Abstract: A method and system are provided for opportunistically initiating a multimedia call between two or more devices such as smart speakers at a time when users of the respective smart speakers are available to participate in a call. A server system receives an instruction from a first smart speaker associated with a first user to establish a real-time communication session with a second smart speaker. In response to that instruction, the server system monitors availability of both the first user and the second user at their respective smart speakers. On determining that both the first user and the second user are available at the same time, the server system initiates the real-time communication session between the first smart speaker and the second smart speaker.

Abstract: Techniques for acoustic voice activity detection (AVAD) is described, including detecting a signal associated with a subband from a microphone, performing an operation on data associated with the signal, the operation generating a value associated with the subband, and determining whether the value distinguishes the signal from noise by using the value to determine a signal-to-noise ratio and comparing the value to a threshold.

Abstract: A computer receives a global command from a communication device using a first communication mode associated with a transaction. The computer stores transaction data, wherein the transaction data is stored for at most a duration of a session life span (SLS) time interval that is predefined. The computer sends a transfer session identifier (TSI) key to the communication device using the first communication mode. The computer receives the TSI key from the first communication device or a second communication device, using at least a second communication mode. The computer associates the transaction data with at least the second communication mode and the first communication mode in parallel, and the computer performs the transaction by use of either or both of the first communication mode and at least the second communication mode, within the SLS time interval that is pre-defined.

Abstract: Messages received by a mobile device can be processed in one of at least two modes. In the normal mode, the messages are handled in accordance with the normal settings of the mobile device. In the filtered mode, or “night-time” mode, incoming messages are examined to determine if they are to be filtered or if they are exempt from filtering. If a message is exempt from filtering, it is processed in accordance with the normal mode. However, if the message is to be filtered, the mobile device is prevented from providing normal message reception alert notifications. Messages can be identified as being exempt to being filtered by editing an address book or selecting entries from an address book. All messages not identified as exempt then get filtered.

Abstract: A system and method can support a virtual private network based on a public telecommunication infrastructure. The virtual private network can receive a request from a caller for place a call to a callee via the virtual private network. The virtual private network can obtain at least one of presence status and location information of the callee. Then, the virtual private network can allow the caller to make a decision on whether or not to proceed with the call. The public telecommunication infrastructure can be based on a legacy network and an Internet Protocol Multimedia Subsystem (IMS) network, and the virtual private network can obtain information from both the legacy network and the IMS network via a service broker.

Abstract: A system and method may include receiving an invite message associated with a calling device over a data network, the invite message requesting establishment of a voice over data communication session, and presenting a plurality of call rejection options, each of the plurality of call rejection options being associated with separate call rejection messages. The system and method may further include determining which one of the plurality of call rejection options is selected, where the plurality of call rejection options permit a called party to intelligently reject a voice over data communication session.

Abstract: The present invention provides a technique for providing a forced hold service such as is used for an emergency services call, which is supported at least in part over a packet network. The forced hold service acts to effectively hold a connection for the call with a called party, even when the caller takes an action that would normally end a call, such as going on hook, pressing end, or the like. When the caller takes an action that would normally end the call, the forced hold service allows the caller to automatically reconnect to the emergency services provider over the held connection upon going offhook, pressing send, or the like. Alternatively, the emergency services provider can effectively re-engage the call wherein the caller is reconnected over the held connection upon going offhook, pressing send, or the like.

Abstract: Send and receive services are operated that send and receive fax messages, respectively, via fax protocols. Each new customer is offered a predetermined amount of the receive service, and a more limited amount of the send service, both free of charge to the customer, as part of a group of services that includes Internet access, Web services, or Internet telephony. Other embodiments are also described and claimed.

Abstract: A configurable PSTN call-handling (CPC) service enables monitoring and custom responses to calls received over multiple subscriber lines. A CPC Client Program runs on a computer and presents a graphical user interface (GUI) that allows the user to associate call-handling options with call characteristics. The GUI also allows the user to customize the notification signals provided and to provide custom responses to certain callers, by associating call characteristics with various GUI appearances, sound files, or images. The CPC relieves the communications network of performing certain call-handling functions, such as call forwarding, by moving decision-making processes to the user's PC. The user can remotely monitor any subscriber line associated with the user from any browser, because the communications network sends notifications directly to an IP address specified by the user. The IP address associated with a computer connected to the subscriber line need not be the IP address specified by the user.

Abstract: A method for changing ICS session media includes: receiving a media type change request comprising a new media type sent from one of a terminal equipment, a mobile switching center (MSC), or a second party; releasing a circuit-switched (CS) call leg based on an original media type between an IMS CS control function (ICCF) and the terminal equipment; establishing a CS call leg based on the new media type between the ICCF and the terminal equipment; and updating the media type of a second call leg between the ICCF and a second party into the new media type or establishing a second call leg based on the new media type between the ICCF and the second party.

Abstract: The present invention provides flexible call processing. In one embodiment, an inbound call from a calling party is received at a call processing system. The inbound call is intended for a called party. A determination is made that a first condition has occurred and/or is occurring. The called party's location is determined via a location based service associated with the mobile device of the called party. Based at least in part on the called party's mobile device location, a phone number stored in computer readable memory is selected. The calling party is connected to a communication device associated with the selected phone number.

Abstract: The disclosed embodiments include a method of initiating a call. The method includes receiving, at a network device over a first network, an identifier associated with a called party. The method determines a first security level associated with the first network and identifies a plurality of networks associated with the called party using the identifier. The method queries a database to determine a second security level for each network in the plurality of networks and automatically selects an identified network from the plurality of networks associated with the called party having the second security level at least equal to the first security level. The method completes a call to the called party using the identified network.

Abstract: The present invention provides a technique for providing a forced hold service such as is used for an emergency services call, which is supported at least in part over a packet network. The forced hold service acts to effectively hold a connection for the call with a called party, even when the caller takes an action that would normally end a call, such as going on hook, pressing end, or the like. When the caller takes an action that would normally end the call, the forced hold service allows the caller to automatically reconnect to the emergency services provider over the held connection upon going offhook, pressing send, or the like. Alternatively, the emergency services provider can effectively re-engage the call wherein the caller is reconnected over the held connection upon going offhook, pressing send, or the like.

Abstract: At least one telephone network element receives an attempt of a caller to place a telephone call to a callee. A telephone number for the call is registered in a telephone number mapping (ENUM) database to receive incoming telephone calls via a Voice over Internet Protocol (VoIP) network. The at least one telephone network element receives an indication of a preference of the caller to carry the telephone call via one of a public switched telephone network (PSTN) and a public land mobile network (PLMN), instead of via the VoIP network. Based on the preference, the method inhibits an ENUM query of the ENUM database for the telephone call and forwards the telephone call to a gateway to carry the telephone call to the callee via one of the PSTN and the PLMN instead of via the VoIP network.

Abstract: A method for displaying speed dial settings is disclosed. The method is used for a mobile communication device, comprising the steps: detecting whether a speed dial setting operation is implemented to a button of a keypad of a mobile communication device, and displaying an indicator for the button if the speed dial setting operation is detected.

Abstract: Included are embodiments for detecting presence in a call control protocol environment. At least one embodiment of a method includes subscribing to events associated with a communication system and receiving a notification for an event, the event being associated with a communication between a first communications device and a second communications device.

Abstract: Directory services schema for telephone call management. The schema facilitates the development of call routing rules as one or more instances of one or more classes. Once created, the rules can be propagated to all other director services servers, and any subsequent updates and system changes are readily propagated to maintain optimum call system management. The rules can be stored in a routing table allowing administrators to define phone routes for various scenarios. System hardware/software changes can be managed using distinguished names and automatically propagated (or replicated) to the other directory services servers on an enterprise network. The architecture includes a schema component for generating a directory services schema, and a call routing component that routes telephone calls according to call routing rules defined by the directory services schema. The rules can be processed in combination with policies of usage attributes and phone number patterns/ranges to manage call routing.

Abstract: Methods and apparatus to provide voice control of dial tone and audio message in a telephone in an initial off-hook period. A telephone system has a telephone with a receiver having an off-hook position. The telephone system provides a user with control of a telephone function. The telephone system includes an audio output to provide an audio message to the receiver in the off-hook position. The audio message includes an option for playing the audio message or a dial tone the next instance the receiver is in the off-hook position. A control interface receives a selected option in response to a user command.

Abstract: A method and apparatus of optimizing transitioning between EVDO and CDMA 1X systems in a hybrid access terminal, the method having the steps of preventing a reconnect data call from being sent from the hybrid access terminal. The preventing step including checking whether a transition between an EVDO and a CDMA 1X system is successful; and if not blocking the data call, and/or checking whether an EVDO received signal strength indication is greater than or equal to a receiver sensitivity on the hybrid access terminal; and if yes, allowing the hybrid access terminal to connect to the EVDO system, and/or delaying a connection timer for connecting to the EVDO system and/or applying fast dormancy to the hybrid access terminal in an EVDO state.

Abstract: A system (10) for controlling telecommunications calls includes a voice XML network gateway (12) including a voice interpreter module (20) and a call center server module (28) association with a telecommunications switch (58). The voice interpreter module (20) receives voice telecommunications signals, and the call center server module (28) receives call center telecommunications data signals. Interpreting circuitry (22, 24) interprets the voice telecommunications signals using the voice interpreter module in association speech recognition application (s) (16). Call center service providing (18) means provides call center service in response to the call center telecommunications data signals in association with call center application program(s).

Abstract: An intermediary communication system, the intermediary communication including: (a) a first network interface, configured for transmitting over a first network connection to a first remote end unit a first sound sequence; and for receiving from the first remote end unit a returning sound sequence that is responsive to the first sound sequence; (b) a processor, configured to determine an echo reduction parameter in response to a relationship between a first sound sequence parameter and a returning sound sequence parameter; and (c) a second network interface, for transmitting to a second remote end unit, over a second network connection, a processed sound sequence that was generated in response to the echo reduction parameter from a preprocessed sound sequence which was generated by the first remote end unit.

Abstract: A system to enable access information stored on a terminal device (cellular phone) to be utilized without imposing a load on a user when performing communications from a communication device (MFP). An address book on the cellular phone is opened (S101, S102), and a name of a communication destination and the access information are selected by use of the address book (S103 to S105). vCard-formatted data is created based on the content selected (S108), and the created data is transmitted to the MFP (S109). The MFP analyzes the vCard-formatted data received from the cellular phone (S201, S202), and the communications are performed by use of the access information contained in the vCard-formatted data (S203 to S216).

Abstract: The invention relates to a telephone call routing system including means for associating calling numbers with geographical areas and means for associating geographical areas with destination numbers. The invention is characterised in that the calling numbers are associated with geographical areas using statistical methods with a certain minimum reliability threshold. The invention is suitable for use in the field of value-added telecommunications services, call routing and network intelligence.

Abstract: A system and method for automatically classifying lists of telephone numbers into one or more predetermined categories is provided. The system initiates calls to each telephone number. A specific request is made of the callee, which confirms the number as being “live-answered.” The system uses speech recognition software to compare audible sounds received on the other end of the line to a plurality of spoken messages to classify each telephone number that has been classified as “not live-answered” for future use or exclusion. One embodiment of the system records each telephone call, enabling the process to be performed using a single call to each telephone number.

Abstract: A method and system for mitigating the impact of latency in the initiation of a packet-based real-time media session. When an initiating station is setting up a requested session, the initiating station will generate and play out a repetitive tone pattern (e.g., ring tone) for the initiating user to hear. The tone pattern is predefined in the initiating station, rather than being received from the network during session setup. The repetitive tone pattern preferably relaxes the initiating user and reduces or prevents “dead air” during session initiation.

Abstract: A method and system for a dial plan parser includes a state machine configured upon a dial plan to which is applied call signaling data, and a call routing module/switch to transfer a call to a next point.

Abstract: The invention relates to the representation, independent of access type, of ISDN accesses (PRA, BRA) in the central core (cVst) of a packet-oriented exchange (P-Vst). By adapting the signaling information transmitted by ISDN accesses in a peripheral unit (PE) it is possible to handle different ISDN access types as a uniform access type in the central core (cVst) of the packet-oriented exchange (P-Vst). The adaptation can be performed for example with the aid of a conversion table for the user data references and call references contained in signaling messages. The invention has the advantage of providing a simple, low-cost, low-overhead representation of ISDN data in the central core of the packet-oriented exchange (P-Vst). Changes such as e.g. a concentration of the user data channels can be carried out easily and flexibly by the network management (NetMg).

Abstract: Network initiated auto redialing is based on an auto redial duration selected or entered by a calling party. An auto redial interval or retry rate may also be selected or entered by the calling party. If the calling party is unavailable when a connection is established with a called party, a message is played or sent to the called party explaining that the call is an auto redial call. The message may explain that the calling party is now unavailable. Additionally, the message may include identifying information regarding the calling party. For example, the message may include a directory number of the calling party.

Abstract: A Voice over Internet Protocol phone (VoIP phone) includes a VoIP module coupled to an external network, a receiving module for receiving data stored in a portable memory device, and a control means coupled to the VoIP module and the receiving module for controlling the VoIP module according to the data stored in the portable memory device received by the receiving module. The VoIP phone can be utilized in public places without risk of compromising the security of personal information.

Abstract: A DTMF generating device for automatically generating DTMF tones corresponding to a predefined phone number comprises a processor, a memory, and a speaker enveloped in a credit card sized casing. The memory has stored therein DTMF tone samples corresponding to the predefined phone number. When a user presses on a portion of the casing, the DTMF tone samples are played by the speaker.

Abstract: Systems and methods are disclosed for performing HyperText Transfer Protocol (HTTP)-like transactions over a point-to-point connection between a client and a server. In one embodiment, the client uses Uniform Resource Locators (URLs) that identify the server by a telephone number that it can be reached at. When a user tries to access such a URL, the telephone number is dialed and a point-to-point connection is established (if it does not already exist) between the client and the server associated with that telephone number. HTTP-like transactions can then take place between the client and server, without the necessity of an Internet Protocol network with gateways, domain name servers, search engines, and network-connected hosts.

Abstract: An electronic apparatus and a multi-functional telephone apparatus in which system data can be set efficiently using plural pictures irrespective of whether the setting is new A relevant information table unit 141 includes a menu table 142 for demonstrating a menu for setting a multi-functional telephone apparatus on a display, not shown, plural data-related tables 143 for demonstrating constituent elements for each setting and a system data table 144 for demonstrating the contents of respective system data. The three tables 142 to 144 are related to one another by a pointer 145. Thus, a desired item is selected from the menu, and a desired SDID is selected from the groups displayed in the data-related tables 143 lying on its lower order side to set corresponding system data.

Abstract: A system and method for monitoring remote computer access and associated costs is provided. The system includes a remotely located communication server in communication with multiple host computer networks and in communication with a network access server. First and second memory devices contain a list of authorized users for the host computer networks and a user log for use by a billing computer to generate bills. The method includes the steps of creating starting and ending time stamps for each authorized user accessing a respective one of the multiple host computer networks and creating a user log to generate bills and monitor host computer network usage.

Abstract: The present invention provides a method and system for providing web audio content directly from audio processors to a telephone without requiring storage of the audio on a media server. In one embodiment, a switch and two stages of switching are used. The switch is coupled between at least one audio source and a network interface controller. A direct access controller is coupled to the audio sources, switch and network interface controller. The direct access controller establishes a first audio channel through the switch in a connection phase. The direct access controller establishes a second audio channel through the switch in an audio transport phase. In the audio transport phase, web audio content is transported directly from a remote web server to an audio source on the second audio channel and then from the audio source to the user of the telephone on the first audio channel.

Abstract: There is provided methods and devices for relaying tone events in a communication system. A first tone data indicative of a first tone is received and buffered. Following, a second tone data indicative of a second tone is received and buffered. The first and second tones can be, for example, DTMF digits. A processor retrieves the first tone data from the buffer and generates the first tone corresponding to the first tone data. The first tone is generated for at least a first predetermined period, which can be approximately 50 msec, for example. After the first tone has ended, the processor waits for a second predetermined period before retrieving the second tone data from the buffer. The second predetermined period can be, for example, 50 msec. Subsequently, the second tone corresponding to the second tone data is generated for at least the first predetermined period of time.

Abstract: A technique for determining the last digit of a dialed sequence. Different interdigit time-out periods are pre-assigned to each digit received. If the time-out period expires, it is considered that a full complement of digits has been received, whereupon the received digits are then transferred toward the destination.

Abstract: An operating surface for CLASS functions of a telephone device of an analog network. The operating surface has two functions, “activate” and “deactivate”, so that the command sequence following the “activate” function is interpreted as CLASS function and is turned on and the following command sequence is interpreted as CLASS function and turned off with the “deactivation” function.

Abstract: A call error prevention method comprises the steps of inputting a call number, checking whether the input number is a recognizable call service code, determining whether the input number contains the same number of digits as a recognized call service code, and generating an input error warning if the input number does not contain the same number of digits as the recognized call service code.

Abstract: A system is provided for automatically dialing a number stored in an electronic organizer on a telephone. The system comprises an electronic organizer including memory for storing a plurality of telephone numbers therein, an input device for selecting one of the telephone numbers, and a communicator for transmitting digits of the selected telephone number. The system further includes telephone circuitry adapted to dial digits of the selected telephone number upon the receipt thereof. Further provided is an interface including a communicator in communication with the communicator of the electronic organizer for receiving the digits therefrom and transmitting the same to the telephone circuitry.

Abstract: A method and system for providing a dialed digit unlock and edit feature for a digital telephone connected to an advanced intelligent network (AIN) wireline network is disclosed. The digital telephone sends at least one digit of a called party number to an originating ISDN switch. The originating ISDN switch determines whether a Q.931 CALL PROCEEDING call control message has been sent from the originating ISDN switch to the digital telephone. If so, then the dialed digit unlock and edit feature is disabled. However, if the Q.931 CALL PROCEEDING call control message has not been sent, then the originating ISDN switch determines whether a key to enable the dialed digit unlock and edit feature has been entered on the digital telephone. If the key has been entered, then the last digit(s) entered for the called party number are erased and the originating ISDN switch awaits a new digit(s) to be entered into the digital telephone.

Abstract: A method and system for providing a dialed digit unlock and edit feature for an analog telephone connected to an advanced intelligent network (AIN) wireline network is disclosed. The analog telephone sends at least one digit of a called party number to an originating central office. The originating central office determines whether an initial address message (IAM) has been sent from the originating central office to a destination central office. If so, then the dialed digit unlock and edit feature is disabled. However, if the IAM has not been sent, then the originating central office determines whether a key to enable the dialed digit unlock and edit feature has been entered on the analog telephone. If the key has been entered, then the last digit(s) entered for the called party number are erased and the originating central office awaits a new digit(s) to be entered into the analog telephone.

Abstract: The present invention concerns a method to perform a telecommunication between a caller terminal and a callee terminal identified by an identification code, such as a phone number.

Abstract: For controlling a call in a telecommunication system, a call is established from a terminal to a first server, where said call is associated with a network browser. Said network browser is used to locate and download a stored file and the contents of the downloaded stored file are interpreted. The interpreted contents of the downloaded stored file are used to control said call.

Abstract: A WLL system having personal stations, either mobile stations or telephone sets with an NIU, a base station and a switching center. For faster connectivity, a method in accordance with the present invention includes the steps of (1) if the personal station is in an off hook state, the personal station transmits an origination message to the base station and assigns a traffic channel between the personal station and the base station, (2) the personal station transmits a destination telephone number to the base station, and (3) the switching center connected to the base station connecting a call to a destination personal station according to the destination telephone number, thereby reducing a delay time in setting up a call, and thereby providing a fast communication service.

Abstract: A custom telephone apparatus and a method of manufacturing same are provided. In addition to conventional telephone features, the custom telephone includes a memory that is permanently pre-programmed with one or more area codes according to a particular geographic region or as requested by a particular user. The telephone includes permanently labeled area code selector switches or buttons that, when depressed by a user, cause the telephone to dial the area code and any required leading “1” or other digits as a prefix to the telephone number entered by the user by way of a conventional key pad. Thus, the user need only depress a single button to dial “1” plus the area code before a telephone number is entered into the telephone. The memory can also be permanently programmed with one or more complete custom telephone numbers as specified in advance by a particular user.

Abstract: A method and device for decoding a sequence of signals in which information is derived, in part, from a duration of (a) one of the signals and/or (b) an inter-signal pause between two of the signals. The method and device may also be used to decode a signal to form decoded information, derived, in part, from a duration of the signal.

Abstract: A method of transmitting a telephone number in a wireless local loop (WLL) system is provided for allowing a user to instantly hear a ring back tone by immediately performing a dialing in a WLL terminal when a flash signal is input by the user operating a hook switch after a telephone number is dialed in an ordinary telephone connected to the WLL terminal. The method checks the state of a hook switch of the ordinary telephone connected to the WLL terminal. When an off-hook sate is determined, it is determined whether a telephone number is input. When it is determined that a telephone number is input, the telephone number is stored in a dial buffer. It is continuously checked whether a flash signal is input while the input telephone number is being stored in the dial buffer. When the flash signal is input, the telephone number stored in the dial buffer is transmitted to the WLL system base station.

Abstract: A system for generating number tones for dialing a telephone device comprising a microphone. The system comprises a housing, a first data entry device, first and second memory devices, an output signal generator, and a transducer. The housing defines a sound opening. The first data entry device is supported by the housing and is associated with a first sequence of stored digits. The first memory device stores the first sequence of stored digits. The second memory device stores frequency data associated with number tones. The output signal generator generates, upon activation of the first data entry device, an output signal based on the first sequence of stored digits and the frequency data. The transducer is mounted within the housing adjacent to the sound opening and converts the output signal into an audible signal. The housing is adapted to be attached to the telephone device with the sound opening adjacent to the microphone.

Abstract: A system and method are provided for special signaling with customer premises equipment. The system includes loop termination equipment coupled to a local loop and comprising a signal generator. The system also includes customer premises equipment coupled to the local loop. The customer premises equipment and loop termination equipment are operable to communicate across a communication link established on the local loop. The loop termination equipment is further operable to control characteristics of a signal generated by the signal generator and to provide the signal to the customer premises equipment as special signaling where the signal has a frequency that does not substantially interfere with voice traffic on the local loop. The special signaling can be used to implement enhanced service to the customer premises including busy signal, queued data request processing, push data and telephony services.

Abstract: Embodiments provide a method, article of manufacture, and apparatus for customizing a telephone calling card operation. In one embodiment, the calling card may be customized to redial a last number by entering a pre-defined shortcut sequence. In another embodiment, a search tool is used to search for a particular number using one or more pre-stored address books.