Abstract: A system and method for diminishing false positive classifications of unsolicited electronic-mail is disclosed. An electronic-mail message is received whereby a set of message classifiers are applied the message. One or more signatures indicative of unsolicited electronic-mail based on at least the application of the aforementioned classifiers is generated. The one or more signatures are applied to subsequently incoming electronic-mail messages whereby unsolicited electronic-mail may be more accurately identified and false positive identification of normal electronic-mail messages are reduced.

Abstract: A method of providing originator information to an intended communications recipient during establishment of a communication session includes reading, at an originating communication terminal, originator information carried by a device located with the originating terminal, and including at least some of the originator information in at least one signal transmitted by the originating communication terminal during establishment of the communication session for transmission to a communication terminal associated with the intended communications recipient.

Abstract: A system and method are disclosed for classifying a message. The method includes receiving the message, identifying in the message a distinguishing property; generating a signature using the distinguishing property; and comparing the signature to a database of signatures generated by previously classified messages.

Abstract: According to one aspect, the subject matter described herein includes a method for SDF based subscription profile repository selection. The method includes receiving a first signaling message. The first signaling message includes service data flow information for identifying a service data flow. The method also includes selecting, using the service data flow information, an SPR from a plurality of SPRs, the SPR containing policy information associated with the service data flow.

Abstract: Providing non-voice capabilities relating to a phone call at a computing device includes receiving a request to begin a telephone call from a first telephone to a second telephone, wherein the requesting is performed using messaging between computing devices connected via a computer network, and wherein the messaging is part of a signaling phase dialog. The signaling phase dialog may be used in providing information associated with the identity of parties of the telephone call. Commencing a call phase dialog between the computing devices may be performed after the telephone call is established over the public switched telephone network. The call phase dialog may allow messages relating to the telephone call to be passed between the computing devices, at least while the telephone call is in progress. The dialogs and messaging may be performed using session initiation protocol (SIP) or another computer communication protocol.

Abstract: A method, a device and a system for media session negotiation are disclosed in the present invention. The method includes: receiving a first media description sent from a telephony server (TS), where the first media description includes a media session parameter of the TS at a local area network LAN side (101); forwarding the first media description to a telephony client (TC) (102); receiving a second media description which is generated by the TC according to the first media description, where the second media description includes a media session parameter of the TC (103); and forwarding the second media description to the TS (104). Through the present invention, negotiation of media session parameters between a TS and a TC in a home network may be achieved.

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 apparatus for playing a multimedia ring tone during a call, where the method includes: receiving an on-hook message from a called terminal; sending a message to a calling terminal and the called terminal to update a session; and instructing a Multimedia Resource Function (MRF) to execute mixing processing to continuously play a multimedia ring tone during the call. Accordingly, a problem in the prior art that a multimedia ring tone cannot be continuously played for a user during a call is solved.

Abstract: The invention relates to methods and a high speed communication device that allow one of a plurality of high speed communication devices connected to a transmission line having a normal impedance to effectively receive data. The high speed communication device can include a transmission line interface connected to the transmission line, a receiver connected to the transmission line interface, and a transmitter selectively coupled to the transmission line interface. The transmitter can have an impedance substantially equal to the normal line impedance. The high speed communication device can present a high impedance to the transmission line with respect to the normal line impedance when the transmitter is not coupled to the transmission line interface, and the high speed communication device can present an impedance to the transmission line that is substantially equal to the normal line impedance when the transmitter is coupled to the transmission line interface.

Abstract: Methods and systems of authorising a user of a first packet-based communication network to access a second packet-based communication network are disclosed. A call setup request is received from a user terminal of the user at a first network element of the first packet-based communication network, the authorisation request comprising a first user identity. Responsive to the call setup request, a request is transmitted to create a second user identity from the first network element to a second network element of the second packet-based communication network. The second network element creates the second user identity for use in the second packet-based communication network. The second user identity is derivable from the first user identity according to a predetermined rule. The second user identity is stored in the second packet-based communication network for use with establishing a call associated with the call setup request over the second packet-based communication network.

Abstract: A method performed by a device adapted for communication over a network includes establishing a presence on the network, obtaining default configuration information for the device via the network, the default configuration information identifying a service on the network, receiving a request for validation information from the service, providing the validation information to the service, the service generating device-specific information in response to the validation information, and obtaining the device-specific information.

Abstract: A method, system and computer instructions for users to visually rearrange telephone call trees are disclosed. A caller can connect to an automated telephone answering service, and download the service's call tree. The caller can view a display with the call tree, and modify or rearrange the order in which the menu options of the call tree are displayed. The user can then store the modified call tree for use in the future.

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 communication control device executes a call-out process of calling out a recipient telephone device while specifying an importance level via a telephone network in response to a request from a transmitter telephone device. A recipient received a call operates a mobile communication terminal after a telephone call ends, accesses an information providing server via a wireless LAN access point and the Internet, and inputs information evaluating whether or not an importance level specified for the received call is appropriate. The information providing server provides the input evaluation information to a storage device. The communication control device determines whether or not to execute a call-out process based on the evaluation information stored in the storage device when there is a call-out request from a transmitter.

Abstract: An apparatus comprising an artificial noise (AN) controller coupled to a digital subscriber line (DSL) transmitter and configured to adjust an AN level for a signal transmitted by the DSL transmitter, wherein the AN level is adjusted based on an actual signal to noise ratio (SNR) of the signal from the receiver to achieve a desired SNR for the signal. Also disclosed is a method comprising maintaining a data rate in a line at about a desired level by adjusting an AN level in a transmitted signal based on a SNR of a received signal.

Abstract: The invention relates to an instantaneous messaging system for mobile terminals comprising one or several instantaneous messaging service servers (S1, S2) able to deliver instantaneous messaging services to mobile terminals (TA, TB1, TC1, TD1, TB2, TE2, TF2) connected to said servers and of which the users (A, B1, C1, D1, B2, E2, F2) have opened an instantaneous messaging session, and a virtual presence server (100) independent of the existing instantaneous messaging service servers (S1, S2), said virtual presence server being connected to at least one mobile client terminal (TA) and to at least one of the existing messaging servers (S1, S2) in such a way that the information intended for or sent by said mobile client terminal transits by said virtual presence server, wherein the instantaneous messaging service servers (S1, S2) are configured to transmit contact lists intended for mobile client terminal (TA), said lists being intercepted by the virtual presence server (100), the latter comprising a means to a

Abstract: A communication system including a media server through which communication packets are exchanged for recording and monitoring purposes is disclosed. A tap is associated with each communication endpoint allowing for cradle to grave recording of communications despite their subsequent routing or branching. An incoming communication is routed to a first tap and upon selection of a receiving party; the first tap is routed to a second tap which forwards communication packets on to the receiving party. The taps may be used to forward communication packets to any number of other taps or destinations, such as a recording device, monitoring user, or other user in the form of a conference.

Abstract: A method of accessing voice services through a personal computing system can include receiving, in the personal computing system, a telephone call from a user registered with the personal computing system, wherein the user is remotely located from the personal computing system, receiving a user spoken utterance over the telephone call, and speech recognizing the user spoken utterance to determine a request for a voice service. The method further can include formatting an electronic message according to the request for a voice service and sending the electronic message over a communications network to a remote computing system in accordance with the request for a voice service. Also, the method can include processing and formatting a response received from the remote computing system and playing back audio to the user over the telephone call.

Abstract: A time authentication system provided can detect an alteration of the case, in which either of the speakers of a conversation makes the alteration, and can verify, in the case of no alteration, that a conversation voice has occurred at a time within a range. When a speaker #1 terminal (2) and a speaker #2 terminal (3), i.e., all the speakers of the conversation request a time authentication station server (1) for the issue of a unique background sound which is difficult for those other than the time authentication station server (1) to estimate beforehand, the time authentication station server (1) issues the background sound and an electronic verification, which is electronically signed by adding a time stamp to the hash of the background sound, to the speaker #1 terminal (2) and the speaker #2 terminal (3).

Abstract: A method and system are provided for preventing data loss in a VoIP system. In particular, during a VoIP call, it is determined whether incoming ringing on a POTS line causes an unacceptable level of signal loss or errors. If so, for subsequent VoIP calls, the CO handling calls to the POTS line is instructed to either answer each call with a busy signal or automatically forward calls to the POTS line to the VoIP line or other selected telephone. Calling returns to normal upon ending of the VoIP call. In this manner, incoming ringing on the POTS line does not result in call dropping or lengthy retraining processes.

Abstract: A relay apparatus for relaying communication between a plurality of wireless terminals and a wired network includes a priority data recording part for recording communication identification data and required bandwidth data, a priority control part for relaying, by priority, the communication recorded in the priority data recording part, a request receiving part for receiving communication identification data and multimedia type data, a selecting part for selecting a multimedia type to be relayed by priority, an updating part for additionally recording data that represents a bandwidth required for relaying a stream of the selected multimedia type and the communication identification data, and a responding part for transmitting a response. The selecting part selects the multimedia type so that a sum of bandwidths represented by the required bandwidth data does not exceed a predetermined upper limit value.

Abstract: A network for coupling at least one telephone service signal to at least one telephone device over a wiring. The network includes: a wiring having at least two conductors for carrying multiple time-domain multiplexed digitized voice channels; an exchange side device coupled to the wiring and operative to couple at least one telephone service signal to at least one digitized voice channel; and at least one subscriber side device coupled to the wiring and operative to couple the at least one telephone device to at least one digitized voice channel.

Abstract: There is described a provides a method for selecting a buddy with which a user wants to connect. The buddy having a buddy identification being listed in a pre-identified group of Internet-accessible destinations. The user using a user telephone means to which is associated identification information. The method comprises: initiating a call from the user telephone means to an Internet-enabled computing device, the call having associated thereto calling line identification (CLID) information; comparing the CLID information to the user telephone means identification information; validating the CLID information using the comparison upon positive validation, the user providing the buddy identification; consulting the pre-identified group of Internet-accessible destinations for a match between the buddy identification and one of the Internet-accessible destinations; and advising the user of a successful or unsuccessful match.

Abstract: Multiple instances of an incoming call in a unified communication system capable of communicating through multiple networks are provided to devices and/or applications associated with the called party. Upon acceptance or a global rejection of the incoming call through one of the end devices/applications, the remainder are provided a cancellation notice for the forked call request that includes information associated with which end device took the action, what type of action was taken, whether the action was based on automatic policy implementation.

Abstract: A system and method for processing a communication signal from a wireline is provided. The system comprises a coupling unit and signal processing circuitry. The coupling unit comprises at least one transformer and is configured to receive the communication signal from the wireline. The coupling unit is also configured to generate a first signal based on amplifying the communication signal according to a first winding ratio of the at least one transformer and a second signal based on attenuating the communication signal according to a second winding ratio of the transformer. The signal processing circuitry is configured to process the first signal and the second signal. The wireline is optionally a powerline.

Abstract: A highly scalable and distributed call/media modeling and control framework is described. In particular, a plurality of anchor point servers are provided which enable one or more applications to exert call and media control over in-progress communication sessions even though the applications were not originally bound to the communication session during its setup.

Abstract: A system for control of apparatuses, in particular of industrial apparatuses, comprising a plurality of interfaces (20), each dedicated to a connection with a respective apparatus (10) to be monitored, and a processing unit (40) to control the apparatuses (10); the processing unit (40) is provided with first receiving means (41) to receive at least one data item from an apparatus (10), a first selection block (42) to select one or more addressees, depending on the received data item, a second selection block (43) to select one or more communication channels (50) for connection with said addressees, and first transmission means (44) to send at least one notification signal representative of the received data to the addressees through the selected communication channel/channels (50).

Abstract: An exemplary computing system may receive a first request from a first voice-communication device to establish a secure-voice-communication session with a second voice-communication device. The computing system may also receive a second request from the second voice-communication device to establish the secure-voice-communication session with the first voice-communication device. The computing system may establish the secure-voice-communication session between the first and second voice-communication devices. Corresponding methods, apparatus, and computer-readable media are also disclosed.

Abstract: A method for providing voice telephony services can include the step of receiving a call via a telephone gateway. The telephone gateway can convey call identifying data to a resource connector. A media port can be responsively established within a media converter that is communicatively linked to the telephone gateway through a port associated with the call. A call description object can be constructed that includes the call identifying data and an identifier for the media port. The call description object can be conveyed to a telephony application server that provides at least one speech service for the call. The telephony application server can initiate at least one programmatic action of a communicatively linked speech engine. The speech engine can convey results of the programmatic action to the media converter through the media port. The media converter can stream speech signals for the call based upon the results.

Abstract: A method to prevent TTY/TDD (teletypewriter/telecommunications device for the deal) probing and unwanted TTY/TDD tone generation on voice gateways comprises detecting a PSTN textphone in a call path, if any; and providing visual clues to an end user, thereby avoiding generation of unwanted text by the end user. The visual clues may be in the form of a dialog box on a computer, warning messages on a phone, and/or the like.

Abstract: The present invention relates to the processing of E-mail messages over a telecommunications network. More particularly, the present invention relates to the detection and monitoring of file formats which can be processed and displayed at an E-mail client. Specific embodiments include, sending an E-mail message to the E-mail client, determining at the E-mail client a file format that the E-mail client can process and display, and indicating to the E-mail sensor server the file format that the E-mail client can process and display. Once the file format is determined, subsequent E-mail messages maybe of the same file format. The file format could be hyper text mark up language (HTML) statements or dynamic HTML(DHTML) statements or Java applets. The method may also include monitoring the status of the E-mail message received at the E-mail client.

Abstract: A system for handling a telephone call within a telephone network, comprising a multi-tenanted CTI server within a telecommunications network adapted to receive telephone calls on behalf of a plurality of outlets, wherein the CTI server is adapted for queuing an incoming call in the network when no agent is available and is further adapted to forward an incoming call to an agent when the agent becomes available. The CTI server is configured to use an indicia received with the received call to identify the caller and to retrieve a profile of the caller by matching the indicia to a previously-stored indicia in a call history database collocated with the CTI server, the profile comprising at least a plurality of previous orders processed by an outlet on behalf of the caller, the profile being used to provide priority call handling for certain callers.

Abstract: A method for establishing a multimodal telephone call is provided. On a first computing device, a telephone call is received from a second computing device. On the first computing device, a response is sent to the second computing device that a telephony session is established between the first computing device and the second computing device. On the first computing device, a request message is sent to register the first computing device for a data session at an online registry service. On the first computing device, a response message is received indicating that the first computing device is registered at the online registry service. The response message includes a key that uniquely identifies the data session. On the first computing device, the key is used to establish the data session with the second computing device.

Abstract: In a telephone conferencing system, a digital communication network such as the Internet is used to establish and control the telephone connections between multiple conferees with the telephone network being the means of exchanging verbal information. Each conferee may have a computer connected to the digital network, and each has an independent telephone instrument connected to the public switched telephone network. An in-charge conferee utilizes his computer containing appropriate software to initiate the conference and to control the participation of the conferees. The in-charge conferee sends digital control signals to a switch interface controlling a telephone switch as a gateway to the telephone network using SS7 control signals. These SS7 control signals include the commands by which the conferee telephones are rung up, brought on line, or dropped from the conference.

Abstract: Telephone calls, data and other multimedia information is routed through a hybrid network which includes transfer of information across the internet. A media order entry captures complete user profile information for a user. This profile information is utilized by the system throughout the media experience for routing, billing, monitoring, reporting and other media control functions. Users can manage more aspects of a network than previously possible, and control network activities from a central site. The hybrid network also contains logic for responding to requests for quality of service and reserving the resources to provide the requested services.

Abstract: Tags, such as XML tags, are inserted into email to separate email content from signature blocks, privacy notices and confidentiality notices, and to separate original email messages from replies and replies from further replies. The tags are detected by a system that renders email as speech, such as voice command platform or network-based virtual assistant or message center. For example, the system can detect the signature block or privacy notice tags and not render the signature block or privacy notice as speech. The system can render an original email message in one voice mode and the reply in a different voice mode. The tags can be inserted to identify a voice memo in which a user responds to a particular portion of an email message. Preferably, an email server that receives and stored the email message inserts the tags into the email. Alternatively, the tags could be inserted by an email client application.

Abstract: A SIP server 100 includes a first memory 20 storing a route management table in which, of route set information specified by a SIP signal on a session-by-session basis, address information as of each Record-Route header value is associated with a logical number, the same address information as of a Record-Route header values being set to the same single logical number; and a second memory 30 storing session information other than the address information as of each Record-Route header value, instance management information for use in session control, and logical numbers. The address information as of each Record-Route header value, which is stored in the first memory 20 is read on the basis of the logical numbers stored in the second memory 30.

Abstract: A telephone call handling system includes a sendings receiver configured to receive sendings sent from the telephones of other party's via a telephone service. A sendings registry automatically inventories each received sending and forwards the inventoried sending to a storage where the sending is stored. Subsequently, a delivery trigger is triggered. The triggered trigger initiates a sendings provider to provide one or more of the inventoried and stored sendings to a deliverer. The deliverer then delivers the provided sending or sendings to one or more of: a speaker, a display, a text message display, an image display, a video display, an email delivery, an web-site poster, a service provider delivery, a delivery port, a ring-back delivery, a ring-back forward delivery and a ring-tone.

Abstract: A communication manager establishes a call between two or more participants on two or more communication devices. The call can be an audio or video call. A call request is sent from one of the communication devices to conference an additional communication device to the call. For example, a participant in the call decides to conference an additional person to further discuss an idea. An information manager gets a roster of the participants. In addition to the call request, the roster of the participants is sent to the additional communication device. The additional communication device receives the call request and the roster. The roster is displayed to the user of the additional communication device. The user can then indicate to answer the call request. If answered, the additional communication device sends an indication that the call was answered and the additional communication device is conferenced into the call.

Abstract: Methods, systems, and products provide remote access to call logs upon request. An incoming call is processed and an address is retrieved that is associated with a called number. An electronic message is sent to the address to identify a calling number of the incoming call. A log of calls is updated with the incoming call. A reply electronic message is received from the address, and the log of calls is sent to the address.

Abstract: An integrated communication interface is provided for composing and sending messages. The interface is multi-configurable to seamlessly switch between different communication methods, e.g., electronic mail, instant messaging, SMS, chat, voice, and the like, without loss of message content. The interface allows a user to begin composing a message to be sent using one communication method, such as electronic mail, and subsequently change the communication method and send the message via a second communication method, such as instant messaging. When the communication method is changed, the user interface may also change to include elements specific to a particular communication method. The integrated communication interface may display information about participants in the communication, such as the participants' presence, i.e., whether they are online and available for communication, and may automatically choose the best method of communication based on the preferences and online presence of the participants.

Abstract: Systems and methods are provided for score-driven management of such recordings. A score is assigned to a recording based at least in part on a plurality of monitored events that are detected for the call being recorded, and such score is then used for managing the recording, such as managing storage and/or post-processing of the recording. A recording manager is provided, which is operable to perform score-driven management of a recording. According to one embodiment, an initial score is determined for a recording at the initiation of the call being recorded. Such initial score may be determined based on fixed factors that are known at the time of the call's initiation. Various events related to the call are monitored during the call, and the score is then revised based on detected events. Management of storage, post-processing, and/or other handling of the recording is then based on the score assigned thereto.

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 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 for transferring data from a first switch to a second switch selectively by line-switching or by packet-switching as well as to a switch for carrying out the method. Data packets are thereby first transferred packet-switched through a packet-switching network to the second switch. With the presence of a corresponding control signal a line-switching connection is established from the first switch to the second switch and the data are then transferred through this connection. Where applicable a renewed changeover to a packet-switching transfer is carried out. A flexible packet-switching or line-switching data transfer linked with dynamic costs between the junctions of a telecommunications network is enabled.

Abstract: The present invention is directed to a method for simultaneously rendering one or more conversations on a communication device. The method includes receiving at the first communication device media associated with the one or more conversations over a network and defining a set of conversations among the one or more conversations to be simultaneously rendered. The received media associated with the one or more conversations is simultaneously rendered regardless if the received media overlaps or not.

Abstract: A platform provides access to multiple telecommunications services via a single telephone number. These services may include, for example, voicemail services, facsimile messaging services, paging services, and outbound calling services. A guest caller may call the single phone number to speak to a subscriber, leave a voicemail message for a subscriber, leave a facsimile message for a subscriber, or place a page to a subscriber. A subscriber may call the phone number to place outbound calls. In addition, a subscriber may call the single phone number to retrieve or send facsimile messages and/or voicemail messages. A subscriber may also call the phone number to configure service options. For instance, a subscriber may select routing options and choose what services are available to guest callers.

Abstract: The invention relates to a method for generating a digit map for a call setup, wherein leading digits of a call number are collected by a gateway and transmitted to a call control server where a call-specific digit map is generated by comparing the leading digits to a numbering plan.

Abstract: A content recommendation system includes first and second communication devices which are connected to each other and in which content recommendation information is transmitted and received between both devices. The first communication device includes a storage section for storing pieces of content recommendation information to be transmitted, a selecting section for selecting a piece of content recommendation information to be transmitted from the storage section, and a transmitting section for transmitting the selected piece of content recommendation information. The second communication device includes a receiving section for receiving the piece of content recommendation information transmitted from the first communication device, and a storage section for storing the received piece of content recommendation information.

Abstract: A system is disclosed for delivering information, preferably electronic television program guide information, to a user over a telecommunications network. The network includes the capability of determining whether the user's connection to the network is in an on-hook or off-hook condition, and the information is received and stored at a user site when the user's connection to the network is in an on-hook condition. Information is delivered from a guide provider to a phone company switch point over a land-based or wireless dedicated line. When the user's phone is on-hook, the guide data is sent to the user site through a splitter. The line splitter provides the signal to a data decoder circuit, the output of which is connected to a data input terminal in communication with a television receiver, VCR, cable box, or other piece of equipment for displaying the program guide data on a screen.