the same as the
 translations in STP2, and the translations in STP3 are the same as the
 translations in STP4. The C links communicate between the paired STPs for
 network management information and SS7 message routing. The STP pair
 cannot function without the C links. Therefore, unnecessary utilization of
 the C links causes congestion and prevents the paired STPs from performing
 their intended function.
 The STPs are connected to Signal Control Points (SCPs) indicated in FIG. 1
 as an SCP 68 and an ISCP 70. The ISCP is an Integrated Signaling Control
 Point, which is basically the same as an SCP but comprises a larger and
 more powerful computer. AIN may also be regarded as another ISCP. SCPs are
 usually used for 800 and credit card services with ISCPs being used for
 AIN. However, this is optional. The ISCP may hold application information
 as well as routing information whereas an SCP contains routing
 information, i.e., routing tables.
 The SS7 network constitutes a highly redundant data network, generally a
 56K switched data circuit. By way of example, an SS7 message from EO2 to
 EO4 might travel any one of 8 possible routes. It could go from EO2 to
 STP1, from STP1 to STP3, STP3 to EO4. One variation on that route would be
 from STP1 down the D link 62 to STP4 to EO4, and so forth. In the event
 that a link between STP3 and EO4 was lost, an SS7 route could be
 established from EO2 to EO4 via STP1 to STP3 and then via C link 58 to
 STP4 to EO4. However, that would be an undesirable route in unnecessarily
 using the C link. A links provide direct connectivity while C links
 provide circuitous routes using extra switches, a situation to be avoided.
 An alternate route would be from STP1 via D link 62 to STP4 to EO4.
 Another reason for not using the C link is to avoid tying up the entire
 STP3-STP4 pair.
 The operation of placing a call from EO2 to EO4 may be described as
 follows: The user at EO2 picks up his phone and dials the number that
 resides in EO4. The SP generates an Initial Address Message (IAM). This
 message would have the destination point code of EO4, namely, point code
 255-201-104. It would have an originating point code of EO2, namely,
 246-103-002, in addition to miscellaneous other information needed for
 call set-up. That message would then be sent to either STP1 or STP2.
 Assuming that the message goes to STP1, STP1 would look at the message and
 determine that the message was not for it as an STP but rather is for EO4.
 STP1 would then investigate possible routings to get to 255 or EO4. B and
 D links are available and STP1 would choose one of the two. Assuming that
 it chooses the B link to STP3, STP3 repeats the same procedure. It
 determines that the message is for 255 or EO4 and puts that message on the
 A link to EO4.
 EO4 gets the IAM which has the called telephone number in it and determines
 whether or not the line is busy. If the line is not busy, EO4 generates an
 Address Complete Message (ACM) to indicate that it received the request
 for a call and that the number is not busy. That message is sent back by
 simply reversing the point codes. Now the destination point code is EO2
 and the originating point code is EO4. The message goes back to EO2 to
 indicate that the IAM was received and processed. As soon as the phone is
 answered at EO4, EO4 sends an Answer Message (ANS) back to EO2 indicating
 that the phone at EO4 was picked up, and at that time the trunks are
 connected together. EO2 connects its user to that trunk and EO4 connects
 its user to that trunk so that communication is established. All such
 messaging may occur in about 600 milliseconds which would be average but
 not necessarily fast.
 The foregoing constitutes the function of the STPs insofar as routing is
 concerned. The STPs look at a point code and if it is not for them they
 just pass it on via a route determined from translations and routing
 tables. The C link is the last route permitted and is not utilized unless
 no other route is available.
 As opposed to the foregoing, where the point code was for EO4 and not STP1,
 the point code may be for STP1. One example of such a situation would be
 the case of an 800 call. The 800 number is a fictitious number which is
 associated with a POTS number in a database in the SCP. Thus if EO2 makes
 an 800 call to EO4 it is necessary to determine the real telephone number.
 EO2 launches a Switching Connection Control Park (SCCP) message, which is
 a database request. This point code has a destination point code of an
 alias which is the point code of STP1 and STP2. STP1 and STP2 have various
 point codes indicated in FIG. 1 as 246-100-000 and 246-101-000. They also
 have alias point codes that indicate that they have a function to perform.
 Upon recognizing such a point code the STP does a data search and
 generates another SCP message to perform a database dip. This returns the
 real telephone number and the STP now has the destination point code of
 the real telephone number message. This is sent back to EO2. STP1
 determines that this message is not for me but for EO2. The message is
 sent back down to EO2. EO2 now has a real telephone number and the system
 performs the IAM and ACM procedure all over again to set up the call. The
 only difference between a regular direct call and an 800 call is the
 necessity to perform the dip to obtain the real number first. This
 procedure takes about 1.3 seconds because of the additional operation. The
 STPs have various databases, such as the 800 database and the credit card
 database, and there is still a further database for AIN. It is these
 databases which are utilized for the purposes of the present invention.
 The SS7 protocol describes how the signal messages are built and routed and
 provides for network management of the SS7 network itself. Thus if a link
 between EO4 and STP3 were to be lost, STP3 generates a transfer restricted
 message (TFR) to all nodes, i.e., all SPs connected to STP3, indicating
 that traffic is not to be sent to STP3 for EO4 because no route from STP3
 to EO4 exists. If both A links to EO4 were down, EO4 would essentially be
 isolated and the STP pair STP3 STP4 would broadcast a transfer prohibited
 (TFP) message indicating that nothing should be sent to the pair for EO4.
 In the transfer restricted situation it would be possible for STP3 to reach
 EO4 via the C link to STP4. This is a non-favored route but would be used
 in necessity. Handling such situations is the purpose of network managing
 messages. Congestion control or TFC accomplishes basically the same thing
 except that it constitutes a more sophisticated message limiting use of a
 circuit by stopping messages below a certain priority. Each message has a
 different priority. IAMs have a priority of 1 where ANS messages have a
 priority of 2.
 Upon congestion occurring in the STP node for EO4 a new call could not be
 sent to EO4 because it constitutes a priority 1 message which is
 restricted because the congestion level is 2. Only priority 2 messages and
 higher would be permitted. If a call is already existing it could be
 answered or released. Releases have a priority of 2 to permit call
 completion. New calls could not be initiated until the congestion had been
 removed or lowered to congestion status 1 or 0.
 The SS7 network constitutes a sophisticated network having a high
 predictability which is spelled out in the predetermined protocol. The SS7
 messages traverse the network at all times. The messages themselves
 comprise digital serial messages of various length that come into the STP.
 The start of the message is identified by a flag which is a zero followed
 by 6 ones and another 0. This constitutes a unique bit pattern in the SS7
 protocol. The protocol ensures that this particular pattern is not
 repeated until the next message. This provides a flag at the beginning of
 a new message. A flag at the end of a message is also provided usually in
 the form of the flag at the beginning of the next message, i.e., a message
 usually contains only one flag. The message is arranged in 8 bits or in
 octets. These octets represent the information carried by the message. The
 message contains both fixed and variable parameters. The Message Transport
 Part (MTP) of the SS7 message is always in the same place. The values
 change but the MTP is always in the same place.
 Referring to FIGS. 2 and 3, the start of a message is indicated at 72 with
 the commencement of the flag 74. The first 7 bits following the flag
 constitute the Backward Sequence Number (BSN). The eighth bit is the
 backward indicator bit which is used to track whether messages have been
 received correctly. The backward sequence number was the forward sequence
 of the other node's message when it was sent. Referring to FIG. 1, if EO2
 sends a message to EO4, EO2s include a Forward Sequence Number (FSN) in
 the 3rd octet of its message. Upon receiving this message, EO4 will
 include a Backward Sequence Number (BSN) equal to the FSN sent in the
 previous message in its next message to EO2. This indicated to EO2 that
 EO4 received the first message. This constitutes a positive acknowledgment
 of receipt of a message. If the eighth bit of the second octet or Backward
 Indicator Bit (BIB) is inverted, it indicates a failure to receive the
 identified message. If the 8th bit in the 2nd octet, Backward Indicator
 Bit (BIB), is inverted, it tells the receiving node that the identified
 message was not received. The accompanying BSN represents the last message
 that was received. The receiving node will then invert its Forward
 Indicating Bit (FIB), 8th bit of the 3rd octet, acknowledging a
 retransmission remission request, and will begin to send the missing
 messages until the transmitting end successfully acknowledges all
 remaining messages, i.e.:
 EO2 sends a message with a FSN of 5 to EO4;
 EO4 transmits a message back to EO2 with an inverted BIB and a BSN of 2,
 indicating that was the last message it received;
 EO2 then inverts its FIB and retransmits message 3;
 If EO4 acknowledges this message correctly (BSN of 3) EO2 will retransmit
 message 4 and then 5.
 Thus between the BIB and FIB and BSN and FSN, the STP keeps track of all of
 the messages sent between the two nodes at each end of a link. This
 provides predictability. If a node fails to receive an acknowledgment
 within a predetermined period of time it will take the link out of service
 because it is receiving no acknowledgments. This is usually a short period
 of time such as 1.6 seconds.
 Every 8 bits represents another part of the message until the end of the
 message. At about the fourth octet there is a length indicator to indicate
 the length of the message. In this case the message is bad in that it
 indicates six which is not a complete message. Assuming a complete message
 where the length indicator indicates 23 octets, this provides another
 means for error detection. Thus if the recipient counts to 28 this
 indicates that something is wrong and the message is sent again.
 Octet 5 is the Service Information Octet (SIO). This indicates whether it
 is a Fill In Signal Unit (FISU), Link Service Signaling Unit (LSSU) or
 Message Signaling Unit (MSU). MSUs are used for setting up calls or 800,
 LSSUs are used for alignment, and FISUs are fill in signals. Thus an LSSU
 is seen only if the link is out of service and going back into service or
 going out of service.
 Octets 6-11 contain the point codes. Thus the point code 235-81-8198 is the
 point code which would be read in FIG. 3. This is backwards as it comes
 from the message which arrives number, cluster, network ID in the order of
 bits received. That constitutes the routing label telling the STP and the
 nodes where the message came from and where it is going. Other parameters
 are involved depending upon the kind of message. If this were a FISU, that
 would be it. There would be 16 other bits that have Cyclic Redundancy
 Codes (CRCs) in them and another flag which would constitute the end. CRCs
 constitute a further error detection code which is a legal 1 function in
 the protocol. From the foregoing it will be seen that the messages contain
 various fields. This describes the basic format of an SS7 message which is
 the same for all messages of the same type.
 The SS7 protocol consists of four basic subprotocols:
 Message Transfer Part (MTP), which provides functions for basic routing of
 signaling messages between signaling points.
 Signaling Connection Control Part (SCCP), which provides additional routing
 and management functions for transfer of messages other than call set-up
 between signaling points.
 Integrated Services Digital Network User Part (ISUP), which provides for
 transfer of call set-up signaling information between signaling points.
 Transaction Capabilities Application Part (TCAP), which provides for
 transfer of non-circuit related information between signaling points.
 Architecture of A Telephone Network with Voice Mail
 In FIG. 5, there is shown a voice mail implementing communication system
 which includes at least one switching system 110 and at least one
 centralized message service voice mail system 120. The switching system
 110 may be a local or "end office" type telephone central office switch,
 such as a 1AESS or 5ESS switch sold by American Telephone and Telegraph.
 The end office switching system 110 typically includes, among other
 components, a space or time division switching matrix, a central
 processing unit, an input/output device and one or more data communication
 units. Structurally, the switching system 110 is a standard central office
 telephone switch. Each subscriber has at least one piece of customer
 premises equipment, illustrated as telephone station sets 131 to 133.
 Local telephone lines 135 to 137 serve as communication links between each
 of the telephone station sets 131 to 133 and the end office switching
 system 110. Although shown as telephones in FIG. 5, the subscriber station
 equipment can comprise any communication device compatible with the line.
 Where the line is a standard voice grade telephone line, for example, the
 subscriber station equipment could include facsimile devices, modems etc.
 The centralized message service or voice mail system in the illustrated
 example comprises voice messaging equipment such as a voice mail system
 120. Although referred to as "voice" messaging equipment, equipment 120
 may have the capability of storing messages of a variety of different
 types as well as voice messages. For example, a single system 120 may
 receive incoming messages in the form of audible messages, such as voice
 messages, as well as text format data messages. The voice messaging
 equipment 120 may also store messages in an image data format, such as
 facsimile. Message service systems having the capability to store messages
 in a variety of audible, data and image formats are known, see e.g., U.S.
 Pat. No. 5,193,110 to Jones et al., U.S. Pat. No. 5,008,926 to Misholi and
 U.S. Pat. No. 4,652,700 to Matthews et al.
 The illustrated voice mail system 120 includes a digital switching system
 (DSS) 121, a master control unit (MCU) 123, a number of voice processing
 units (VPUs) 125 and a master interface unit (MIU) or concentrator 127.
 The master control unit (MCU) 123 of the voice mail system 120 is a
 personal computer type device programmed to control overall operations of
 the system 120.
 Each of the voice processing units 125 also is a personal computer type
 device. The voice processing units 125 each include or connect to one or
 more digital mass storage type memory units (not shown) in which the
 actual messages are stored. The mass storage units, for example, may
 comprise magnetic disc type memory devices. Although not specifically
 illustrated in the drawing, the voice processing units 125 also include
 appropriate circuitry to transmit and receive audio signals via T1 type
 digital audio lines. To adapt the system 120 to receive information other
 than voice and/or offer services other than voice mail, one or more of
 VPUs 125 might be reprogrammed to run other types of applications and/or
 process other types of incoming information. For example, one such unit
 might process facsimile information, one might process E-mail, etc.
 An Ethernet type digital network 129 carries data signals between the MCU
 123 and the voice processing units 125. The Ethernet network 129 also
 carries stored messages, in digital data form, between the various voice
 processing units 125. The system 120 further includes T1 type digitized
 audio links 128 between the DSS switch 121 and each of the voice
 processing units 125.
 The voice mail system 120 connects to the switching system 110 via a number
 of simplified message desk interface (SMDI) type data lines 141.
 Specifically, these SMDI links 141 connect between one or more data units
 (not shown) in the end office switching system 110 and the MIU 127 in
 system 120. Each SMDI line 141 carries 2400 baud RS-232 data signals in
 both directions between the voice mail system 120 and the switching system
 110. The MIU 127 is a data concentrator which effectively provides a
 single connection of as many as 32 SMDI lines into the MCU 123 of the
 voice mail system.
 The voice mail system 120 also connects to the end office switching system
 110 via a number of voice lines 143 which form a multi-line hunt group
 (MLHG) between the switch matrix within the switching system 110 and the
 DSS switch 121 of the voice mail system 120. Typically, the MLHG lines 143
 consist of a number of T1 type trunk circuits which each carry 24 voice
 channels in digital time division multiplexed format.
 The above described voice mail system architecture is similar to existing
 voice mail type central messaging systems, such as disclosed in U.S. Pat.
 No. 5,029,199 to Jones et al., although other messaging system
 architectures such as disclosed in the other patents cited above could be
 used.
 For purposes of the present embodiment, the voice mail system 120, or other
 centralized messaging system, will further comprise a ring count change
 interface 151. The interface 151 connects to the Ethernet network 129 and
 provides two-way data communication between the network 129 in the voice
 mail system 120 and a multi-services platform (MSP) 153. For example, the
 unit 151 might provide a 9600 baud data channel over a line to the
 platform 153.
 The interface 153 will receive packets of data over the Ethernet network
 129 indicating changes in the status of the various subscribers'
 mailboxes. These packets of data will identify a particular subscriber and
 indicate the number a number of rings for future use in processing calls
 for that subscriber. The interface 153 forwards the ring count change data
 packets to the platform 153. The interface also receives data signals from
 the MSP 153, for example acknowledgements of transmitted data and/or
 signals indicating actual changes of status information by the switching
 system 110. In enhanced embodiments, the interface might include some data
 processing capabilities, as well. Also, the interface can provide
 instructions to change some other parameter of the call forwarding
 procedure, such as the subscriber's forwarding number.
 The multi-services platform 153 connects to the end office switching system
 110 via a recent change-memory administration channel (RC-MAC) 155. RC-MAC
 155 is a data link to the processor of the switching system 110 for
 inputting data into the translation tables used by the switching system
 110 to control switched communications operations with regard to each
 subscriber's line. The multi-services platform is a processor for
 receiving various service change instructions, including those from the
 interface 151 and from other sources, processing the instructions as
 necessary to make them compatible with switch programming, and forwarding
 instructions to the switching system 110 to change specific relevant
 translation table data stored in the switching system. In response to the
 change of status data from the ring count interface 151, the
 multi-services platform 153 provides appropriate data packet signals on
 the RC-MAC channel 155 to the end office switching system 110 to change a
 particular subscriber's ring count for forwarding on no answer. The
 instructions from the MSP 153 will identify a specific subscriber's line
 and will specify a ring count or ringing interval for use in determining
 when a call for that subscriber has gone unanswered and should be
 forwarded to the voice mail system 120. The multi-services platform may
 also forward instructions to change other parameters of the call
 forwarding function.
 Operation via an RC-MAC channel to change data in a switching system
 relating to call forwarding is described in U.S. Pat. No. 5,012,511 to
 Hanle et al., the disclosure of which is incorporated herein in its
 entirety by reference. The multi-services platform 153 is the same as or
 substantially similar to a processor used in the patented system to
 process various translation memory change requests, both from RC-MAC
 terminals and a voice response unit.
 Overview of Operation of Voice Mail Network
 In various operations discussed in more detail below, calls can be
 forwarded to the voice mail system 120 in response to calls to
 subscriber's lines. The switching system 110 may also route some calls
 directly to the voice mail system 120 in response to callers dialing a
 telephone number assigned to the lines 143 going to the voice mail system
 120. When the end office switching system 110 directs a call to the voice
 mail system 120, whether as a forwarded call or as a direct call in
 response to dialing of a number for accessing the system 120, the
 switching system places the call on any available channel on the
 multi-line hunt group lines 143.
 When the end office switching system 110 forwards a call to the voice mail
 system 120, the switching system 110 will also provide various data
 relating to the call via one of the SMDI links 141 and the MIU 127. In
 particular, the switching system 110 transmits data to the MCU 123 of the
 voice mail system 120 indicating which line of the multi-line hunt group
 143, i.e. which T1 trunk and which channel on the trunk, that the new call
 will come in on. The exchange 110 also transmits data via SMDI link 141
 identifying the called telephone number and the telephone number of the
 caller. For a call forwarded to a mailbox, the data from the exchange
 indicates the reason for the forwarding, and the caller telephone number
 (typically the directory number assigned to the called subscriber's normal
 telephone line) identifies which subscriber the forwarded call relates to.
 The master control unit 123 uses the multi-line hunt group line
 information and the subscriber's directory number to internally route the
 forwarded call though DSS switch 121 and one of the internal T1 links 128
 to an available voice processing unit 125 and identifies the relevant
 subscriber to that voice processing unit via the Ethernet 125.
 For each party who subscribes to a voice mail service provided by the
 centralized messaging system 120, the MCU 123 stores information
 designating one of the voice processing units 125 as the "home" unit for
 that subscriber. Each voice processing unit 125 stores generic elements of
 prompt messages in a common area of its memory. Personalized elements of
 prompt messages, for example recorded representations of each subscriber's
 name spoken in the subscriber's own voice, are stored in designated memory
 locations within the subscriber's "home" voice processing unit.
 In voice mail systems of the type discussed above, a subscriber's "mailbox"
 does not actually correspond to a particular area of memory. Instead, the
 messages are stored in each "mailbox" by storing appropriate
 identification or tag data to identify the subscriber or subscriber's
 mailbox which each message corresponds to.
 Each time a call comes in to the voice mail system 120, the master control
 unit 123 controls the digital switching system 121 to provide a
 multiplexed voice channel connection through to one of the voice
 processing units 125. Typically, the call connection goes to the "home"
 voice processing unit for the relevant subscriber. The voice mail
 subscriber is identified by data transmitted from the switching system
 110, as described above, if the call is a forwarded call. If all 24 T1
 channels to the "home" voice processing unit are engaged, the central
 processing unit 123 controls switch 121 to route the call to another voice
 processing unit 125 which is currently available.
 The voice processing unit connected to the call retrieves prompt messages
 and/or previously stored messages from its memory and transmits them back
 to the calling party via the internal T1 line 128, the DSS switch 121 one
 of the MLHG lines 143, end office switching system 110 and the calling
 party's telephone line, such as line 135 or line 137. The voice processing
 unit 125 connected to the call receives incoming messages from the caller
 through a similar route and stores those messages in digital form in its
 associated mass storage device.
 When the incoming call is a forwarded call, the connected voice processing
 unit 125 provides an answering prompt message to the caller, typically
 including a personalized message recorded by the called subscriber. After
 the prompt, the voice processing unit 125 records a message from the
 caller and identifies that stored message as one for the called
 subscriber's mailbox.
 At times the connected voice processing unit 125 will not have all
 necessary outgoing messages stored within its own associated memory. For
 example, a forwarded call normally will be connected to the called
 subscriber's "home" voice processing unit 125, but if the home unit is not
 available the forwarded call will be connected to a voice processing unit
 125 other than the subscriber's home voice processing unit. In such a
 case, the connected unit 125 requests and receives from the home unit 125
 the personalized components of the answering prompt message via the data
 network 129. The connected voice processing unit 125 will store the
 transferred message data in its own memory, and when necessary, will play
 back the transferred data from its own memory as outgoing messages in the
 exact same manner as for any prompts or greeting messages originally
 stored in its own memory.
 The connected voice processing unit 125 also will store any incoming
 message in its own associated memory together with data identifying the
 message as one stored for the called subscriber's mailbox. As a result,
 the system 120 actually may store a number of messages for any given
 subscriber or mailbox in several different voice processing units 125.
 Subsequently, when the voice mail subscriber calls in to the voice mail
 system 120 to access the subscriber's mailbox, the call is connected to
 one voice processing unit 125. Again, this call typically goes to the home
 unit 125 but would go to a different available one of the units 125 if the
 home unit is not available at the time. In response to appropriate DTMF
 control signals received from the subscriber, the connected voice
 processing unit retrieves the subscriber's messages from its own memory
 and plays the messages back to the subscriber. If any messages are stored
 in other voice processing units, the connected unit 125 sends a request
 the other units 125 to download any messages for the subscriber's mailbox
 those units have actually stored. The downloaded messages are stored in
 the memory of the connected voice processing unit 125 which replays them
 to the subscriber.
 Voice Mail Architecture in (AIN) Network
 FIG. 6 shows an architecture for providing centralized messaging type
 services, such as voice mail, using AIN for its conventional purpose. In
 the communication system shown in that drawing, elements corresponding to
 identical elements in FIG. 5 are identified with identical reference
 numerals. For example, the voice mail system 120 in FIG. 5 is identical to
 that shown in FIG. 6 and connects to the switching system via the
 multi-line hunt group (MLHG) 143, the SMDI links 141, the multi-services
 platform or MSP 153 and the associated RC-MAC channel 155. The voice mail
 system 120 operates essentially as described above, with respect to FIG.
 5.
 In the embodiment of FIG. 6, the end office switching system 210 is a
 Service Switching Point (SSP) capable switching system. SSP's are
 appropriately equipped programmable switches (such as a 5ESS) 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. The SSP functionality may reside in an end office such as
 shown at 210, or the SSP functionality may reside in a tandem office such
 as shown at 211, which in turn provides trunk connections to one or more
 other end offices 215 which lack SSP capability. End offices without such
 functionality route AIN calls to one of the SSP type offices.
 The SSP's 210 and 211 connect to each other via trunk circuits for carrying
 large numbers of voice communications, such as the trunk circuit shown as
 thick dark line 255 in FIG. 6. The SSP's 210 and 211 also connect to an
 STP 239 via data links 251, 257, for signaling purposes. An STP can
 connect to a large number of the SSP's. The STP 239 provides data
 signaling communications between the SSP's 210, 211 and with the ISCP 240.
 Although shown as a single STP, the AIN may include a number of STP's
 organized in an appropriate hierarchy to handle the expected level of
 signaling traffic. The data links 251, 257 between the SSP type switching
 systems 210, 211 and the STP 239 are typically SS7 (Signaling System 7)
 type CCIS interoffice data communication channels. The STP 239 in turn
 connects to other STP's and to the ISCP via a packet switched network 253
 which may also be an SS7 network. The above described data signaling
 network between the SSP type offices and the ISCP is preferred, but other
 signaling networks could be used.
 The messages transmitted between the SSP's 210, 211 and the ISCP 240 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. For example, an initial TCAP query from an SSP includes
 among other data a "Service Key" which is the calling party's address, and
 the digits dialed by the caller. TCAP also specifies a standard message
 response format including routing information, such as primary carrier ID,
 alternate carrier ID and second alternate carrier ID and a routing number
 and a destination number. The TCAP specifies a number of additional
 message formats, for example a format for a subsequent query from the SSP,
 and formats for "INVOKE" messages for instructing the SSP to play an
 announcement or to play an announcement and collect digits.
 There could be one or more ISCP's per state, to avoid overloading existing
 CCIS data links. Alternatively, the ISCP could be implemented on a LATA by
 LATA basis or on a regional operating company basis, i.e. one database for
 the entire geographic area serviced by one of the Regional Bell Operating
 Companies. In fact, if federal regulations permit, the database service
 may be offered nationwide.
 The ISCP 240 is an integrated system. Among other system components, the
 ISCP 240 includes a Service Management System (SMS) 241, a Data and
 Reporting System (DRS) 245 and the actual database referred to as a
 Service Control Point (SCP) 243. The ISCP 240 also typically includes a
 terminal subsystem referred to as a Service Creation Environment or SCE
 242, for programming the database in the SCP 243 for the services
 subscribed to by each individual subscriber to one of the AIN services.
 The SMS 241 validates service logic and data entered by the TELCO or the
 subscriber, and manages the process of actually updating the data files in
 the SCP database 243.
 Each central office switching system or SSP normally responds to a service
 request on a local communication line connected thereto, for example an
 off-hook followed by dialed digit information, to selectively connect the
 requesting line to another selected local communication line. The
 connection may be made locally through only the connected central office
 switching system. For example, for a call from station 131 to station 132
 the end office type SSP 210 provides the call connection without any
 connection to another central office. When the called line connects to a
 distant station, for example for a call from station 11 to station 231,
 the connection is made through the connected end office switching system
 SSP 210 and at least one other central office switching system, such as
 tandem SSP 211 and end office 215, by means of the telephone trunks
 interconnecting the various office switching systems.
 In the normal call processing, the central office switching system responds
 to an off-hook and receives dialed digits from the calling station. The
 central office switching system analyzes the received digits to determine
 if the call is local or not. If the called station is local and the call
 can be completed through the one central office, the central office
 switching system connects the calling station to the called station. If,
 however, the called station is not local, the call must be completed
 through one or more distant central offices, and further processing is
 necessary. If at this point the call were connected serially through the
 trunks and appropriate central offices between the caller and the called
 party using in channel signaling, the trunks would be engaged before a
 determination is made that the called line is available or busy.
 Particularly if the called line is busy, this would unnecessarily tie up
 limited trunk capacity. The CCIS system through the STP's originally was
 developed to alleviate this problem.
 In the CCIS type call processing method, the local central office suspends
 the call and sends a query message through one or more of STP's. The query
 message goes to the central office to which the called station is
 connected, referred to as the "terminating" central office. The
 terminating central office determines whether or not the called station is
 busy. If the called station is busy, the terminating central office so
 informs the originating central office which in turn provides a busy
 signal to the calling station. If the called station is not busy, the
 terminating central office so informs the originating central office. A
 telephone connection is then constructed via the trunks and central
 offices of the network between the calling and called stations. The
 receiving central office then provides a ringing signal to the called
 station and sends ringback tone back through the connection to the calling
 station.
 The call processing routines discussed above are similar to those used in
 existing networks to complete calls between stations. In an AIN type
 network system, these normal call processing routines would still be
 executed for completion of calls between customer stations, when call
 processing does not involve one of the AIN services.
 In an Advanced Intelligent Network (AIN) type system, such as shown in FIG.
 6, certain calls receive specialized AIN type processing under control of
 data files stored in the SCP database 243 within the ISCP 240. In such a
 network, the SSP type offices 210, 211 of the public telephone network
 detect a call processing event identified as an AIN "trigger" For ordinary
 telephone service calls, there would be no event to trigger AIN
 processing; and the local and toll office switches would function normally
 and process such calls as discussed above, without referring to the SCP
 database 243 for instructions. An SSP type switching office which detects
 a trigger, however, will suspend call processing, compile a TCAP formatted
 call data message or "query" and forward that message via a common channel
 interoffice signaling (CCIS) link 251 or 257, the STP 239, and link 253 to
 the ISCP 240 which includes the SCP database 243.
 The TCAP query message contains a substantial amount of information,
 including for example data identifying the off-hook line, the number
 dialed and the current time. Depending on the particular AIN service, the
 ISCP uses a piece of data from the query message to identify a subscriber
 and access the subscriber's files. For example, for some form of
 terminating type AIN service the dialed number would correspond to the
 called AIN subscriber, therefore the ISCP 240 uses the dialed number to
 access the subscriber's data file within the SCP database 243. From the
 accessed data, the ISCP 240 determines what action to take next. If
 needed, the ISCP 240 can instruct the central office to obtain and forward
 additional information, e.g., by playing an announcement and collecting
 dialed digits.
 Once sufficient information about the call has reached the ISCP 240, the
 ISCP accesses its stored data tables to translate the received message
 data into a call control message. The call control message may include a
 substantial variety of information including, for example a destination
 number and trunk group selection information. The ISCP 240 returns the
 call control message to the particular SSP 210 or 211 which initiated the
 query via CCIS links and the STP 239. The SSP then uses the call control
 message to complete the particular call through the network.
 The SSP type switches can recognize a variety of events as triggers for
 activating a query and response type AIN interaction with the ISCP, and
 different AIN services use different types of triggers. The present
 invention involves a call forwarding or call redirect type AIN service and
 uses a dialed destination number as the triggering event. This type of
 trigger is sometimes referred to as a terminating trigger. Other types of
 AIN type services using the dialed number of the terminating station or
 subscriber as the trigger are disclosed in commonly assigned U.S. Pat. No.
 5,353,331 entitled Personal Communication Services Using Wireless/Wireline
 Integration, and U.S. patent application Ser. No. 07/888,098 filed May 26,
 1992, entitled Method for Concurrently Establishing Switch Redirection for
 Multiple Lines, the disclosures of these two commonly assigned
 applications being incorporated herein in their entirety by reference.
 In the AIN embodiment shown in FIG. 6, the voice mail system 120 operates
 exactly as in the first embodiment shown in FIG. 5. The SSP type switching
 system 210 provides a forward on `no answer` condition of the type used in
 the embodiment of FIG. 5. The difference is that the switching system 210
 will use the forwarding operation only for the high count forwarding when
 no new message is stored. In such cases, the switching system 210 will use
 only one relatively high ring count threshold for all calls to any given
 voice mail subscriber's line. This threshold value may be a high default
 value. The SSP type switching system 210 will execute this forwarding
 routine with a high threshold for calls to a subscriber after a signal
 from the voice mail system indicating that all new messages for that
 subscriber have been replayed.
 In switch based call forwarding of the type discussed above, if a called
 subscriber's line is available, the switching system terminates calls for
 a subscriber on the subscriber's line. The switching system forwards the
 call to the forwarding number, e.g., a number associated with the
 multi-line hunt group 143 into the voice mail system, only if no one
 answers the call for a certain ringing interval or a certain ring count.
 In an AIN, such as shown in FIG. 6, the network can reroute a call without
 first terminating the call on the called line. The AIN actually redirects
 the call to the destination during initial call processing, without
 waiting for a no-answer condition. The illustrated AIN embodiments of the
 present invention rely on such AIN type call redirection to route calls
 for subscriber's who have messages waiting directly to the voice mail
 system 120.
 To initiate AIN type call redirection, the switching system sets a
 destination number trigger in its internal translation information
 associated with a particular subscriber's line. The trigger is set in
 response to a signal from the voice mail system 120 indicating that the
 system 120 has stored new messages for that subscriber.
 While the trigger associated with a subscriber's line is active, when the
 SSP switching system 210 receives a call to that subscriber, the SSP will
 suspend call processing and query the ISCP 220 for a destination number to
 actually route the call to. The ISCP 240 will return a number associated
 with the multi-line hunt group 143, and the SSP type switching system 210
 will connect the call to one of the lines of that group 143. To the
 caller, the first ringback heard will correspond to the first ring at the
 voice mail system. This results in a forwarding to the voice mail system
 without a prior ring.
 In the system of FIG. 6, the voice mail system 120 will still send some
 form of signal to the switching system 210 through the interface 151, the
 MSP 153 and the RC-MAC channel 155 equivalent to the ring count change to
 high instruction. In response, the switching system 210 will cancel the
 terminating trigger designation associated with the particular
 subscriber's line. The next call to the subscriber will therefore be
 forwarded by the switch after a high number of rings without an answer on
 the subscriber's line.
 The construction and operation of the voice mail systems shown in FIGS. 5
 and 6 are described more fully in commonly assigned patent application
 Ser. No. 08/121,855 allowed Jun. 13, 1995 (Attorney Ref. No. 680-068),
 entitled Toll Saver for Centralized Messaging Systems, the disclosure of
 which commonly assigned application is incorporated herein in its entirety
 by reference.
 Referring to FIG. 4 there is shown a simplified diagram of a public
 switched network such as illustrated and described with more detail in
 connection with FIG. 1. The network in FIG. 4 includes a voice mail system
 associated with each of the switching systems. FIG. 4 shows two SSP's 310
 and 312 which comprise end office switching systems 314 and 316. The end
 office 314 represents an end office in the region of one regional
 operating company, while end office 316 represents an end office in the
 region of a different operating company. The SSP's in a given region are
 connected together by local trunks (not shown) and the SSP's 310 and 312
 are connected via access tandems (not shown) and inter-exchange carrier
 network trunks or ICN trunk 318 in FIG. 4.
 The SS7 network, indicated generally at 320, includes a series of STP's
 322, 324, 326 and 328 designated STP 1, STP 2, STP 3 and STP 4. Each STP
 is connected to the other STP's by A links indicated at 330, 332, 334 and
 336. STP 1 and STP 2 constitute a mated pair of STP's connected by C links
 338, each mated pair serving its respective transport area. STP 1 is
 connected to STP 3 by B link 342 and to STP 4 by D link 344. STP 2 is
 connected to STP 4 by B link 346 and by D link 348.
 The STP's are connected to ISCP's 350 and 352 by A links 354, 356, 358 and
 360.
 Each switching system 314 and 316 in this illustration comprises an end
 office and is connected to customer premises equipment, illustrated as
 telephone stations 362, 364, 366, 368, 370 and 372. Local telephone lines
 or local loops serve as communication links between each of the telephone
 stations and its end office switching system. It will be understood that
 the subscriber station equipment may also comprise other communication
 devices compatible with the line, such as facsimile devices, modems, etc.
 Each switching system 314 and 316 is also provided with a centralized
 message service or voice mail system shown in FIG. 4 as 374 and 376. These
 systems may be of the type illustrated and described in detail in
 connection with FIG. 5 and 6. Although referred to as voice messaging
 equipment, the systems 374 and 376 may have the capability of storing
 messages of a variety of different types as well as voice messages. For
 example, a single system may receive incoming messages in the form of
 audible messages such as voice messages, as well as text format data
 messages. The equipment may also store messages in an image data format
 such as facsimile.
 The voice mail systems 374 and 376 connect to the switching systems 314 and
 316 via SMDI data lines 378 and 380 and by multi-line hunt groups (MLHG's)
 382 and 384. Typically, the MLHG lines consist of a number of T1 type
 trunk circuits which each carry 24 voice channels in digital time division
 multiplexed format.
 The operation of the system shown in FIG. 4 according to the invention may
 be as follows:
 A subscriber associated with telephone station 362 desiring to leave a
 voice message in the mailbox of a remote subscriber, such as the
 subscriber associated with telephone station 368, may use a telephone
 station to access his own voice mailbox in the voice mail system 374. This
 may be accomplished by dialing a number associated with the voice mail
 system 374 for this purpose. The voice processing unit of the voice mail
 system may operate its voice menu to direct the caller to depress a
 specified key when satisfied with the message in a known fashion. It may
 then query the caller as to whether he desired to send the message and, if
 so, to depress another specified key. The voice unit then will instruct
 the caller as to the procedure for keying in the identify of the
 destination and to depress a further specified key to send the message.
 This foregoing procedure is not intended to be exclusive and other
 procedures for leaving and commanding the dispatch of a message which are
 described in the background patents discussed above may be utilized. In
 all cases the message is digitized in conventional fashion and stored in
 the mailbox of the sender. The caller may go on-hook after depressing the
 designated send key. The depression of the send key causes the generation
 of a tone or signal which is recognized by the SSP 310 as a trigger.
 In response to the trigger, the SSP frames a TCAP inquiry message which is
 directed via one or more of the STPs 322 and 324 to the ISCP 350 for
 instructions. The TCAP message includes information identifying the
 calling station and mailbox and the called station and the fact that the
 caller is requesting mailbox-to-mailbox message transfer. The ISCP
 consults its database to establish the existence and identity of a mailbox
 for the called number. If the identity of such a mailbox is found, the
 ISCP then originates a response to the SSP to packetize and dispatch one
 or more SS7 packets to the called directory number and mailbox (if
 available) with an appropriate routing label and handling instructions and
 carrying as the included information in the SS7 packets the digitized
 voice retrieved from the mailbox of the sender. An illustrative packet is
 shown in FIG. 7 with the digital message information incorporated at 386.
 The information may be in T1 protocol which is conventionally the output
 digital signal of mailbox equipment used in the public switched telephone
 networks regardless of manufacture. Thus any translation which is
 necessary between the digitized message in the mailbox and the T1 or
 equivalent protocol used in the SS7 packets inherently occurs in the
 equipment furnished by the voice mail system manufacturer.
 The number of SS7 packets which may be required will be dependent upon the
 length of the message as in conventional packet communication. Each packet
 includes suitable header information in the conventional manner. In this
 case if the identity of the destination mailbox was established from the
 database of the ISCP 350, that identity will be included in the outgoing
 packets. However, if the existence and/or identity of a mailbox associated
 with the destination directory number is not subject to determination in
 the database of the ISCP 350, the SSP 310 is instructed by the ISCP 350 to
 include in the packet header appropriate directions to the remote SSP 312
 to cause triggering and the formation and dispatch of a TCAP inquiry
 message to the associated ISCP 352. In such a case the ISCP 352 conducts a
 dip of its database and provides the requested information to the SSP 312.
 The packet is thereupon processed through the SSP 312 and voice mail
 system 376 to digitally record the contents of the remotely originated
 information. Again the voice mail system is so designed as to inherently
 handle any translation necessary to communicate with the switching system
 in T1 or equivalent protocol. The fact that the packets may not arrive at
 the destination in the same order as originated is of no consequence in
 that real-time voice communication is not involved in the transfer.
 The dispatched SS7 packet communication proceeds through the common channel
 signaling SS7 network until all of the packets are received at the
 destination. It is a feature of the invention that the redundancy of the
 SS7 network and packet switching techniques permits packets traveling
 different routes to the same destination. This redundancy is utilized as a
 feature of the invention to enable to existing SS7 network to handle the
 digital packet communication involved without requiring modification of
 the SS7 system.
 When the packets reach the destination SSP 312 and have been deposited in
 the mailbox of the addressee, the voice mail system 376 effects customary
 notification of the mailbox proprietor that a message is waiting. The
 proprietor may then access the mailbox in conventional fashion and have
 the message delivered as an audio voice message in the usual fashion. The
 recipient then has the option of returning a message in a converse fashion
 by depressing the appropriate keys at his telephone station which utilize
 the information in the packet header to reverse the origination and
 destination identifications. If the mailbox-to-mailbox communication
 feature is furnished by the involved telephone companies as an extra
 feature, it will be appreciated that either or both ISCP's 350 and 352 may
 ascertain from their appropriate databases the authorization of the user
 to access the service.
 Because currently available ISCP's include billing modules they may also
 effect billing. The data may be sent out through the ISCP so that it can
 either be directed to the revenue accounting office on a direct line or it
 may send a TCAP message back into the SSP or end office switch to the
 originating number responsible for the origination of the call. Billing
 can be accomplished in any desired fashion, such as an bits per second,
 call set-up, number of packets, or any combination of the same. The
 billing information may go into the journal on the switch to be forwarded
 to the revenue accounting office.
 According to another embodiment, the invention provides a system and method
 for transferring voice mail or messages to called parties who are not
 voice mail subscribers and thus do not possess individual or personal
 mailboxes. Pursuant to this embodiment of the invention, Voice Mail
 Systems 374 and 376 in the simplified network illustrated in FIG. 4 are
 provided with multiple unsubscribed mailboxes, which are here described as
 public mailboxes or mailboxes for temporary hire. It will be understood
 that such mailboxes may constitute mere addresses or addressable storage
 or memory in the voice mail system storage. Such mailboxes may be utilized
 according to a first embodiment of the invention in the following fashion.
 A caller at telephone station 362 connected to central office 314 makes a
 call to a remote called party at station 370 at central office 316. In
 this case the common channel signaling system 320 determines that the call
 cannot be completed because of a busy or a no answer situation. The
 attempt to establish a voice connection between the two telephone stations
 is terminated and the caller is directed, as by voice prompt, to the voice
 mail system 374 associated with the originating central office 314. The
 voice processing unit associated with the voice mail system 374 informs
 the caller that the line is busy or that there is no answer and inquires
 as to whether or not the caller would like to leave a message. It also
 indicates that if the caller chooses to leave a message the charge will
 be, for example, twenty-five cents, which will be charged to his telephone
 bill. The Voice Processing Unit requests a yes or no response, either by
 voice or DTMF key or the equivalent. Where the response is affirmative,
 the caller is invited to leave the message in the conventional voice mail
 fashion and the message is stored in a public mailbox in the voice mail
 system 374. Appropriate messaging then occurs via the SMDI link 378 to
 effect billing to the caller. Subsequent to termination of the deposit of
 the message as digitally stored data, the message is transferred in
 digital form from the public mailbox in voice mail system 374 to a
 temporarily mating or corresponding public mailbox in voice mail system
 376. Such transfer is via the common channel signaling link pursuant to
 the invention as previously described. Following deposit or storage of the
 message in the destination voice mail system 376, that voice mail system
 initiates attempts to reach the called party or addressee at telephone
 station 370 to announce to that party that a message has been deposited
 for retrieval. The same announcement may include the instruction that the
 message may be retrieved by depression of a stated DTMF key. The actuation
 of the key may create a record constituting a receipt for the originating
 party. The digitally stored voice message is then delivered from the voice
 mail system 376 to the caller at station 370 as an audio voice message in
 the usual fashion. The notification of the receipt may be transferred to
 the billing record of the originating caller via the common channel
 signaling system and receipt noted with the billing for the delivery of
 the message.
 As a still further feature of the invention, the original invitation to
 leave a message to the caller can include a further inquiry as to whether
 or not the caller requests a reply. The announcement may indicate that the
 delivery of the request and delivery of any reply would entail an
 additional charge of, for example, twenty-five cents. In the event that
 the caller requests a reply, the information which is transferred via the
 common channel signaling system pursuant to the invention includes an
 appropriate bit to indicate that a reply is requested. When the
 destination voice mail system delivers the message it responds to that bit
 by voicing a message that informs the recipient that a reply is requested.
 Instructions as to delivering a reply are provided to the called party or
 addressee by the destination voice mail system. The called party may then
 record the reply as digitized data in the local voice mail system 376.
 Subsequent to termination of the connection between the destination voice
 mail system and the called party, the reply is transferred via the common
 channel signaling system back to the originating voice mail system 374 as
 previously described. The digitally stored reply is then delivered to the
 original calling party by a call from the voice mail system to the
 originating telephone station 364. The reply is also delivered as an audio
 voice message.
 In the embodiment of the invention just described, the situation involved a
 busy or no answer condition. It is still another feature of the invention
 to offer the service of audio voice message delivery without an attempt to
 establish two-way telephone connection with the called party. Such a
 service may be set up using a real or virtual directory number to trigger
 the service. Dialing such number establishes a connection to a voice mail
 system local to the calling party having public or for hire mailboxes as
 previously described. This may be a public mailbox in the local voice mail
 system 374 in FIG. 4. The caller is invited to speak the message and the
 voice processing unit of the voice mail system may then operate its voice
 menu to direct the caller to depress a specified key when satisfied with
 the message in a known fashion. It may then query the caller as to the
 destination directory number. This may be followed by an inquiry to
 establish whether the caller requests a reply. Billing information is
 provided to the caller and suitable billing signaling is effected, as by
 use of the SMDI link 378 to the local voice mail system 374.
 Following storage of the digitized voice message and digitized signaling
 regarding delivery and response, the digitized message is transferred via
 the common channel signaling system to a destination public mailbox in a
 voice messaging system designated by the ISCP on the basis of the
 directory number of the called party. This mailbox may be in the remote
 voice mail system 376 where the digitized message and instructions are
 stored. Delivery of the message is then effected in the same manner as
 previously described. Any reply is first stored in the public mailbox in
 voice mail system 376 and subsequently transferred through the common
 channel signaling system to the originating voice mailbox. The reply is
 then delivered to the original calling party by a telephone call to the
 originating telephone station.
 As a still further additional feature of this embodiment of the invention,
 the methodology may be utilized to provide a 900 directory number type
 service. For example, an arrangement may be made for a well known
 celebrity to provide specified short duration responses to questions from
 fans. According to this embodiment of the invention, one or more
 pre-designated mailboxes is provided at each voice mail system offering
 the service. In-calling fans, such as using the telephone stations 362-366
 in FIG. 4, are connected to the local voice mail system 374 via the
 multi-line hunt group (MLHG) 382. Such callers record their queries in
 pre-designated mailboxes or in one mailbox using multiple addresses. The
 callers are billed in a conventional fashion using appropriate SMDI
 signaling and billing procedures.
 The digitized stored questions are transferred in due course as digitized
 messages over the common channeling signaling system as previously
 described. The messages are received and stored in the remote voice mail
 system 376, preserving the address of the query originator. The
 destination voice mail system 376 may be located anywhere in the system
 but is preferably local to the responding celebrity. A contractual
 arrangement is made with the celebrity whereby the celebrity periodically
 establishes a connection with the remote voice mail system and seriatim
 retrieves and responds to the questions.
 Appropriate records for payment to the celebrity are created at the remote
 voice mail system and central office and associated platforms to effect
 the creation of an invoice and payment based upon the number of inquiries
 to which a response is made. Each response is digitally stored in the
 voice mail system local to the celebrity and subsequently delivered via
 the common channel signaling network to the enquiring callers or fans.
 Such delivery may be carried out in any convenient manner. Thus a call may
 be made from the local voice mail system 374 to the telephone station
 which initiated the question and the response may be delivered.
 Alternatively, the callers may merely be notified that their responses are
 ready for retrieval.
 Referring to FIG. 8 there is shown another embodiment of the invention
 wherein communication between voice mail systems in two remote telephone
 networks is implemented through use of the Internet. The internetwork
 commonly known as the Internet had its genesis in U.S. Government (called
 ARPA) funded research which made possible national internetworked
 communication systems. This work resulted in the development of network
 standards as well as a set of conventions for interconnecting networks and
 routing information. These protocols are commonly referred to as TCP/IP.
 The protocols generally referred to as TCP/IP were originally developed
 for use only through Arpanet and have subsequently become widely used in
 the industry. TCP/IP is flexible and robust; in effect, TCP takes care of
 the integrity and IP moves the data. Internet provides two broad types of
 services: connectionless packet delivery service and reliable stream
 transport service. The Internet basically comprises several large computer
 networks joined together over high-speed data links ranging from ISDN to
 T1, T3, FDDI, SONET, SMDS, OT1, etc. The most prominent of these national
 nets are MILNET (Military Network), NSFNET (National Science Foundation
 NETwork), and CREN (Corporation for Research and Educational Networking).
 In 1995, the Government Accounting Office (GAO) reported that the Internet
 linked 59,000 networks, 2.2 million computers and 15 million users in 92
 countries.
 According to the instant embodiment of the invention the remote Voice Mail
 Systems (VMS) are handled as remote LANs. Each of these LANs is connected
 to the Internet over an interface or gateway connection with a standard
 LAN environment (IEEE 802.3, 802.4, 802.5) over a LLC (logical link
 control) utilizing CSMA/CD, token ring, token bus, or the like. The LLC
 procedure is that part of the protocol that governs the assembling of DLL
 frames and their exchange between data stations independent of how the
 transmission medium is shared. The protocol provides transparency to the
 network layer with respect to the underlying LAN media. The interface
 includes a conventional IP router or a bridge-like IP router to implement
 the IP and TCP protocols. By way of example a bridge-like router may
 comprise a router and function of the type described in Perlman et al.
 U.S. Pat. No. 9,309,437, entitled "Bridge-Like Internet Protocol Router,"
 issued May 3, 1994, or similar equipment. The Internet itself is linked
 largely by telephone lines which are mostly T-1 lines.
 According to this embodiment of the invention each linked telephone company
 has its own Internet access and IP address. The customers of the telephone
 companies or users of the service need have no individual Internet access
 or address.
 FIG. 8 shows the architecture of two public switched telephone networks
 (PSTNs) of the type previously described with respect to FIG. 4 employing
 voice mail systems such as described with respect to FIGS. 5 and 6. The
 PSTNs are shown as clouds 400 and 402 having voice mail systems 404 and
 406 of the type described. The voice mail systems are connected to the
 Internet 414 via interfaces, routers or gateways 408 and 410 of the type
 described. FIG. 9 shows such a voice mail system such as that shown in
 FIG. 5 connected to the interface 408 and Internet 414. The connection in
 the voice mail system 120 is made to the Ethernet 129 via link 428. As
 previously described, the Ethernet carries stored messages in data form in
 addition to other types of data signaling. Messages destined for the
 interface 408 are directed to the router therein by the voice mail system
 master control unit (MCU) 123. The PSTNs serve subscribers or customers
 via illustrative telephone terminals 416-426, which may if desired be POTS
 terminals.
 The operation of the service and system is as follows:
 In a first example a subscriber to voice mail service in one telephone
 network desires to send a voice message to a subscriber in another network
 who is known to have a voice mail box in that network. This may be viewed
 as a subscriber to the voice mail service 404 in the telephone network 400
 in FIG. 8 desiring to send a voice message to a subscriber to the voice
 mail service 406 in the telephone network 402. Such voice mail services
 are shown at 374 and 376 in FIG. 4. The voice mail services may be of the
 type shown in detail in FIGS. 5 and 6.
 A subscriber associated with telephone station or terminal 416 in FIG. 8
 (362 in FIG. 4) desiring to leave a voice message in the mailbox of a
 remote subscriber, such as the subscriber associated with telephone
 station or terminal 426 in FIG. 8 (368 in FIG. 4), may use a telephone
 terminal to access his own voice mailbox in the voice mail system 404 in
 FIG. 8 (374 in FIG. 4). This may be accomplished by dialing a directory
 number associated with the voice mail system 404 or 374 for this purpose.
 The voice processing unit 125 of the voice mail system 120 in FIG. 5 may
 operate its voice menu to direct the caller to depress a specified key
 when satisfied with the message, in a known fashion. The voice processing
 unit may then query the caller as to whether he desires to send the
 message and, if so, to depress another specified key. The voice unit then
 will instruct the caller as to the procedure for keying in the directory
 number of the destination and to depress a further specified key to send
 the message. This foregoing procedure is not intended to be exclusive and
 other procedures for leaving and commanding the dispatch of a message
 which are described in the background patents discussed above may be
 utilized.
 The message spoken by the user into the telephone creates an analog signal
 which is digitized in conventional fashion and stored in the mailbox of
 the party sending the message, i.e., in the voice mail system 404 or 374
 in FIGS. 8 and 4, respectively. The caller may then go on-hook after
 depressing the designated send key. The depression of the send key causes
 the voice mail system 404 to send a signal to the Internet interface 408
 and initiate the transfer of the message to the interface. The message
 sent to the interface contains the directory numbers of the intended
 recipient and the sending party along with routing and handling
 information. This may be affixed to the message in the storage process in
 accord with the type of service being requested, i.e., mailbox to mailbox
 transfer in this case.
 The Internet address of the connection of the remote telephone network or
 company 402 is retrieved from a database associated with the telephone
 network or company 400. The database may reside in an SCP or ISCP for the
 network 400, in the voice mail system 404, or in an intelligent peripheral
 in the network. This internet address information is forwarded to the
 interface 408 with the message to be delivered. The interface 408 acts in
 router fashion to encapsulate the message and address information in
 TCP/IP format and dispatch the same to the destination Internet address
 with an appropriate routing label and handling instructions. These
 handling instructions direct the addressee telephone network 402 to
 retrieve from its appropriate database the identity of the addressee and
 to verify its subscription to a mailbox. The transmitted message is then
 stored in the subscriber mailbox provided for the addressee directory
 number with an appropriate address. As with the previously described
 embodiments of the invention the addressee mail system is so designed as
 to inherently handle any translation in protocol which may be necessary.
 When the message packets reach the destination telephone system and have
 been deposited in the mailbox of the addressee, the voice mail system 406
 effects customary notification of the mailbox proprietor that a message is
 waiting. The proprietor may then access the mailbox in conventional
 fashion and have the message delivered as an audio or audible voice
 message in the usual fashion. If the addressee so desires and programs his
 voice mail service the message may be delivered by autodialed call to the
 addressee. Alternatively the telephone company may offer such delivery as
 part of the basic service.
 The party receiving the message may then have the option of returning a
 message in a converse fashion by depressing the appropriate keys at his
 telephone terminal. This utilizes the information in the packet header to
 reverse the origination and destination identifications and sends the
 reply back in the same fashion in which it was delivered.
 According to another embodiment of the invention a system and method is
 provided for transferring voice mail or messages to called parties who are
 not voice mail subscribers and thus do not possess individual or personal
 mailboxes or Internet addresses. Pursuant to this embodiment of the
 invention, voice mail systems 404 and 406 in FIG. 8 (374 and 376 in the
 network illustrated in FIG. 4) are provided with multiple unsubscribed
 mailboxes, which are here described as public mailboxes or mailboxes for
 temporary hire. It will be understood that such mailboxes may constitute
 mere addresses which may be appended to messages stored in the voice mail
 system storage. Such mailboxes may be utilized according to one embodiment
 of the invention in the following fashion.
 The involved telephone networks have established therein in known fashion a
 real or virtual directory number to trigger this embodiment of service.
 Dialing such a number at a terminal connected to the sending network
 establishes a connection to a voice mail system which is local to the
 calling party and which has public or for hire mailboxes as described.
 This may be a public mailbox in the local voice mail system 400 in FIG. 8
 (374 in FIG. 4). The caller is queried as to the destination directory
 number. This may be followed by an inquiry to establish whether the caller
 requests a reply. The caller is then requested to speak the message and to
 correct the same if necessary. The voice processing unit of the voice mail
 system may then operate its voice menu to direct the caller to depress a
 specified key when satisfied with the message in a known fashion. Billing
 information is provided to the caller and suitable billing signaling is
 effected, as by use of the SMDI link 378 to the local voice mail system
 374 in FIG. 4.
 Following storage of the digitized voice message and digitized instructions
 regarding delivery and response, the digitized message is transferred via
 the Internet 414 to the destination telephone network which is indicated
 by the destination directory number. The message arrives at the
 destination telephone network by virtue of its Internet address. This
 address is determined by a database search at the source telephone network
 as previously described. When the message arrives at the destination
 telephone network with header instructions for mailbox storage in the
 central messaging system of that network, the network and messaging system
 conduct a database search to confirm that the destination number does not
 have a subscribed mailbox. The message is then tagged and stored in an
 addressable public mailbox. This mailbox may be in the remote voice mail
 system 406 in FIG. 8 (376 in FIG. 4), where the digitized message and any
 accompanying instructions are stored. Delivery of the message is then
 effected in the same manner as previously described. Any reply from the
 recipient is first stored in a public mailbox in voice mail system 406 or
 376 and subsequently transferred through the Internet to the originating
 voice mailbox. The reply is then delivered to the original calling party
 by a telephone call to the originating telephone station or by retrieval,
 as desired.
 It will be seen that this embodiment of the invention vastly enlarges the
 economically feasible scope of mailbox-to-mailbox service by providing
 virtually worldwide coverage limited only by the availability of mailbox
 facilities at the source and destination. It is not necessary that both or
 even one of the communicating users be subscribers to mailboxes on a
 continuing basis. The invention permits such customers to access
 communication services of the Internet with no knowledge thereof nor
 possession of or access to computer terminals and related equipment. Both
 the deposit of the message and the delivery thereof may be completely
 analog using only the simplest of telephone terminals.
 Architecture of Systems Using Multimode Intelligent Peripherals (IPs)
 Referring to FIGS. 10 through 13 there is shown another embodiment of the
 invention for providing improved diverse telephone network service in
 combination with the Internet.
 FIG. 10 illustrates an integrated Advanced Intelligent Network (AIN) in a
 telephone network providing voice and data communications connectivity. In
 a typical situation, a local telephone operating company (TELCO) would
 deploy, operate and maintain such an integrated network, which may be
 considered representative of a type of network which may be used in the
 clouds 400 and 402 of FIG. 8.
 In the network shown in FIG. 10, each central office switching system (CO)
 511, 513, 515, 517 is labeled as an SSP. These Service Switching Points
 are appropriately equipped programmable switches 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 the
 illustrated embodiment, the CO-SSPs are end offices.
 As shown in FIG. 10, all of the end office switches 511, 513, 515 and 517
 are equipped and programmed to serve as SSPs. The illustrated embodiment
 is perhaps an ideal implementation which would make a variety of Advance
 Intelligent Network AIN services widely available at the local office
 level throughout the network. Other AIN implementations provide the SSP
 functionality only at selected points in the network, and end offices
 without such functionality forward calls to an SSP switching office having
 tandem switching capabilities.
 SSP capable central office switching systems typically consist of a
 programmable digital switch with CCIS communications capabilities. The
 structure of an exemplary CO which may serve as the SSP type COs in the
 system of FIG. 10 will be discussed in more detail below, with regard to
 FIG. 11.
 With reference to FIG. 10, the SSP type COs 511 and 513 connect to a first
 local area STP 523, and the SSP-COs 515 and 517 connect to a second local
 area STP 525. The connections to the STPs are for signaling purposes. As
 indicated by the circles below STPs 523 and 525, each local area STP can
 connect to a large number of the SSP-COs. The central office SSPs are
 interconnected to each other by trunk circuits (illustrated in FIG. 10 as
 bold lines) for carrying telephone services.
 The local area STPs 523 and 525, and any number of other such local area
 STPs (not shown) communicate with a state or regional STP 531. The state
 or regional STP 531 in turn provides communications with the ISCP 540. The
 STP hierarchy can be expanded or contracted to as many levels as needed to
 serve any size area covered by the Advanced Intelligent Network (AIN) and
 to service any number of stations and central office switches. Also,
 certain switching offices within the network, whether SSPs or not, may
 function primarily as tandem type offices providing connections between
 trunk circuits only.
 The links between the central office switching systems (COs) and the local
 area STPs 523 and 525 are typically SS7 type CCIS interoffice data
 communication channels. The local area STPs are in turn connected to each
 other and to the regional STP 531 via a packet switched network. The
 regional STP 531 also communicates with the ISCP 540 via a packet switched
 network.
 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 as previously described. Each query and response includes data
 fields for a variety of different pieces of information relating to the
 current call. For example, an initial TCAP query from the SSP includes,
 among other data, a "Service Key" which is the calling party's address.
 TCAP also specifies a standard message response format including routing
 information, such as primary carrier ID, alternate carrier ID and second
 alternate carrier ID and a routing number and a destination number. The
 TCAP specifies a number of additional message formats, for example a
 format for a subsequent query from the SSP, and formats for "INVOKE"
 messages for instructing the SSP to play an announcement or to play an
 announcement and collect digits and a "SEND TO RESOURCES" message to
 instruct the SSP to route to another network node.
 As shown, the ISCP 540 includes a Service Management System (SMS) 541, a
 Data and Reporting System (DRS) 545 and the actual database referred to as
 the Service Control Point (SCP) 543. The ISCP also typically includes a
 terminal subsystem referred to as a Service Creation Environment or SCE
 542 for programming the database in the SCP 543 for the services
 subscribed to by each individual customer. These components of the ISCP
 540 communicate with each other via a token ring network 544.
 The SCP database 543 stores data tables used to control telephone services
 provided through the network to callers using telephone stations. The SCP
 543 also stores at least some data for controlling data services through
 the integrated network.
 FIG. 11 is a simplified block diagram of an electronic program controlled
 switch which may be used as any one of the SSP type COs in the system of
 FIG. 10. As illustrated, the CO switch includes a number of different
 types of modules. In particular, the illustrated switch includes interface
 modules 551 (only two of which are shown), a communications module 553 and
 an administrative module 555.
 The interface modules 551 each include a number of interface units 0 to n.
 The interface units terminate lines from subscribers' stations, trunks, T1
 carrier facilities, etc. Where the interfaced circuit is analog, for
 example a subscriber loop, the interface unit will provide analog to
 digital conversion and digital to analog conversion. Alternatively, the
 lines or trunks may use digital protocols such as T1 or ISDN. Each
 interface module 551 also includes a digital service unit (not shown)
 which is used to generate call progress tones.
 Each interface module 551 includes, in addition to the noted interface
 units, a duplex microprocessor based module controller and a duplex time
 slot interchange, referred to as a TSI in the drawing. Digital words
 representative of voice information are transferred in two directions
 between interface units via the time slot interchange (intramodule call
 connections) or transmitted in two directions through the network control
 and timing links to the time multiplexed switch 557 and thence to another
 interface module (intermodule call connection).
 The communication module 553 includes the time multiplexed switch 557 and a
 message switch 559. The time multiplexed switch 557 provides time division
 transfer of digital voice data packets between voice channels of the
 interface modules 551 and transfers data messages between the interface
 modules. The message switch 559 interfaces the administrative module 555
 to the time multiplexed switch 557, so as to provide a route through the
 time multiplexed switch permitting two-way transfer of control related
 messages between the interface modules 551 and the administrative module
 555. In addition, the message switch 559 terminates special data links,
 for example a link for receiving a synchronization carrier used to
 maintain digital synchronism.
 The administrative module 555 includes an administrative module processor
 561, which is a computer equipped with disc storage 563, for overall
 control of CO operations. The administrative module processor 561
 communicates with the interface modules 551 through the communication
 module 553. The administrative module 555 also includes one or more
 input/output (I/O) processors 565 providing interfaces to terminal devices
 for technicians such as shown at 566 in the drawing and data links to
 operations systems for traffic, billing, maintenance data, etc. A CCIS
 terminal 573 and an associated data unit 571 provide a signaling link
 between the administrative module processor 561 and an SS7 network
 connection to an STP or the like (see FIG. 10), for facilitating call
 processing signal communications with other COs and with the ISCP 540.
 As illustrated in FIG. 11, the administrative module 555 also includes a
 call store 567 and a program store 569. Although shown as separate
 elements for convenience, these are typically implemented as memory
 elements within the computer serving as the administrative module
 processor 561. For each call in progress, the call store 567 stores
 translation information retrieved from disc storage 563 together with
 routing information and any temporary information needed for processing
 the call. The program store 569 stores program instructions which direct
 operations of the computer serving as the administrative module processor.
 Although shown as telephones in FIG. 10, the voice grade type terminals can
 comprise any communication device compatible with a voice grade type
 telephone line. Although all of the links to the telephone stations are
 illustrated as lines, those skilled in communications arts will recognize
 that a variety of local transport media and combinations thereof can be
 used between the end office switches and the actual telephone stations,
 such as twisted wire pairs, subscriber loop carrier systems, radio
 frequency wireless (e.g., cellular) systems, etc.
 In accord with this embodiment of the present invention, one or more
 Intelligent Peripherals (IPs) are added to the network to provide
 auxiliary call processing capabilities. As shown in FIG. 10, two of the
 SSP type central offices 513 and 517 connect to Intelligent Peripherals
 535 and 537, respectively. In the preferred embodiment, the IPs each
 connect to the associated SSP switch via a primary rate Integrated
 Services Digital Network (ISDN) link through an appropriate interface unit
 in one of the interface modules 551 of the switch (see FIG. 11). The ISDN
 link carries both voice and signaling data. The IPs also connect via a
 packet switched data communication network, such as X.25, to the ISCP 540.
 The X.25 data communication network forms a second signaling network
 separate from the SS7 network and the network of trunk circuits
 interconnecting the switching offices.
 In an Advanced Intelligent Network (AIN) type system, such as shown in FIG.
 10, certain telephone calls receive specialized AIN type processing under
 control of data files stored in the SCP database 543 within the ISCP 540.
 In such a network, the SSP type local offices of the public telephone
 network include appropriate data in the translation tables for customers
 subscribing to AIN services to define certain call processing events
 identified as AIN "triggers". Using the translation table data from disc
 memory 563, the SSP will detect such triggering events during processing
 of calls to or from such AIN service subscribers.
 The SSP type switches can recognize a variety of events as triggers for
 activating a query and response type AIN interaction with the ISCP. A
 number of different AIN triggers are used, depending on the precise type
 of service the AIN will provide a particular subscriber. For example, if a
 subscriber has a speech responsive autodialing service, an off-hook
 immediate trigger may be stored in the translation table file for that
 subscriber in the SSP. The SSP would detect the trigger each time the
 subscriber goes off-hook on that line and then attempt to obtain further
 instructions from the ISCP.
 For ordinary voice grade telephone service calls, there would be no event
 to trigger AIN processing; and the local and toll office switches would
 function normally and process such calls without referring to the SCP
 database for instructions. In a first mode of operation, an SSP type
 office (CO or tandem) which detects a trigger will suspend call
 processing, compile a TCAP formatted call data message and forward that
 message via a common channel interoffice signaling (CCIS) link and STP(s)
 to the ISCP 540 which includes the SCP database 543. The ISCP accesses its
 stored data tables to translate the received message data into a call
 control message and returns the call control message to the office of the
 network via CCIS link and STP(s). The SSP then uses the call control
 message to complete the particular call through the network. For AIN calls
 requiring a processing feature provided by the peripheral platform, the
 call control message would instruct the SSP to route the call to the
 associated peripheral platform.
 In the network of FIG. 10, the ISCP 540 transmits a "SEND to RESOURCE" type
 TCAP message instructing an SSP, such as SSP 517, to access a resource and
 collect digits. This message identifies a particular resource, in this
 case an ISDN type voice channel to the associated peripheral announcement
 platform 537. Each time the ISCP sends such a "SEND to RESOURCE" message
 to an SSP, the ISCP concurrently sends a message through the X.25 data
 link to the associated peripheral announcement platform. This message
 tells the platform what message to play on the specified ISDN channel at
 that time. If the message announcement platform has a text-to-speech
 converter, the announcement could take the form of virtually any desired
 script.
 The IP 537 performs DTMF digit collection and voice announcement functions
 on telephone calls, for a wide variety of telephone services available
 through the network. According to this embodiment of the invention as
 discussed in more detail later, the IP may also offer voice recognition
 capabilities and may include various data communications means, e.g., for
 FAX mail services, E-mail services, etc., as well as voice services.
 The illustrated preferred form of this multifeature embodiment of the
 invention includes two signaling communications systems carrying data to
 and from the ISCP 540. The communications links of the first such
 signaling network appear in the drawing as dashed lines, and the
 communications links of the second such signaling network appear in the
 drawing as lines formed by parallel bars. The first signaling network
 provides communications between the ISCP 540 and the SSPs 511, 513, 515,
 517 and between the individual SSPs 511, 513, 515, 517. The second
 signaling network provides communications between the ISCP 540 and the IPs
 535, 537. More specifically, the SCP 543 connects to the SSPs via the SS7
 network and the STPs 525 and 531. For the second signaling communication
 system a router shown as a small rectangle on the ring 544 provides a
 two-way communication connection to a data network, for example an
 Ethernet (IEEE 802.3) type local area network, another token ring, or a
 mixture of token ring and local area network, etc., going to the
 individual IPs 535, 537. Other types of high speed data network can be
 used between the ISCP 540 and the IPs 535, 537. Typically, the second
 signaling network will provide higher capacity data transport than the
 first signaling communication network.
 One IP may connect to one SSP. Alternatively, an IP may connect to two or
 more switching systems, or two or more IPs may connect to the same
 switching office. For example, in the illustrated network, the IP 535
 connects to two SSP type central office switching systems 513, 515. The IP
 537 connects to one SSP number central office switching system 517. The
 precise number of IPs in the network and the number there of connected to
 different switching systems is determined by projected traffic demands for
 IP service features from the subscribers, lines connected to the various
 switching systems.
 In the preferred embodiment, the connection from the IP to the SSP would
 utilize a primary rate ISDN type trunk line for carrying both voice
 channels and signaling information. However, a number of alternate
 implementations of this connection can be used. For example, the
 connection may take the form of a T1 circuit carrying a number of
 Multiplexed Centrex line channels. If additional data signaling is
 necessary from the switch to the IP, a Simplified Message Desk Interface
 (SMDI) link can be provided. SMDI is a standard form of maintenance port,
 available on many types of telephone switching systems, through which
 calling party number information can be supplied. For older switching
 systems not capable of call transfer through ISDN signaling or signaling
 on T1 Centrex lines, an additional switch could be added between the IP
 and the SSP.
 The AIN topology illustrated in FIG. 10 is exemplary in nature, and other
 network topologies can be used. For example, the illustrated networks
 include SSP functionality in each of the end office switching systems. In
 some networks, at least some of the end offices may not have SSP
 capabilities. Each such end office would connect to a trunk which in turn
 feeds calls to a tandem switching system with SSP capabilities. The SSP
 tandem communicates with the ISCP, as in the implementation described
 above. For the SSP capable end office switches that may be present in the
 network, they communicate directly with the ISCP, in the same manner as in
 the embodiment of FIG. 10. In such networks, each peripheral announcement
 platform or IP could connect to one or more of the non-SSP end offices,
 one or more SSP capable end offices and/or to the SSP capable tandem. The
 SSP capable tandem office switch is a digital switch, such as the 5ESS
 switch from AT&T; and the non-SSP type end offices might be 1A analog type
 switches.
 FIG. 12 illustrates a first, preferred embodiment of the IP used in the
 network of FIGS. 10 and 11. In this implementation, the IP consists of two
 or more general purpose computers 1101A, 1101B, such as IBM RS-6000s. Each
 general purpose computer will include a digital voice processing card for
 sending and receiving speech and other audio frequency signals, such as an
 IBM D-talk 600. Each voice processing card will connect to a voice server
 card 1103A or 1103B which provides the actual interface to T1 or primary
 rate interface ISDN trunks to the SSP type switching office. The plurality
 of computers may have associated dedicated disk storage 1105A, 1105B, and
 the IP will include a shared disk memory 107.
 Each computer will also include an interface card for providing two-way
 communications over an internal data communications system, an Ethernet
 type local area network 1109. The Ethernet carries communications between
 the individual computers and between the computers and a router which
 provides an interconnection to the second signaling communications network
 going to the ISCP. A router 1111 connected to local area network 1109
 provides a two-way coupling of the IP to the second data network, for
 example an Ethernet (IEEE 802.3) type local area network, a token ring, or
 a mixture of token ring and local area network, etc., at least for
 communications to and from the ISCP 540. If the X.25 network 220 serves as
 the signaling network between the ISCP and the IPs, then only one such
 router connected to that network would be included within the IP.
 The IP may also include another general purpose computer 1115 configured as
 a terminal subsystem, for use as a maintenance and operations center (MOC)
 and providing operations personnel access to the IP. The number of
 processors provided in the IP and the number of voice servers will depend
 on project service demands. One additional processor and associated voice
 server will be provided as a backup (not shown).
 Each general purpose computer 1101A, 1101B will run a node manager, an
 IP/ISCP Interface program, appropriate voice processing software and a
 variety of application software modules to offer the proposed services of
 the IP. The central administrator or "Node Manager" program module,
 running on each computer, will monitor and control the various IP
 resources and operations.
 The digital voice processing card and associated software will provide
 speech synthesis, speech recognition capabilities and DTMF tone signal
 reception, for use in a number of different applications. The speech
 synthesis and DTMF tone signal reception, for example may replace the
 announcement and digit collection functions of the SSP switches in various
 existing AIN services. These functions can also be used to permit
 subscribers to input parameters relating to Internet services and a
 variety of other types of service program modules, for example a voice
 mail server module and/or a fax mail server module.
 FIG. 13 illustrates an alternate embodiment of the IP used in the network
 of FIG. 12. The alternate architecture utilizes separate modules for
 different types of services or functions. By way of example, this version
 of the IP may have one or two Direct Talk type voice server modules 1203A,
 1203B for interfacing the trunk to the SSP, a separate speech recognition
 module 1205, a server module 1209 for voice mail, a server 1207 for fax
 mail services, a server 1215 for E-mail and E-mail type services, and an
 Internet interface, router or gateway module 1237 for Internet connection.
 The speech recognition module 1205 preferably provides both speech-to-text
 and text-to-speech conversion capabilities, i.e., it may perform both
 speech recognition as well as speech synthesis. The various modules
 communicate with one another via an internal data communication system
 1210, which again may be an Ethernet type local area network (LAN).
 The Direct Talk modules 1203A, 1203B provide voice message transmission,
 dialed digit collection and autodialing capabilities, as in the earlier
 embodiment. The Direct Talk modules also include processor control
 capabilities as will be described. The modules 1203A, 1203B provide line
 interfaces for communications to and from those servers which do not
 incorporate line interfaces. For example, for facsimile mail, the Direct
 Talk module connected to a call may demodulate incoming facsimile data and
 convert the data to a digital format compatible with the internal data
 communication network or LAN 1210. The data would then be transferred over
 network 1210 to the fax server 1207. For outgoing facsimile transmission,
 the server 1207 would transfer the data to one of the Direct Talk modules
 over the network 1210. The Direct Talk module would reformat and/or
 modulate the data as appropriate for transmission over the ISDN link to
 the SSP and provide any desired out dialing. The Direct Talk modules
 provide a similar interface function for the other servers, such as the
 voice mail server 1209 and E-mail server 1215.
 The illustrated IP also includes a communication server 1213. The
 communication server 1213 connects between the data communication system
 1210 and the router 1211 which provides communications access to the
 second signaling communication system and the ISCP 540 and other IPs which
 connect to that signaling communication system. The communication server
 1213 controls communications between the modules within the IP and the
 second signaling communication system and manages certain module-to-module
 communication within the IP. The SSP switch routes voice grade telephone
 calls to the different elements of the IP in response to instructions from
 the ISCP.
 In the initial implementation using general purpose computers (FIG. 12),
 each of which offers all service functionalities, the decision to route to
 a particular one of the computers would be a resource
 availability/allocation decision. If necessary data can be exchanged
 between the computers via the internal data communications network, e.g.,
 if a message for a particular subscriber's service is stored in the disc
 memory associated with one computer but the other computer is actually
 processing the call.
 In the second implementation (FIG. 13) the ISCP may instruct the SSP to
 route the call to the Direct Talk module or server with instructions for
 further routing within the IP to the specific module capable of providing
 a calling customer's requested service. For example, if the subscriber has
 some form of speech recognition service, the call may be routed to the
 speech recognition module or resource 1205 within the IP. If the
 subscriber has a voice mail service, however, the ISCP may instruct the
 SSP to route the call to one of the lines going to one of the voice server
 modules 1203A, 1203B and thence to the voice mail module 1209. The modules
 1203A or 1203B may receive outgoing voice messages from the voice mail
 server 1209 for transmission to the caller. The modules 1203A or 1203B may
 decode DTMF signals and supply appropriate data to the voice mail server
 for control purposes. The modules 1203A or 1203B will also format incoming
 voice messages for transmission over internal network 1210 and storage by
 voice mail server 1209.
 Communications between the IP and the ISCP alternatively may utilize
 generic data interface (GDI). The GDI command set is simpler and more
 generic, and the commands can carry more data. Also, either the ISCP or
 the IP can initiate communications using GDI. This permits a wider variety
 of routing and processing routines. Again using a voice telephone call as
 an example, in response to a triggering event, the SSP may again receive
 instructions to route a call in progress to the IP. However, rather than
 waiting for a subsequent query from the IP, while the SSP is routing the
 call the ISCP may instruct the IP to prepare to receive a call on a
 particular circuit. For example, for a call which might require speech
 recognition processing, the ISCP may instruct the IP to retrieve
 appropriate recognition templates from memory.
 As outlined briefly above, the IP 537 performs a variety of functions on
 AIN type voice grade calls, in addition to the control functions relating
 to the Internet services.
 The operation of the system of the invention is now described in terms of
 examples of its multifaceted modes of operation.
 EXAMPLE 1
 Referring to FIGS. 10 and 13, a caller desiring to use the new service
 dials from a telephone station such as A in FIG. 10 or 1239 in FIG. 13 a
 predetermined advertised directory number. The receiving SSP (511 in FIG.
 10 and 1241 in FIG. 13) recognizes this number as requiring a TCAP query
 to the ISCP 540 via intervening STP(s). The ISCP returns a Send to
 Resource message to the SSP directing the SSP to route the call to the
 Intelligent Peripheral (IP) 535. Simultaneously, the ISCP may dispatch a
 data message over the second data network to the IP communications server
 1213. In response to its instructions from the ISCP the SSP 1241 connects
 the calling telephone station 1239 to one of the IP Direct Talk servers
 1203A or 1203B. The IP communications server 1213 pursuant to its
 instructions from the ISCP directs the Direct Talk server 1203A to play a
 series of information gathering prompts and collect digits which are keyed
 in by the caller in response to the prompts. These prompts query the
 caller as to the service desired, collect information, and provide
 information. The information obtained by the Direct Talk server will
 include information as to the type of input desired, the type of delivery
 desired, and destination address information. The destination information
 comprises at least the destination telephone directory number and possibly
 a facsimile number and/or an E-mail address.
 In the case where the caller indicates a desire to send a voice message for
 voice delivery, the Direct Talk server 1203A and IP communication server
 1213 in response to these instructions direct the voice mail server 1209
 to handle the message. This voice mail server may be of the type described
 in detail in connection with the previously discussed embodiment of the
 invention and the voice mail system illustrated in FIG. 5. The voice mail
 server and the caller then interact in the manner previously described to
 deposit or store in the voice mail server the message to be transmitted.
 Following the caller pressing the "send" key, as directed in the voice
 prompt interchange, the message is transferred from the voice mail server
 1209 to the Internet interface 1237 and thence to the Internet 414, in the
 manner described in detail in connection with the above-described voice
 mail embodiment of the invention.
 The message thus dispatched is transferred via the Internet to the
 telephone system represented by the destination Telco cloud shown, for
 example, in FIG. 8. This telephone system is similar to that illustrated
 in FIGS. 10 and 13. The message arrives from the Internet cloud 414 in
 FIG. 13 at the Internet interface 1237 at the destination or receiving
 Intelligent Peripheral (IP) 535. The received TCP/IP Internet message
 incorporates in its header the necessary addressing and instructions for
 delivery. These instructions are decoded in the recipient Internet
 interface 1237 and IP communications server 1213 and the incoming message
 delivered to the voice mail server 1209 for storage and subsequent
 delivery, as described in connection with the previously discussed
 embodiments of the invention. The delivery may be effected from the voice
 mail storage in voice mail server 1209 via the LAN 1210 to one of the
 Direct Talk servers 1203A or 1203B. The selected Direct Talk server
 connects to an SSP in the destination or recipient telephone network.
 Connection is made from that SSP to the destination directory number to
 connect to telephone station 1239.
 EXAMPLE 2
 In the case where a caller indicates a desire to input a voice message for
 delivery by facsimile, the initial sequence may be as previously described
 in connection with Example 1. However, in this instance the requested
 voice to facsimile communication requires that the speech be recognized
 and converted to text. Thus, the Direct Talk server 1203A and the IP
 communication server 1213, as a result of the interactive exchange and
 direction, instruct the voice recognition resource 1205 to handle the
 message. As previously described this resource is capable of
 speech-to-text as well as text-to-speech conversion. It is also provided
 with storage and message handling capabilities of the type described for
 the voice mail system or resource illustrated in FIG. 5. When the message
 is voice inputted and stored in a form acceptable to the caller and the
 caller presses the "send key," the stored voice message is processed by
 the voice recognition resource and translated to text. The text is stored
 in text form in the voice recognition resource 1205. This text message is
 then transferred from the voice recognition resource 1205 via the LAN 1210
 to the Internet interface 1237 and thence to the Internet in the manner
 previously described.
 When the message is received at the destination telephone network it is
 processed through the Internet interface to that network into the protocol
 of the IP LAN 1210. The address and instruction information encapsulated
 in the message is handled by the IP communications server and the text
 message is directed to the facsimile server 1207. The facsimile server
 also is provided with input and output storage and processing capabilities
 analogous to those provided in the voice mail system of FIG. 5. The
 facsimile server buffers the text message and translates it to facsimile
 protocol, including the facsimile address information and instructions for
 handling. The message is then delivered to one of the Direct Talk servers
 1203A or 1203B.
 The selected Direct Talk server outdials the facsimile address directory
 number into the SSP 1241. Upon connection being established the facsimile
 message is delivered from the SSP to the recipient facsimile machine 1243
 which is connected to the destination telephone network. The facsimile
 server 1207 and/or Direct Talk server incorporate a conventional redial
 functionality wherein the destination number is periodically redialed if
 the initial attempt is unsuccessful. Also, while the facsimile server 1207
 is preferably provided with its own input and output storage, it is
 possible to utilize for this purpose the storage available in the voice
 mail server 1209. In this instance, the IP communications server,
 facsimile server and voice mail server will direct the appropriate flow of
 message and control information between those servers over the LAN 1210.
 As a still further alternative, a multipurpose in and out storage (not
 shown) may be provided for the IP and connected to the LAN 1210 to handle
 necessary storage for all connected servers.
 EXAMPLE 3
 In the situation where the caller indicates a desire to deposit a voice
 message for delivery as an E-mail type message, this information is
 obtained through the initial interactive voice prompt and speech
 interchange previously described. This type of delivery also requires
 speech recognition and speech-to-text conversion. Thus the caller's
 message is obtained and translated to text by the voice recognition
 resource 1205 as described in connection with foregoing Example 2. In this
 case the text is delivered from the speech recognition resource 1205 to
 the E-mail server 1215 for translation into E-mail style format. The same
 in-storage and out-storage occurs in the E-mail server and the translated
 E-mail type message is delivered to the Internet interface 1234. The
 message is then delivered via the Internet to the destination telephone
 system in the case of E-mail type messages. However, in the situation
 where the sending party provides an Internet Email address, the message
 which was inputted as a voice message may be delivered directly to the
 addressee by the Internet.
 Upon receipt of the E-mail type message at the destination telephone
 system, the message is forwarded from the recipient Internet interface to
 a Direct Talk server. The Direct Talk server connects to the destination
 local loop and destination PC, such as the PC 1245.
 While the translation from the text protocol which results from the
 speech-to-text translation in the speech recognition resource was
 translated into E-mail protocol by the E-mail server 1215 at the sending
 or originating end before traversing the Internet, the translation to
 E-mail protocol may occur at the receiving end, in the case of E-mail type
 messages. Thus, the product of the speech-to-text conversion in the
 sending IP may be conveyed directly to its Internet interface. The message
 is then sent via the Internet to the destination telephone network. At the
 destination network the output of the Internet interface may be fed to the
 destination IP E-mail server for translation to the necessary E-mail
 protocol for delivery to the destination PC. It will be recognized in all
 instances that the LAN 1210 maintains its own internal protocol and
 encapsulates the various received and delivered protocols with the
 necessary translations in the respective servers.
 EXAMPLE 4
 In the preceding examples, the caller has expressed the desire to input the
 message as a voice message. The invention also entails a caller requesting
 input in the form of a data signal. This may be a facsimile input from a
 facsimile machine 1243 or an E-mail type input, such as an input from a PC
 1245. In such cases the desires of the caller are ascertained from the
 initial interactive exchange of voice prompts and spoken directions via
 the telephone terminal. The results of this interaction are signaled to
 the originating IP 535. This causes either the E-mail server 1215 or
 facsimile server 1207 to undertake handling of the message. Storage occurs
 in the selected server as previously described and the appropriate
 protocol message is delivered by the E-mail server or the facsimile server
 to the Internet interface. The input in this situation is from either the
 facsimile machine 1243 or the PC 1245 after the connection to the IP has
 been established.
 The message is then transmitted through the Internet to the destination
 telephone system wherein the recipient Internet interface delivers the
 message to a Direct Talk server and then to the destination facsimile
 machine or PC in the destination telephone network. Alternative to this
 procedure the originating or source IP may deliver the data signals from
 the facsimile machine or PC, as the case may be, through the Direct Talk
 server to the originating Internet interface. The signal is then delivered
 via the Internet to the destination Internet interface with appropriate
 header instructions for handling by the destination IP. Thus, the
 destination IP may direct the message to the E-mail server 1215 or
 facsimile server 1207 for processing prior to delivery to the destination
 Direct Talk server and its connection to the destination facsimile machine
 1243 or PC 1245. Such an Internet transmission may be used in preference
 to conventional facsimile-to-facsimile service where large distances are
 being traversed. In the case of true Internet E-mail messages delivery may
 be made directly by the Internet as previously described.
 The invention offers the advantage that the user has the option to have the
 facsimile or E-mail type messages delivered by voice if the user so
 desires. In this case the incoming text message in the receiving IP is
 sent to the text-to-voice function of the voice recognition resource. From
 there it is delivered over a Direct Talk server to the destination
 telephone station.
 It will be seen from the foregoing that the embodiment of the invention
 illustrated in FIGS. 10-13 provides to the public at large the ability to
 transfer messages over the long distances spanned by the Internet for
 input and delivery in any of a variety of formats, without the user of the
 service having access to the Internet or possessing the equipment needed
 to produce a facsimile or E-mail signal. Voice input is effective to
 achieve such an end through a voice prompt system with which the public is
 familiar. As with the earlier described voice mailbox embodiment of the
 invention, the calling party may request a reply, and this may be
 delivered by the receiving party in a similar fashion without the need for
 individual Internet access. This embodiment of the invention also makes
 possible the dispatch and delivery of facsimile or E-mail type messages
 through voice connection to the local telephone company or network.
 It will be readily seen by one of ordinary skill in the art that the
 present invention fulfills all of the objects set forth above. After
 reading the foregoing specification, one of ordinary skill will be able to
 effect various changes, substitutions of equivalents and various other
 aspects of the invention as broadly disclosed herein. It is therefore
 intended that the protection granted hereon be limited only by the
 definition contained in the appended claims and equivalents thereof.