Abstract: Network servers in a session initiation protocol (SIP) telecommunication network implement playback of announcements to end-users by embedding programming scripts defining how the announcements are to be played in a SIP message. In particular, the scripts may define the sequence in which a series of announcements are to be played, duration information relating to a playback length of the announcements, and repetition information defining how many times an announcement is to be repeated. By including a script in a single message, announcement instructions may be efficiently communicated in the network.

Abstract: The present invention provides a system and method for providing certified voice and/or multimedia mail messages in a broadband signed communication system which uses packetized digital information. Cryptography is used to authenticate a message that has been compiled from streaming voice or multimedia packets. A certificate of the originator's identity and electronic signature authenticates the message. A broadband communication system user may be provisioned for certified voice and/or multimedia mail by registering with a certified mail service provider and thereby receiving certification. The called system user's CPE electronically signs the bits in received communication packets and returns the message with an electronic signature of the called system user to the calling party, along with the system user's certificate obtained from the service provider/certifying authority during registration. The electronic signature is a cryptographic key of the called party.

Abstract: Disclosed are systems, apparatus, methods, and computer readable media for interfacing with a phone provider in an on-demand service environment. In one embodiment, a proxy server is in communication with a phone system. The phone system provides a phone call and is operated by the phone provider. The proxy server is configured to: receive information from the phone system regarding a phone event in relation to the phone call, and output a message including the phone event information to a data network. A client machine is in communication with the proxy server over the data network and in communication with a data provider separate from the phone provider. The client machine is configured to: receive the message from the proxy server, and display the phone event information in a first part of a user interface in a web browser. A second part of the user interface can display information received from the data provider.

Abstract: Systems and methods for centralized presence management of local and remote users are provided. In exemplary embodiments, presence management information for the local and remote users are determined. Permissions established for at least one presence management user are reviewed to determine an amount of access to the presence management information to provide to the presence management user. The presence management information is then provided to the presence management user based on the permissions.

Abstract: A telecommunication and multimedia management apparatus and method that supports voice and other media communications and that enables users to: (i) participate in multiple conversation modes, including live phone calls, conference calls, instant voice messaging or tactical communications; (ii) review the messages of conversations in either a live mode or a time-shifted mode and to seamlessly transition back and forth between the two modes; (iii) participate in multiple conversations either concurrently or simultaneously; (iv) archive the messages of conversations for later review or processing; and (v) persistently store media either created or received on the communication devices of users. The latter feature enables users to generate or review media when either disconnected from the network or network conditions are poor and to optimize the delivery of media over the network based on network conditions and the intention of the users participating in conversations.

Abstract: A method and arrangement for providing a ring-back presentation to a calling terminal after receiving a call setup request from the calling terminal (A) for a first communication session (308) with a called terminal (B). A second packet-based communication session (310) is established with the calling terminal, independent of the first session, and pre-defined media content is provided as said ring-back presentation (312) to the calling terminal over the second session. Thereby, ring-back presentations to waiting callers can be selected from a great range of media types, including visual media. A calling user can also control the playout or display of the ring-back presentation irrespective of when the call is answered.

Abstract: A telecommunication and multimedia management apparatus and method that supports voice and other media communications and that enables users to: (i) participate in multiple conversation modes, including live phone calls, conference calls, instant voice messaging or tactical communications; (ii) review the messages of conversations in either a live mode or a time-shifted mode and to seamlessly transition back and forth between the two modes; (iii) participate in multiple conversations either concurrently or simultaneously; (iv) archive the messages of conversations for later review or processing; and (v) persistently store media either created or received on the communication devices of users. The latter feature enables users to generate or review media when either disconnected from the network or network conditions are poor and to optimize the delivery of media over the network based on network conditions and the intention of the users participating in conversations.

Abstract: A system and method of updating automatic speech recognition parameters on a mobile device are disclosed. The method comprises storing user account-specific adaptation data associated with ASR on a computing device associated with a wireless network, generating new ASR adaptation parameters based on transmitted information from the mobile device when a communication channel between the computing device and the mobile device becomes available and transmitting the new ASR adaptation data to the mobile device when a communication channel between the computing device and the mobile device becomes available. The new ASR adaptation data on the mobile device more accurately recognizes user utterances.

Abstract: The automatic selection of interfaces can result in the optimization of interaction between entities. Such selection of interfaces can be applied specifically in the context of interactive voice response systems, and can allow the interface presented to a caller to be optimized based on factors including the caller's usage of the system.

Abstract: A computer-implemented system and method for generating audio content using a telephony system. The method includes receiving a telephone call initiation request through a computer implemented graphical interface, initiating a telephone call to a selected telephone number, recording at least a portion of the telephone call to generate audio content, and posting the generated audio content on the computer implemented graphical interface to allow one or more users to access the generated audio content.

Abstract: A system and method for interfacing unified message processing systems with legacy voice mail, e-mail and facsimile systems, located behind corporate firewalls. The system includes a unified message server, a proxy interface and a message protocol convertor. The proxy interface is configured to access the legacy system in response to a request from a unified message server. Messages stored on the legacy system are converted by a protocol convertor to a predetermined format compatible with the unified message server. The converted messages are then transferred to a unified message server which is capable of providing messages from different messaging system, such as voice mail, e-mail and facsimile to users in a predetermined format. The invention permits enterprise wide communication systems to provide unified messaging without abandoning pre-existing legacy messaging system.

Abstract: Voicemail is received at a voicemail server and converted to an audio file format; it is then sent or streamed over a wide area network to a voice to text transcription system comprising a network of computers. One of the networked computers plays back the voice message to an operator and the operator intelligently transcribes the actual message from the original voice message by entering the corresponding text message (actually a succinct version of the original voice message, not a verbose word-for-word conversion) into the computer to generate a transcribed text message. The transcribed text message is then sent to the wireless information device from the computer. Because human operators are used instead of machine transcription, voicemails are converted accurately, intelligently, appropriately and succinctly into text messages (SMS/MMS).

Abstract: An exemplary method provides access to call origination features contained in an internet protocol multimedia subsystem (IMS) for a subscriber of the IMS who originates a call request from customer premises equipment (CPE) outside of the IMS. The call request is received at a public service identity (PSI) server in the IMS with a phone number to which the call request is routed. An IMS internal desk number is retrieved based on the number of the outside CPE. A first modified call request is transmitted from the PSI server to a telephony application server (TAS) identifying the IMS internal desk number as a request universal resource identity (RURI). Call origination treatment as specified by the subscriber is performed by the TAS.

Abstract: A third party call control (3PCC) application program interface (API) permits for users to use a web browser or other Internet capable software to place a call. The third party call control application program interface includes a first uniform resource locator operable over the Internet to effect a call between first and telephonic devices to be completed. The first uniform resource locator includes identification of the first telephonic device and identification of the second telephonic device.

Abstract: A network telephony system is provided which enables unified messaging services. The system generally includes at least one user agent operatively coupled to a data network and a signalling server operatively coupled to the data network. The user agents are telephony endpoints, such as standalone Internet telephony appliances or personal computers with appropriate telephony software. A messaging server is provided which is operatively coupled to the data network and is responsive to the signaling server. The system also includes a media server which is operatively coupled to the network and includes computer data storage media for storing message files. The media server is responsive to the messaging server and, on occurrence of a message condition, is directly accessible to a calling party to store a message file for subsequent retrieval by a called party.

Abstract: Disclosed are an apparatus and method of managing a multiple agents via a session client. One example method of operation may include selecting multiple endpoint devices to manage, establishing a connection to a server, and identifying at least two agent identifiers identifying multiple agents corresponding to the multiple endpoint devices. The method may also include transmitting a session setup command to the server including the at least two agent identifiers and receiving a confirmation that a session has been established between the session client and the multiple agents of the multiple endpoint devices.

Abstract: A context aware help system generates context data for web sessions. The context data may be used either by a human operator or by an automated response system to help resolve user web session problems. For example, a user may initiate a telephone call to the context aware help during a web session. The context data may be displayed to a telephone operator in response to the telephone call. The telephone operator may use the context data to determine what web pages and user inputs were previously entered during the web session. This allows the telephone operator to quickly analyze web session problems and provide more effective and efficient web session support. The help system may alternatively, or in addition, provide other types of automated audio or data support.

Abstract: A method of providing transports for a data distribution middleware over a plurality of transport networks is provided. A data distribution middleware with a pluggable transport layer is provided. A plurality of transport plugins in the transport layer are provided. Aliases are assigned to each of the transport plugins of the plurality of transport plugins, wherein at least one of the transport plugins of the plurality of transport plugins has a plurality of aliases.

Abstract: An Electronic Document Answering Machine is adapted to connect to remote information sources and to download digital documents including e-mail and WEB pages, and to store the documents for later review by a user. The Answering machine has an alert device for alerting a user that new documents are available for review, and an initializing device for a user to cause documents to be communicated for review one-at-a-time. In a preferred embodiment documents are communicated over a speaker via voice synthesis. In an alternative preferred embodiment a small display is provided wherein documents may be displayed in alphanumeric characters. The answering machine can be implemented as a stand-alone device or can be connected via a serial port to a host PC for transferring documents to the PC for processing and review. In other embodiments the Answering Machine is implemented as a system in a PC, with alert and selective input added as an interface.

Abstract: A mobile terminal and group chat controlling method thereof are disclosed, by which a chat content transceived in an performed group chat after a join-impossible state can be received in the course of performing the group chat if impossible to join the group chat. The present invention includes performing a group chat with a plurality of counterparts, displaying a chat window of the performed group chat, if the mobile terminal is impossible to join the group chat (i.e., if the mobile terminal enters a join-impossible state) in the course of performing the group chat, receiving chat contents transceived among a plurality of the counterparts in the performed group chat after the join-impossible state, and displaying the received chat contents.

Abstract: Embodiments of the present disclosure provide methods, systems, and devices of user notification. In this regard, one embodiment of such a method, among others, can be broadly summarized by the following: capturing an image in response to an event; relaying capturing of the image to a telephone device; and sending a message to a user at a remote location via the telephone device, the message notifying the user of the capturing of the image.

Abstract: The invention relates to the field of computer telephony integration. In particular the invention relates to displaying and producing computer based documents seamlessly to users that relate to their telephone calls. The best mode of the invention is considered to be the use of the invention with social media, such as Facebook, where the mobile station (30) searches for the social media page of the caller based on the Caller ID and displays it to the recipient of the call on the mobile station screen, possibly with other Internet search results. The methods and arrangements of the invention allow the user to obtain the newest public information about the person who calls him or he decides to call, or any documents that might be intermittent between the people. In other words the Desktop or screen of the invention automatically arranges itself to a state in which it is easiest to work during the call, using searches made based on call parameters, such as caller ID. The invention adds “bandwidth”, i.e.

Abstract: In one embodiment, a method for providing voice communications in a packet switched network protocol through a home network is provided, the method comprising: receiving, at a first home network device, an incoming call in the packet switched network protocol; notifying a second home network device of the incoming call; receiving an indication from the second home network device that the second home network device accepts the call; and forwarding the incoming call to the second home network device.

Abstract: A method of processing a data stream comprising audio data exchanged over a network between a server (SERV) and at least one telephone terminal, the data stream corresponding to a telephone call from said terminal during which a user has produced at least one event. The method comprises a step a) consisting in extracting from the stream audio data (INST2) corresponding to each event, and a step b) consisting in executing at least one task relating to the extracted audio data (INST2) and executable by a software application (AL), the software application being designed to interact with the network by using a data transmission protocol that is not specifically audio. The method further comprises a step c) of introducing into said software application (AL) at least one instruction (INST2?) relating to the extracted audio data (INST2) and adapted to activate the step b).

Abstract: An instant message associated with a conference call is received. At least some of the contents of the instant message are converted into speech, which may be represented as an audio stream. The speech is then provided to at least one participant associated with the conference call. The speech provided to the at least one participant may further include additional speech in addition to the speech corresponding to the contents of the instant message. The additional speech could, among other things, identify a participant who sent the instant message.

Abstract: A Uniform Resource Locator (URL), or a portion thereof, or an e-mail address are employed as auxiliary indicators of dialing of a voice call from a mobile phone to an URL- or specific e-mail address-associated part, with the part originating the call not identifying the telephone number of the other part An automated conversion is performed from an URL, or a portion thereof, or e-mail address, for an intended telephone number, or multiple intended telephones, in response to requirements started at the telephone system by the caller A part with presence in the Internet achieves a better availability with the public by means of voice telephone calls, since the URL's and e-mail addresses are convenient, known by, and significant to the clients.

Abstract: An Internet Protocol Multimedia Subsystem (IMS) includes a Serving-Call Session Control Function (S-CSCF). The S-CSCF includes a Service Capability Interaction Manager (SCIM) configured to invoke one or more application services in response to a message being received by the S-CSCF.

Abstract: An advertising-effectiveness determination method includes receiving a call made based on identification information provided to an information-providing medium in association with advertisement information. The call is connected to a call receiver indicated by the identification information. Input information input into a calling equipment of the call receiver and sent from the calling equipment is received. An effectiveness of the advertisement information is determined based on the received input information.

Abstract: Activities performed by a client at a site of an organization are monitored and stored in an activity database. A call is thereafter received from the client at a telephone system of the organization, which performs an identification of the client. The identification is provided to a cross-channel router as a request for a prediction on where the call from the client should be routed, and the router indexes into the activity database based on the identification of the client to obtain the monitored activity of the client. The router employs predicting rules and the obtained activity to generate a prediction of client needs, and returns the generated prediction to the telephone-based system, which presents prioritized telephone prompts to the client according to the returned prediction to help in routing the call.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a communication device having a controller to capture a voice recording, add the voice recording to a greeting to produce a personalized message, and deliver the personalized message upon scheduled delivery to at least one recipient based on a communication identifier of the at least one recipient, where the greeting includes at least one among audio, video, and image data, and where the communication device communicates with a notification server for creating the personalized message. Other embodiments are disclosed.

Abstract: Voice data from speech start till mute are transmitted as a section voice file to the opposite side voice data transmitting and receiving system, thus permitting voice data transmission free from voice interruption. By transmitting voice data from the speech start till the mute it is possible to eliminate packet losses in the data transmission channel and perform voice communication in the same voice quality as in the transmitting side. Besides, because of the use of HTTP communication port it is possible to obtain voice communication exceeding the rooter without any special setting.

Abstract: In system and method of prioritizing a voice mail message, a voice mail message is received comprising caller information and chronological information, a voice mail priority is associated with the first voice mail message based on at least one of a matching criteria, a call history criteria, a chronological criteria, a frequency metric, and a missed call value, and a voice mail list is ordered according to the voice mail priority.

Abstract: A schedule for a desired call, obtained, for example as an SMS message sent from a user's mobile station, enables a server to automatically initiate set-up of a voice call for the user's mobile station at a scheduled time selected by the user and indicated in the message. The user can specify time and a telephone number for the call, in advance, in the call scheduling request message. However, at the time of the call, the user does not need to read a reminder or dial the call. The call scheduling and automatic set-up services described by way of examples herein may be very useful, for example, for attending important conference calls. This may also be very useful for setting a reminder to call a special person to wish him/her well on his/her birthday/anniversary etc.

Abstract: Example systems and methods to manage location information for emergency services support of an internet protocol (“IP”) communication device are disclosed. A disclosed example method involves storing first and second geographic location information in a first list of geographic location information associated with an internet protocol device. The first list is stored at a service provider server. At least a partial copy of the first list is communicated from the service provider server to the internet protocol device for storing as a second list of geographic location information in the internet protocol device.

Abstract: A system and for exchanging information to enable embodiment IP communication comprises a first endpoint corresponding to a first user, a second endpoint corresponding to a second user, and an information exchange (PURLX) unit. The PURLX unit comprises: a processor, a table and a phone URL exchange engine. The processor is configured for communication over the communications channel with a first endpoint and a second endpoint. The table stores information about the first user and the second user including a first association between the first user and one or more endpoint device addresses and a second association between the second user and one or more endpoint device addresses. The table is coupled for communication with the processor. The exchange engine establishes IP communication between the first user and the second user using information from the table.

Abstract: The present invention discloses method, system and network server for recording use of network service capability by applications (APPs), and the key points of the methods comprise: a network server acquires an identity of the APP and type information of the network service capability providing services for the APP, conducts service logic processing, and writes a message of use of network service capability, which comprises at least the identity of the APP and the type information of the network service capability providing services for the APP. With the methods of this invention, the purpose of making effective records of use of network service capability by APPs provided by service providers is implemented. Furthermore, management of APPs, such as charging, may be conducted by utilizing the records.

Abstract: One preferred embodiment of the present invention provides a system and method for sending instant messages having audio recordings. Briefly described, in architecture, one embodiment of the system, among others, includes a communications client of a user that is configured to generate an instant message having an audio recording and send the instant message having an audio recording to another user. Methods and other systems are also provided.

Abstract: Disclosed are systems and techniques by which a caller may specify a callee's telephone number and be connected directly to a carrier provided voice mail facility associated with the identified the telephone number, even though the callee's carrier may not be the same as the caller's carrier. In the disclosed technique, a telephony server places a “Send a call” request to a server which then sends a signaling call that busies out the channel associated with the callee. The telephony server places a second call (the actual voice message) upon confirmation that the signaling call has been initiated, forcing the second call to the carrier's voice mail facility associated with the callee, since the first signaling call busied the first channel. Prior to sending the signaling call, a database look up is performed to determine which carrier services the callee number. Once the carrier is determined, another memory look up is performed to determine the time delay associated with that carrier.

Abstract: A call setup method and terminal in an IP network, in which in response to a request for an outbound call from a first terminal, it is identified whether or not an index matching an identification of a second terminal exists in a matching table. The identification of the second terminal is inputted at the request for the outbound call. If the index matching the identification of the second terminal exists in the matching table, a call initialization message is directly transmitted to the second terminal. In response to call setup, a call state synchronization message is transmitted to a call manager.

Abstract: In one embodiment, a scheme is disclosed for effectuating a call in a network environment including a circuit-switched (CS) network and an IP multimedia subsystem (IMS) network. Call information associated with a call is sent from a user equipment (UE) device to an application server (AS) node disposed in the IMS network. The call information includes at least one of a call reference number and a called party's URI. When a message is received at the UE device from the AS node, which message includes the call reference number and an IP multimedia routing number (IMRN), the returned call reference number is verified that it remains valid based on a local timer mechanism associated with the UE device. The IMRN is then sent to the application server in order to facilitate a session with respect to the called party.

Abstract: The present invention moves service control, including call control, for a user element from any number of subsystems to a multimedia subsystem (MS). Call control for originating or terminating a call in any subsystem as well as transferring the call between user elements is anchored at a continuity control function (CCF) in the MS. A call may be transferred by the CCF from a first user element to a second user element in the same or different subsystems upon receiving a request from the second user element. A user may invoke a transfer from the first user element to the second user element by having the second user element send an appropriate request to the CCF. The request may be provided to the CCF by initiating a call from the second user element that is intended for the CCF, when the call with the first user element is active.

Abstract: In a caller information notifier system, a calling terminal transmits a calling signal having a phone number for callback inserted which is different from the phone number of the calling terminal. When a callback is tried from a called terminal, a communication network server, where the calling signal passes, determines the relationship between the phone number of the calling terminal and the phone number for callback inserted in the calling signal to authenticate the phone number for callback. A terminal which receives the calling signal extracts the phone number for callback from the calling signal and keeps the extracted number as a callback phone number. The callback phone number is, for instance, an FMC service number. The probability of success in reaching a caller is thereby increased.

Abstract: A method for accessing and browsing the internet through the use of a telephone and the associated DTMF signals is disclosed. The preferred embodiment provides a system that converts the information content of a web page from text to speech (voice signals), signals the hyperlink selections of a web page in an audio manner, and allows selection of the hyperlinks through the use of DTMF signals generated from a telephone keypad. Upon receiving a DTMF signal corresponding to a hyperlink, the corresponding web page is fetched and again delivered to the user via one of the available delivery methods such as voice, fax-on-demand, electronic mail, or regular mail.

Abstract: Described is automatically testing the quality of an audio channel between a caller and a callee that includes a device under test, such as a VoIP or other gateway. An analyzer receives timestamps from a caller and callee during a calling session, including timestamps for when the callee initially provides audio (e.g., speech) to the caller, when the caller initially detects sound, when the caller initially provides audio to the callee, and when the callee initially detects sound. The analyzer uses the relative timing of the timestamps and the speech recognizer's outcome to determine whether the audio channel is experiencing interference or echo. When the audio includes speech, a confidence level corresponding to accuracy of speech recognition also may establish the audio channel's quality. Random selection and timing of output may be employed, such as to vary the testing patterns during repetitive tests.

Abstract: An apparatus and methods for utilizing a stand-alone Voice over Internet Protocol (VoIP) telephone with traditional plain old telephone service (POTS) telephone support is disclosed. A stand-alone VoIP telephone with an analog telephone adaptor supporting a plurality of external analog telephone sets and a digital Internet protocol (IP) interface to support VoIP voice data communication over a data channel is presented. The VoIP telephone is used to establish VoIP service line data connections between a VoIP telephone and a data switch to an IP network. The VoIP telephone receives an analog telephone signal from an external analog telephone and establishes a voice connection between the external analog telephone and the VoIP telephone.

Abstract: Embodiments of the invention provide an enhanced telephone system. The telephone system comprises a database that comprises phone numbers and menus corresponding to the phone numbers. Further, the menus comprise options for selection. The telephone system comprises means for comparing a dialed number to the phone numbers in the database, displaying a menu based on a result of the comparison, enabling selection of the options from the displayed menu, and establishing a connection with a destination of the dialed number based on the selection of the options.

Abstract: The present invention relates to telephone call processing. In one embodiment, a call processing system receives from a caller a first call for a user, the first call including call signaling information. The call processing system determines whether the call signaling information includes caller identification information and whether the call signaling information indicates that the caller identification information is restricted. If the caller identification information is restricted, the call processing system instructs the caller to take a first action so that the caller identification information can be provided to the user. The call processing system determines whether the caller has performed the first action during the first call, and if the caller has performed the first action during the first call, the call processing system provides at least a portion of the caller identification information to the user.

Abstract: An approach is disclosed for providing an audio phone card. A telephone call is received for an audio service from a voice station by user. Audio content is transmitted to the user over the telephone call, wherein playback of the audio content is controlled by the user via the telephone call.

Abstract: Users in an enhanced communication system are enabled to tag other users so that the tagging user is notified when both the tagged and tagging users meet a predefined criterion of presence states and/or available modalities to the tagging and the tagged users. The tagging user may then, optionally, initiate a multimodal session with the tagged user utilizing the set of modalities originally specified or another set.

Abstract: Disclosed are methods and apparatus for composing and communicating Digital Composition Coded Multisensory Messages (DCC MSMs). The present invention also relates to the design, composition, creation, and communication of the multisensory messages. Multisensory messages, while rich in content and meaning, are to be composable from a great variety of platforms, from cell phones to mainframes.