Abstract: A method and apparatus to form an audio bridge session are described.

Abstract: Disclosed herein is a three-way call detection system and method for detecting the addition of a third party to a pre-existing telephonic connection between a first party and a second party.

Abstract: A call management technique implemented using a call routing engine is disclosed. In the technique, local (e.g., ACD) switching resources and a public network are controlled as a single virtual switch for purposes of implementing call processing features.

Abstract: Disclosed are systems and methods which detect unauthorized, impermissible, or otherwise undesired call activity through a silence detection technique employing multiple samples. One embodiment monitors a call in progress to detect an event such as might be indicative of unauthorized call activity. Thereafter, a sampling process may be invoked which analyzes multiple samples of the call to make a determination as to whether or not a particular unauthorized call activity is being attempted. A party to the call may be muted to thereby prevent that party from thwarting the unauthorized call activity determination. Embodiments implement techniques for defeating parties to calls learning the unauthorized call activity detection system in order to avoid detection, such as by implementing a plurality of detection techniques.

Abstract: Discloses are systems and methods for detecting undesired call activity. Preferred embodiment systems and methods comprise detecting a call signal level, determining if the call signal level is below a predetermined silence level threshold, and measuring a duration the call signal level remains below the predetermined silence level threshold.

Abstract: A technique for sharing multimedia sessions between users that share a directory number allows incoming calls to be routed to both phones. Once one party picks up, the other party is alerted to this fact. Both parties may then exchange multimedia information to help in the handling and disposition of the call.

Abstract: A telephone exchanging apparatus 30 comprises a joined terminal table 341, a joined terminal table operating portion 352, and a reception side designating portion 355. The joined terminal table 341 stores terminal identification information that identifies a telephone terminal 40 that will join a conference. The joined terminal table operating portion 352 operates the joined terminal table 341 corresponding to joined terminal table operation commanding information. The reception side designating portion 355 designates the reception side of a speech packet transmitted from the telephone terminal 40. Since the joined terminal table 341 is operated by the joined terminal table operating portion 352, even if the number of people who join a conference increases, they can be easily added or deleted.

Abstract: A telecommunications system and method is disclosed for providing call transfer services within H.323 networks, in which the transferring end-point that invokes call transfer relays the media packets received from the transferred end-point to the transferred-to end-point, and likewise relays media packets received from the transferred-to end-point to the transferred end-point. However, the transferring end-point is taken out of the call after transferring the call so that it can make/receive new calls.

Abstract: A local conferencing exchange subscriber line conferencing system and method using a series of input signals from the phone of a subscriber, such as a dual-tone multifrequency (DTMF) sequence of key inputs starting with a star key input, to initiate the conference. The conference is set up in a bridge and the phone of the subscriber is connected to the bridge. All subsequent callers to the subscriber's phone number at the local exchange carrier are routed and connected to the conference at the bridge. When the conference call ends, the subscriber's phone is returned to normal service.

Abstract: Disclosed herein is a three-way call detection system and method for detecting the addition of a third party to a pre-existing telephonic connection between a first party and a second party. The system comprises a meter for measuring the amplitude of a line voltage of a telephonic connection between a first party and a second party.

Abstract: A method of establishing a third party call can include, responsive to a call initiated by a third party to a directory number of a called party, determining that the called party is engaged in an existing telephone call with one or more parties. The telephone call can be processed in a telephony application, wherein the telephony application can determine whether the third party has authority to join the existing call with the called party. If so, the third party can be connected with the existing call.

Abstract: Methods and systems are provided that can automatically facilitate busy line validation (BLV) and emergency interrupt (EI) services with the aid of various sensors and systems that can automatically determine the status of a given phone-line, and further provide a menu of services to a calling party. A calling party that encounters a busy signal during an attempt to call a called party can validate the status of the called line without the intervention of a human operator. If the called line is currently used for a voice conversation, as opposed to a modem or facsimile call, then the calling party can optionally interrupt the ongoing conversation with the permission of the called party.

Abstract: The present invention provides a method and communication system for sending a data message from a called phone to a calling phone while maintaining an active communication between the called phone and a first phone. The calling phone sends a call request to the called phone while the called phone is involved in an active communication with the first phone. The called phone sends a data message to the calling phone in response to the call request. The data message is sent while maintaining an active call with the first phone, so that the first phone is not placed “on hold”.

Abstract: A system and method for speaker identification in a conference calling unit. The method identifies the speaker, locates the speaker identifier, transfers the speaker identifier to an outgoing path and sends the speaker identifier to endpoints participating in the conference call. If the endpoints have the capability of displaying caller identification, their displays will be updated dynamically with the speaker identifier in place of the caller identifier. The system includes a conference call unit. The unit has a processing unit operable to extract the caller identification of the line having the speaker and a network interface unit operable to send the caller identification to endpoints participating in the call as speaker identification.

Abstract: An Internet Protocol Telephony Network and public switched telephone network that allows one or more call waiting callers to dynamically join in an existing and to establish a multiple-party conference call including the call waiting call. A call waiting call may also be added to an existing conference call.

Abstract: A system and method for encoding the speech of multiple speakers for transmission on a limited medium such as ISDN or ADSL. The system and method encode one or more “dominant” speakers on a first channel, and merges all other speakers into a second channel. A “dominant” speaker may be defined according to the needs of the application, but intuitively it is a speaker in a collaboration session who has the “floor”. The method and system transmit speech through a teleconferencing system having an available bandwidth. Speech from a speaking one, or ones, of the plurality of participants is passed while speech from non-speaking participants is rejected. The system responds to the speech from the one, or ones of the speaking participants passed by the speech detector, and determines which of the detected speaking one, or ones, thereof is a dominant speaker.

Abstract: The present invention enables a party which is placed on hold in a first communication session to enter a second communication session and continue to monitor the status of the first communication. For example, in a broadband communication system telephone call or multimedia call, two parties may be in an active communication session with one another and a first of the two parties to the call may receive another incoming call. The first party then places the second party on hold. At this time the second party may wish to place the communication session with the first party on hold to initiate another communication session with a third party. However, the second party would also like to know when the first party takes the communication session off hold so that they can resume their communication. As a result, the second party places the communication on “monitor” hold.

Abstract: Methods and systems are provided for monitoring calls routed to a telephone network voice mail system from a variety of remote telephone locations. When a subscriber receives a telephone call, a determination is made as to whether the subscriber's line is busy or whether the call goes unanswered. If remote call monitoring services are activated, a three-way communication is set up between the calling party, the voice mail system and the subscriber via a remote call monitoring directory number provided by the subscriber. The subscriber is patched into the call on a listen-only basis, but upon command by the subscriber, the subscriber may answer the call. The subscriber may set up a number of remote call monitoring filters, including an “always monitor” list, a “never monitor” list, and a schedule of days or times during which the subscriber allows or excludes remote call monitoring.

Abstract: In a customer contact center (100), a plurality of customers' communications are serviced simultaneously by one resource (120-128) (agent or port). A conferencing function (136) connects the plurality of customers (110-118) to the one resource and/or to each other. When a resource becomes available, a batch service function (140) determines (206-226), for each skill of the resource, the value to the contact center of having the resource presently serve communications needing that skill, and uses the conferencing function to conference (234-236) a plurality of communications needing the skill having the highest value with the resource. The conference connection may be listen-only, listen-and-talk, or listen-only/then listen-and-talk.

Abstract: A system and method are disclosed for recording a telephone call upon demand by a party to the telephone call. The method includes the steps of receiving a request from a first party to record a telephone call between the first party and a second party, establishing a third-party connection with the first party and the second party, receiving the telephone call, and storing the content of the telephone call in memory for subsequent delivery or retrieval.

Abstract: Embodiments of the present invention provide a network based voice mail system that includes advantages of traditional answering machines. When a call comes in that is not answered, a Remote Answering Device may pick up after a preselected number of rings and play a prerecorded greeting that is stored locally and then prompt the caller to wait for the beep. The Remote Answering Device may use three-way calling service to conference the user's voice mail number into the call. When a conference call is established it plays the beep and then plays the message which is being recording live on the speaker. If a call is received by the network based voice mail system that did net first pass through the Remote Answering Device, the network based voice mail system may synchronize the Remote Answering Device with the network based voice mail system.

Abstract: A telephone accessory device includes a first circuit that detects an off-hook condition of a telephone and a second circuit responsive to the first circuit wherein the second circuit simulates an on-hook condition when the off-hook condition is detected.

Abstract: A multipoint control unit (104) is provided which allows for dynamic codec selection. According to one embodiment, the MCU (104) causes endpoints (102, 106) to renegotiate their codec selections if a most-commonly available codec is not being used, upon entry of new parties to a teleconference. Alternatively, the codec renegotiation may be performed each time a user speaks, to optimize for maximum transmission quality or for minimizing transcoding.

Abstract: The present invention is a method and apparatus for selectively controlling the interconnectivity of two or more active telephone lines and/or one or more audio characteristics of each of the active telephone liens during a multi-line call. A bridge or a call connection control may be implemented to provide selective call connection for each active telephone call. The user may thereby control the interconnectivity of each parry to the user and to the other parties to the conference or multi-line call. An audio controller may also be implemented within a multi-line telephone or at a telephony switch. The call connection control or the volume control may be implemented within the telephone network such as a switch or locally at the telephone unit. The user may provide the desired interconnectivity or volume parameters for the multi party call using a user interface.

Abstract: A telecommunications system containing a plurality of local and transit exchanges in which signaling and traffic data are routed between remote exchanges via a network service having routing code recalculation is described. The system is adapted to provide an ISDN end-to-end supplementary service between two subscribers in which a subscriber access code is generated by modification of a called subscriber routing code, used by the calling subscriber local exchange. The local exchanges that have an applications service element for providing the ISDN supplementary service, include a mechanism to trigger a query, which is sent to a translation device whenever the supplementary service is requested and a subscriber routing code is detected. The subscriber access code returned from the translation device is forwarded to the applications service element, which, will establish an end-to-end dialogue with it's counterpart at the called subscriber local exchange and provide the ISDN supplementary service.

Abstract: A system and method are disclosed for recording a telephone call upon demand by a party to the telephone call. The method includes the steps of receiving a request from a first party to record a telephone call between the first party and a second party, establishing a third-party connection with the first party and the second party, receiving the telephone call, and storing the content of the telephone call in memory for subsequent delivery or retrieval.

Abstract: A method and system for real-time monitoring voice mail during an active call includes generating a duplicate of a calling party voice stream being recorded by voice mail for a called party on a call with a previously connected party. The duplicate calling party voice stream is combined with a previously connected party voice stream to generate a combined voice stream for the called party. The combined party stream is played to the called party during the call.

Abstract: A telephone system includes a storage device, which receives a telephone number and stores the telephone number. A customer premises equipment (CPE) device includes a three-way calling feature such that the stored telephone number is enabled to be called from the CPE over a switched network and a connection is made by the CPE over the switched network between the telephone number and a user at a location remote from the CPE.

Abstract: A three-way communication session is established between a first and a second party, wherein the first party is additionally connected to a third party via a previously established communication. An indication (such as the telephone number associated with the third party, an alphanumeric character, or an icon) is then transmitted to the second party to notify him/her of the presence of the third party in the three-way communication session.

Abstract: Intelligent network (IN), comprising at least one Service Switching Point, SSP (2), at least one Service Control Point, SCP (3), one or more Intelligent Peripherals, IPs (5), and/or one or more Voice Response Systems, VRSs (4), which IN is capable of interacting with a PSTN or ISDN (1). The essence of the concept is an additional interface, in the form of a control member (6), between the IPs and/or VRs on one side and the Service Control Point(s) on the other side. Such makes it possible that the (advanced) IPs and VRSs are capable of exchanging all sorts of information with the SCP, without the IPs or VRSs consequently being occupied for an extended period of time. As a result, the SCP may dispose of (additional) information which in a regular IN cannot be made available to the SCP(s).

Abstract: A method of establishing a third party call can include, responsive to a call initiated by a third party to a directory number of a called party, determining that the called party is engaged in an existing telephone call with one or more parties. The telephone call can be processed in a telephony application, wherein the telephony application can determine whether the third party has authority to join the existing call with the called party. If so, the third party can be connected with the existing call.

Abstract: A method and system which utilize a graphical user interface to identify potential participants in a teleconference, specify a user-controlled dial-up/hang-up order, and monitor the status of participants to the teleconference. The method and system receive conference commands from a World Wide Web (WWW) browser and translate the conference commands into commands that control a telephone bridge.

Abstract: A process, apparatus and system to support call tunneling, a private conversation between a subset of conference call participants, conducted over any network, the private conversation excluding other conference call participants. When call tunneling is activated, participants in the call tunnel route their audio/video/multimedia data selectively. The selectively routed data is sent only to other participants in the call tunnel, thus creating a private conversation within the conference call. The resulting private conversation may be identical in format type (audio, video, or multimedia) as the simultaneously occurring conference call.

Abstract: Interacting call appearances are realized by enabling voice input/output to be simultaneously available on any combination of call appearances at a given time on a given terminal. Consequently, a user may, for example, listen to the voice output of one or more call appearances that are active at the terminal at a given time. Additionally, the user may direct voice input to one or more call appearances that are active at the terminal at a given time. Thus, the user, at any given time, may listen or speak to one or more parties on unrelated calls. Additionally, a user can employ more than one voice input/output device at his/her terminal. Specifically, the ability to simultaneously use more than one call appearance at a terminal at a given time is realized by separating call control from media routing and input/output control. Thus, the appearance of a call at a terminal is separated from the voice input/output that is required for the call.

Abstract: A cellular telephone subscriber who wishes directory assistance services is connected in the conventional manner to an operator who identifies a destination telephone number desired by the subscriber. As in the prior art, the operator then initiates a call connecting the subscriber to the destination telephone number. In a preferred embodiment, the operator locates a desired destination telephone number in a computer database, and can select automatic dialing of the located number. Further, rather than dropping all further involvement with the call, the preferred embodiment of the present invention continually monitors the connection thereby established for a predetermined DTMF signal issued by the customer, such as that obtained by pressing the “*” button. If such a signal is detected, the customer is transferred to a directory assistance operator, who can then provide whatever further assistance is needed (e.g. redialing a busy number, or providing further directory assistance).

Abstract: A point-of-presence (POP) call center system capable of answering, servicing, queuing and routing of calls at local points of presence to reduce communications costs and enhance operational efficiency for toll-free inbound call centers. The POP call center system includes a set of point-of-presence call center gateways distributed at points of presence close to the point of call origination.

Abstract: A stable telephone call barge-in without operator intervention is described wherein a telephone line is provided with two separate directory numbers. One directory number is the usual directory number associated with the line. The second directory number is considered the priority directory number which also translates in circuit switched telephony as the same line ID. Therefore, when a call is in a stable state, a person dialing the regular directory number will receive a busy signal. The person can then disconnect and dial the priority number. When the priority number is received, advantageously, a password is entered into the system to verify that the user of the priority number has permission to barge in. If the password is valid, the existing call is disconnected and the party barging in is connected to the called party. The other call may be placed on hold or alternatively may be disconnected with or without an announcement.

Abstract: A system, method and computer program product for achieving local number portability (LNP) costing and network management support is disclosed. The disclosure includes a LNP graphical user interface (GUI) implemented as, for example, a WEB or Internet based tool to audit related charges, and to enable users to determine ported number status on a real-time basis.

Abstract: A caller-controlled barge-in telephone service, managed by telephone systems, intercepts telephone calls directed to instantly busy telephone lines predesignated as objects of this service, and: 1) verifies that the called line is a respectively predesignated object of this service and that the caller is entitled to invoke the service relative to that line; and 2) forms a bridging connection between the caller and parties to a telephone call instantly busying the called line. Verification that a caller is entitled to the service can be implemented by playing a voice announcement requesting the caller to speak or key in an authorization code. If the caller fails to respond or returns an invalid code, the caller is either disconnected or transferred to a voice mail service. Disconnection of the caller may be preceded by a voice announcement (e.g. a spoken “goodbye”).

Abstract: A telephone system which includes voice mail screening is disclosed. The telephone system includes a switching system for receiving a call from a calling party. The system also includes a voice mailbox which is coupled to the switching system if the called party does not answer the call. The system further includes a telephone for receiving the call from the calling party. The telephone is capable of screening the calling party when the calling party is coupled to the voice mailbox.

Abstract: An electronic mail pop-up control system in a dealing system for lightening a burden on a dealer handling a large amount of information at the same time by reducing the number of operations to be done on a dealing talk terminal, said control system lightening a burden on a dealer by displaying an electronic mail which is one of means for exchanging information between talking terminals 1 on a display 1-26 of a called side talking terminal 1 without any operation performed by a dealer at the called side.

Abstract: A network based voice mail system without losing any of the advantages of traditional answering machines includes a small device, which looks like an answering machine and costs about the same, and three-way calling service. The small device (called a Remote Answering Device) looks just like an ordinary answering machine. It has a speaker, a message number indicator and the basic play back control buttons like “play”, “delete”, “stop”, “rewind”, “skip”, etc. When a call comes in that is not answered by a human, the Remote Answering Device will pick up after a preselected number of rings and play a prerecorded greeting that is stored locally and then prompt the caller to wait for the beep. The Remote Answering Device then uses three-way calling service to conference the user's voice mail number into the call, by sending a hook flash signal and speed dialing the user's voice mail number.

Abstract: A method of interrupting a call between two parties in a switched telecommunications. A monitoring switch monitors the call for a party initiated interrupt signal. Whenever the monitoring switch receives an interrupt signal initiated by its associated party, the monitoring switch initiates a call between the party initiating the interrupt signal and a serving platform, and parks the other party at a port of the monitoring switch. The monitoring switch then monitors the status of the call between the party initiating the interrupt signal and the serving platform, and the status of the parked call so that the parties may be reconnected at the conclusion of the call between the requesting party and the serving platform or informed if one of the parties disconnects before they are reconnected.

Abstract: A point-of-presence (POP) call center system capable of answering, servicing, queuing and routing of calls at local points of presence to reduce communications costs and enhance operational efficiency for toll-free inbound call centers. The POP call center system includes a set of point-of-presence call center gateways distributed at points of presence close to the point of call origination that are connected by a virtual private network to premises call center gateways at business locations where the call centers reside.

Abstract: The invention relates to a process for the operation of a network with terminals (EG1, . . . , EGm) that can be connected to each other via transmission nodes (K1, . . . , Km) of the network, whereby, with the existence of a connection between two terminals, other terminals, accordingly authorized, can link into this connection. If one terminal (EG1) of the terminals (EG1 to EG6), sharing the connection, wants to maintain a connection with only one terminal (EG2) and wants to separate the other terminals (EG3 to EG5) from the connection, this terminal (EG1) sends out a corresponding request to all transmission nodes (K1, . . . , K5) that are part of the interlinking network, which is made up of the terminals (EG1 to EG6). After receipt of the request, each transmission node of the interlinking network sends back a corresponding confirmation to the transmission node from which it received the request.