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Небесная энциклопедия

Космические корабли и станции, автоматические КА и методы их проектирования, бортовые комплексы управления, системы и средства жизнеобеспечения, особенности технологии производства ракетно-космических систем

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Мониторинг СМИ

Мониторинг СМИ и социальных сетей. Сканирование интернета, новостных сайтов, специализированных контентных площадок на базе мессенджеров. Гибкие настройки фильтров и первоначальных источников.

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Форма поиска

Поддерживает ввод нескольких поисковых фраз (по одной на строку). При поиске обеспечивает поддержку морфологии русского и английского языка
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Применить Всего найдено 5580. Отображено 100.
12-01-2012 дата публикации

System and method for efficient call management for directory assistance services

Номер: US20120011146A1
Принадлежит: Individual

A communication assistance system includes a first database having a plurality of listings, each listings has at least one contact name and a corresponding contact number. A means is provided for receiving communications from a user among a plurality of users desiring to access the listings. An operator terminal displays a search screen among a plurality of search screens, each of which maintain at least a partially different arrangement of content and search windows for receiving search terms to search for listings contained in the first database, where the displayed search screen is determined based on criteria contained in a search request from the user. The operator terminal is further configured to receive the communication from the user and retrieve a listing from the first database using the displayed search screen.

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09-02-2012 дата публикации

Associated device discovery in ims networks

Номер: US20120036248A1
Принадлежит: Aylus Networks Inc

A method of associating multiple user endpoints (UEs) with a single IMS session in an IMS network having a serving node for controlling at least one IMS session for a user and at least a first access network for providing access to UEs. The method involves associating a first UE with the user and with an IMS session; discovering a second UE in a proximity of the first UE; discovering information about the second UE; communicating the information about the second UE to the serving node; the serving node utilizing computer-implemented policy logic to determine whether to associate the second UE with the user and the IMS session; and if the policy logic determines that the second UE is to be associated, the serving node associating the second UE with the IMS session while retaining the association with the first UE.

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23-02-2012 дата публикации

Method and Apparatus for Telephonically Accessing and Navigating the Internet

Номер: US20120047216A1
Принадлежит: Ben Franklin Patent Holding LLC

A method for accessing and browsing the interne 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.

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01-03-2012 дата публикации

Determination of bypass zones from network configuration settings

Номер: US20120051261A1
Принадлежит: Microsoft Corp

Bypass zones for a network are identified by generating bypass identifiers that identify the bypass zones from network configuration settings. During call setup the bypass identifier assigned to an endpoint for the call is identified and the bypass identifier assigned to a gateway for the call is identified. A determination is then made as to whether the bypass identifier assigned to the gateway is the same as the bypass identifier assigned to the endpoint. If the bypass identifiers are the same, then a mediation server may be bypassed for the call. If the bypass identifiers are not the same, then the mediation server remains in the media path for the call.

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31-05-2012 дата публикации

Apparatus, systems and methods for managing incoming and outgoing communication

Номер: US20120137009A1
Принадлежит: Individual

A system and method for managing incoming and outgoing communications may include a definition of if, when, and who may communicate with a recipient. The identification of the communication may be concealed in that no actual addresses, phone numbers, or other addressing identifications are required to be exchanged by the communication initiator and recipient. In an example, if the database contains call management settings for a call recipient, the application logic may evaluate the rules to determine if a particular caller is authorized to connect with the call recipient at the current time and date. The application logic may connect the call utilizing the public telephone switch. The caller and call recipient phone numbers may be stored in a database.

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05-07-2012 дата публикации

IP Multimedia Subsystem (IMS)-Based Pre-negotiation of Video Codec For Video Single Radio Video Call Continuity

Номер: US20120169825A1
Автор: Milan Patel
Принадлежит: InterDigital Patent Holdings Inc

Systems, methods, and instrumentalities are disclosed to provide pre-negotiation of a video codec. An IP multimedia subsystem (IMS) entity, such as a service centralization and continuity application server (SCC AS), may send a first session initiation protocol (SIP) message to a circuit switched domain entity via an IMS core network. The first SIP message may correspond to a video call in an on-going IP multimedia subsystem (IMS) session. The first SIP message may include one or more video codecs supported by the SCC AS and the UE associated with the video call. The SCC AS may receive a second SIP message from the circuit switched domain entity. The second SIP message may include the video codec, which may be one of the video codecs included in the first SIP message. The SCC AS may send the video codec to a user equipment (UE) associated with the video call.

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05-07-2012 дата публикации

Method for re-configuring a communications device

Номер: US20120170731A1
Принадлежит: British Telecommunications plc

A communications device ( 12 ) maintains profile data including a device identity and a calling line identity (CLI) associated with a PSTN connection ( 28 ) both on the device itself and also on an authentication platform ( 32 ), in order that the device can access a service ( 52 ) which requires knowledge of the CLI by way of a packet-switched connection ( 30 ). In the event that the CLI data in the communications device is corrupted or lost, the communications device transmits a request for re-authentication to a reactivation server ( 46 ) together with the device identity. On receipt of the request, the reactivation server retrieves the stored profile ( 50 ) from the authentication platform ( 32 ), and returns it to the communications device ( 12 ) to allow the profile to be restored. This process can be done without the user needing to send a new request for service over the PSTN link ( 28 ).

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26-07-2012 дата публикации

Enhanced E911 Network Access for a Call Center Using Session Initiation Protocol (SIP) Messaging

Номер: US20120189107A1
Принадлежит: TeleCommunication Systems Inc

A switched emergency call (e.g., a 911 call, an alarm company call) forwarded by a telematics call center is converted into a session initiation protocol (SIP) packetized phone call at the call center, and routed over an IP network, for presentation to an emergency services gateway, which connects to a selective router via dedicated circuits, gaining full access to the Enhanced 911 network. This provides a PSAP receiving a call from a telematics call center or other call center with all features available in an Enhanced 911 network, e.g., callback number of the 911 caller, and location of the 911 caller. Location of the caller is provided using a VoIP positioning center (VPC), queried from the call center. In this way, the switched emergency call is converted into a SIP packetized phone call and routed without further passage through the public switched telephone network (PSTN).

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20-09-2012 дата публикации

Tandem Access Controller Within The Public Switched Telephone Network

Номер: US20120236762A1
Принадлежит: Individual

A system that includes a tandem access controller (TAC) coupled to the PSTN, where the TAC allows a subscriber to set-up and change the configuration of the phone line or other communications device, including selective call forwarding, using the web. The TAC is coupled internally to the PSTN in a local service area and is outside the central office of the subscriber. A calling party makes a first call to the subscriber using the subscriber's public telephone number. The TAC receives the first call prior to the call reaching the subscriber's terminating central office, which in some cases avoids a toll. The TAC then carries out the subscriber's instructions for the first call, such as making one or more second calls using telephone numbers different from the subscriber's public telephone number. When the second call is answered, the answering phone is connected by the TAC to the caller.

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20-09-2012 дата публикации

Pbx mobility system with multiple call legs

Номер: US20120238276A1
Принадлежит: Research in Motion Ltd

A private branch exchange (PBX) mobility system and associated method for re-using of call legs in an enterprise system. The enterprise system includes a session management platform (SMP) having a private branch exchange (PBX) mobility enabler, the PBX mobility enabler having third-party control communications to one or more PBXs over a session control interface, such as a Session Initiation Protocol (SIP) interface. The PBXs are in communication with one or more media servers. The PBX mobility enabler is configured to provide control communications to the PBXs over the session control interfaces to re-use a wireless call leg and connect to other call legs.

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27-09-2012 дата публикации

Protocol interworking device for network telephone system and method for using the same

Номер: US20120246328A1
Автор: Jun Song, Song Guo

A protocol interworking device communicates with at least one first network telephone terminal supporting a first protocol and at least one second network telephone terminal supporting a second protocol. The protocol interworking device includes a protocol converter supporting both the first and second protocols, and a transferring telephone enabled to communicate with both the first and second network telephone terminals by the protocol converter. When the first network telephone terminal sends a call request for calling the second network telephone terminal to the protocol interworking device, the transferring telephone hangs up with the first network telephone terminal and sends another call request to call the second network telephone terminal. When successfully connecting to the second network telephone terminal, the transferring telephone reconnects with the first network telephone terminal, and the first and network telephone terminals communicate with each other through the protocol interworking device.

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08-11-2012 дата публикации

Method and apparatus in which call signaling messages bypass in-transparent switching nodes or networks

Номер: US20120281691A1
Принадлежит: Individual

A method is disclosed for initiating a call in a communication network between a first signaling entity and a second signaling entity each connected to a respective node and the nodes connected to a network wherein communications between the signaling entities and the nodes are conducted using a first protocol and communications are carried over the network using a second protocol. Call setup information is also exchanged between the first node and the second node over a separate connection.

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22-11-2012 дата публикации

Voice-over-ip hybrid digital loop carrier

Номер: US20120294304A1
Принадлежит: AT&T Intellectual Property II LP

Certain exemplary embodiments can comprise a method of use comprising: for a call between a local IP network and a remote non-IP network, converting between IP packets and PCM robbed bit signaling via a VoIP channelized router; providing the PCM robbed bit signaling to a TDM switch via the VoIP channelized router; and/or converting between IP packets and GR303 call reference values via the VoIP channelized router.

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29-11-2012 дата публикации

Circuit switching user agent system, communicating device, and service providing method used therefor

Номер: US20120302238A1
Принадлежит: NEC Corp

A circuit switching user agent system includes a service processing device arranged in a home network of an IMS (IP Multimedia Subsystem)/MMD (Multi-Media Domain). The circuit switching user agent system also includes a communication device arranged in a visited network where a mobile terminal has visited, the communication device having a function of interconversion between a UNI (User-Network Interface) signal in a circuit switching network to which the mobile terminal is connected and a signal used in the IMS/MMD.

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27-12-2012 дата публикации

Method and System For Calling Traditional Circuit Switched Domain Network User By Packet Core Network

Номер: US20120327930A1
Автор: Minggang Gao
Принадлежит: ZTE Corp

A method and a system for calling a traditional circuit switched domain network user by a packet core network. The method includes: after receiving a request message from a call session control unit, a media gateway control unit sends the call session control unit a first call progress message carrying a preset prompt tone indicating please wait; the media gateway control unit sends an initial address message to the traditional circuit switched domain network and start a preset timer; if the timer expires and the traditional circuit switched domain network does not respond, the media gateway control unit sends the call session control unit a second call progress message carrying a preset prompt tone indicating that the other party is temporarily unaccessible, and the media gateway control unit proceeds according to a preset policy. Resource waste due to waiting is avoided and better user friendliness of a service is provided.

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10-01-2013 дата публикации

Dual Tone Multi-Frequency Signal Transmission Method and Device

Номер: US20130012169A1
Автор: Qiansheng Zhou
Принадлежит: ZTE Corp

A method and apparatus for transmitting a dual tone multi-frequency (DTMF) signal are disclosed by the present invention. The scheme of the present invention includes: when receiving an input instruction for inputting a DTMF character transmitted by a network side, obtaining the DTMF character to be transmitted and caching the DTMF character in a character queue ( 102 ); when receiving a character modification command, modifying the cached character queue according to character modification indication information carried in the character modification command ( 104 ); and when determining that all DTMF characters to be transmitted are already cached, transmitting the DTMF characters in the cached character queue to the network side according to the order of the DTMF characters in the character queue ( 106 ). The technical scheme of the present invention can improve the operability of the service and the accuracy of character input.

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25-04-2013 дата публикации

Call Control Entity for a Communication Network

Номер: US20130100883A1
Принадлежит: Telefonaktiebolaget LM Ericsson AB

A communication network is described, comprising: an access network, a switching control network comprising call control entities arranged for association with subscribers, where said access network is arranged for routing signalling of a given subscriber to a call control entity associated with said given subscriber, and an Internet Protocol based network for providing a set of services centred in said Internet Protocol based network to one or more of said subscribers, wherein said switching control network comprises a group of first call control entities and a group of second call control entities, said first call control entities are arranged for providing call control services to subscribers having circuit switched subscriptions, and said second call control entities are arranged for providing a gateway to connect to said Internet Protocol based network for subscribers having subscriptions to said set of services centred in said Internet Protocol based network, and each first call control entity has a subscriber association redirector for redirecting an association of a subscriber associated with said each first call control entity and having a subscription to said services centred in said Internet Protocol based network to said group of second call control entities.

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13-06-2013 дата публикации

Enhanced E911 Network Access for a Call Center Using Session Initiation Protocol (SIP) Messaging

Номер: US20130149988A1
Принадлежит: TeleCommunication Systems Inc

A switched emergency call (e.g., a 911 call, an alarm company call) forwarded by a telematics call center is converted into a session initiation protocol (SIP) packetized phone call at the call center, and routed over an IP network, for presentation to an emergency services gateway, which connects to a selective router via dedicated circuits, gaining full access to the Enhanced 911 network. This provides a PSAP receiving a call from a telematics call center or other call center with all features available in an Enhanced 911 network, e.g., callback number of the 911 caller, and location of the 911 caller. Location of the caller is provided using a VoIP positioning center (VPC), queried from the call center. In this way, the switched emergency call is converted into a SIP packetized phone call and routed without further passage through the public switched telephone network (PSTN).

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12-09-2013 дата публикации

Method and apparatus for network maintenance and supervision of a controlled on-board audio portion

Номер: US20130235997A1
Автор: Zohar Halachmi
Принадлежит: SellARing Ltd

A method for network maintenance and supervision of an on-board controlled audio agent, the method constituted of: providing a plurality of audio portions, each of the provided audio portions being selected responsive to user descriptive data associated with a call initiator; providing a reimbursement rule associated with each of the audio portions; selecting, for each call instance, a particular one of the plurality of audio portions responsive to the determined reimbursement rules; and outputting the particular one of the plurality of audio portions at each call instance initiation such that at least a portion of the particular audio portion is heard by the call initiator prior to call connection.

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12-09-2013 дата публикации

Method and System for Stereo Echo Cancellation for VOIP Communication Systems

Номер: US20130236004A1
Принадлежит: Broadcom Corp

An exemplary embodiment of the present invention is directed toward a method and system for cancelling line echo in the presence of a known secondary audio signal. Filter adaptation is enabled in the presence of a known secondary audio source such as the sound of a computer game, a music signal or other secondary audio sources that would otherwise prevent echo cancellation due to an apparent double talk condition. It is emphasized that this abstract is provided to comply with the rules requiring an abstract which will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or the meaning of the claims.

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19-09-2013 дата публикации

Method and apparatus for wideband and super-wideband telephony

Номер: US20130242858A1
Автор: Gilbert A. Amine
Принадлежит: Microsemi Semiconductor US Inc

A gateway includes at least one network interface, at least one analog telephony interface, and a processing unit operable to receive a bandwidth signal over the at least one analog telephony interface from a telephony device and configure an audio bandwidth of a telephony connection for the telephony device over the at least one network interface based on the bandwidth signal.

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26-09-2013 дата публикации

Method of passing signal events through a voice over ip audio mixer device

Номер: US20130250817A1
Принадлежит: RadiSys Canada ULC

In Voice-over-IP or like systems, audio mixers are used for implanting conferencing applications for IP networks. In such systems, a problem exists as how to pass through user signal events, such a DTMF digits, in-band as well as out-of-band telephony events (RFC 2833/4733) because they are not handled well in such systems. In accordance with certain embodiments, a signalling event in incoming audio packets or user signalling event received out-of-band is detected at one of the inputs. A meta-data representation of the signal event is attached to the audio and passed through the mixer for regeneration, as in-band or out-of-band telephony events, on the other side of the mixer.

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03-10-2013 дата публикации

METHOD, SYSTEM AND APPARATUS FOR IMPLEMENTING MULTIMEDIA RING BACK TONE SERVICE

Номер: US20130259219A1
Принадлежит:

A method, system and apparatus for implementing a Multimedia Ring Back Tone (MRBT) service are provided. The method includes: receiving a call request originated by a calling terminal; parsing a tone playing policy specifying whether to play a caller tone or a callee tone or whether to filter a tone; performing caller tone media negotiation or callee tone media negotiation, or performing no tone negotiation according to the parsing result; and playing the caller tone or the callee tone to the caller, or playing no tone. With the technical solution of the present invention, the caller tone service can be implemented in the IMS domain. Whether a caller tone or a callee tone is played to the caller is determined according to a preset policy. Furthermore, the tone filtering service is also implemented. This gives a user freedom in experiencing the MRBT service. 1. A method for implementing a multimedia ring back tone filtering service , comprising:receiving a call request originated by a calling terminal;determining whether to filter a tone according to the call request; andfiltering the tone according to the decision.2. The method of claim 1 , where the filtering the tone comprises filtering a caller tone or a callee tone.3. The method of claim 1 , wherein the call request carries a tone filtering flag which is added in the call request according to a tone filtering policy by the calling terminal;wherein the determining of whether to filter a tone according to the call request comprising determining whether to filter a tone according to the tone filtering flag.4. The method of claim 3 , wherein the determining of whether to filter a tone according to the tone filtering flag comprising:determining by a tone application server, whether to filter a tone according to the tone filtering flag; ordetermining by a call session control function server, whether to send the call request to a tone application server.5. The method of claim 1 , wherein the determining of whether to ...

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03-10-2013 дата публикации

METHOD, SYSTEM AND APPARATUS FOR IMPLEMENTING MULTIMEDIA RING BACK TONE SERVICE

Номер: US20130259220A1
Принадлежит: Huawei Technologies Co., Ltd.

A method, system and apparatus for implementing a Multimedia Ring Back Tone (MRBT) service are provided. The method includes: receiving a call request originated by a calling terminal; parsing a tone playing policy specifying whether to play a caller tone or a callee tone or whether to filter a tone; performing caller tone media negotiation or callee tone media negotiation, or performing no tone negotiation according to the parsing result; and playing the caller tone or the callee tone to the caller, or playing no tone. With the technical solution of the present invention, the caller tone service can be implemented in the IMS domain. Whether a caller tone or a callee tone is played to the caller is determined according to a preset policy. Furthermore, the tone filtering service is also implemented. This gives a user freedom in experiencing the MRBT service. 1. A method for implementing multimedia ring back tone service , comprising:receiving by a caller tone application server, a call request originated by a calling terminal;parsing by the caller tone application server, a tone playing policy according to the call request, wherein the tone playing policy specifies whether to play a caller tone or a callee tone;determining by the caller tone application server, whether to perform caller tone media negotiation according to the parsing result;substituting by the caller application server, an early media SDP of a calling tone for the early media SDP of a called tone in a received response when it is determined that caller tone media negotiation is to be performed;sending by the caller application server, the received response which comprises the early media SDP of the calling tone, to the calling terminal;instructing by the caller tone application server, a Media Resource Server (MRS) to play the caller tone to the calling terminal according to the media negotiation result upon reception of an alerting signal from the callee.2. The method of claim 1 , wherein after ...

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10-10-2013 дата публикации

PROACTIVE TELEPHONE NUMBER MAPPING CONFIGURATION MANAGEMENT

Номер: US20130266132A1
Автор: Ku Bernard
Принадлежит:

Efficient telephone number mapping (ENUM) based call routing during area code splits is described. An ENUM domain management component can associate an ENUM domain name with multiple records when a numbering plan area code split links an old area code of a telephone number corresponding to the ENUM domain name with a new area code. An ENUM provisioning component can perform permissive dialing of the old and new area code during ENUM based call routing by utilizing the multiple records. 1. A system , comprising:a memory to store instructions; and in response to determining that second data representing a numbering plan area code split relates an old area code of area codes that are assigned to a telephone number corresponding to a telephone number mapping domain name to a new area code of the area codes, associating first data representing the telephone number mapping domain name with records representing the area codes;', 'facilitating, based on the records, use of the old area code and the new area code associated with call routing utilizing the first data representing the telephone number mapping domain name;', 'permitting, for a predetermined period of time, dialing of the old area code and the new area code to facilitate connection to a telephone number mapping communication termination point device;', 'relating the first data representing the telephone number mapping domain name to a record of the records based on the predetermined period of time;', 'preventing a modification of the first data representing the telephone number mapping domain name during the predetermined period of time; and', 'in response to determining that a permissive dialing period representing the predetermined period of time has ended, disassociating the first data representing the telephone number mapping domain name from one of the records that is associated with the old area code., 'a processor, coupled to the memory, that facilitates execution of the instructions to perform operations ...

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24-10-2013 дата публикации

COMMUNICATION DEVICE ANSWERING ENHANCEMENT SYSTEM AND METHOD

Номер: US20130279684A1
Принадлежит:

An answering enhancement system originating on a called-party's communication device and working in conjunction with other resident software and hardware that becomes operational once the called-party's communication device is connected to an incoming call through a communication link. The answering enhancement system is not a part of or a function of the communications network. Once activated by a communications transmission, the answering enhancement system may instruct the called-party's communication device to play media files to calling-party or to the called-party by acting on associations pre-selected by the called-party. The media files may be played or displayed to the calling-party in replace of a conventional audible call progress signal or ringback tone and to the called-party as a ringtone and/or visual alert. The system allows the called-party to access the call and converse with calling-party as it normally would. After the media files have played for a pre-determined period of time the call may be transferred to a voicemail recording system residing in the called-party's communication device or network voicemail recording system for later retrieval. The media files may then be played or displayed to the calling-party as a voicemail announcement. 1. A system for customizing a communications device connected to a communications network , the system comprising:a first communication device associated with a called party;a media file stored in a storage device accessible by said first communication device;an identifier stored in the storage device accessible by said first communication device and associated with said media file, wherein said identifier identifies a second communication device associated with a calling party;wherein said first communication device is configured to complete a communication link with an incoming call received from the second communication device, identify the second communication device, and identify said media file by cross ...

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31-10-2013 дата публикации

METHOD, SYSTEM AND APPARATUS FOR IMPLEMENTING MULTIMEDIA RING BACK TONE SERVICE

Номер: US20130287199A1
Принадлежит:

A method, system and apparatus for implementing a Multimedia Ring Back Tone (MRBT) service are provided. The method includes: receiving a call request originated by a calling terminal; parsing a tone playing policy specifying whether to play a caller tone or a callee tone or whether to filter a tone; performing caller tone media negotiation or callee tone media negotiation, or performing no tone negotiation according to the parsing result; and playing the caller tone or the callee tone to the caller, or playing no tone. With the technical solution of the present invention, the caller tone service can be implemented in the IMS domain. Whether a caller tone or a callee tone is played to the caller is determined according to a preset policy. Furthermore, the tone filtering service is also implemented. This gives a user freedom in experiencing the MRBT service. 1. A method for implementing multimedia ring back tone service , comprising:receiving a call request originated by a calling terminal;parsing a tone playing policy according to the call request, wherein the tone playing policy specifies whether to play a caller tone or a callee tone;performing caller tone media negotiation or callee tone media negotiation according to the parsing result; andplaying the caller tone or the callee tone to the calling terminal according to the media negotiation result of the caller tone or the callee tone, respectively, upon reception of an alerting signal from the callee.2. The method of claim 1 , wherein before performing the caller tone media negotiation or the callee tone media negotiation comprising:forwarding by a caller tone application server, the call request after receiving the call request, wherein a caller tone flag or a callee tone flag is added in the forwarded call request according to the tone playing policy.3. The method of claim 2 , wherein the performing of the caller tone media negotiation or the callee tone media negotiation comprising:determining by a callee ...

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07-11-2013 дата публикации

System and Method for Provision of a Second Line Service to a Telecommunications Device

Номер: US20130295899A1
Принадлежит: Individual

A method for routing calls between a third party telecommunications device (“TD”) and a subscriber TD associated with a primary service and a second line service (“SLS”) involves associating the SLS number of the subscriber, the primary number of the subscriber and the primary number of a third party via a common relationship number. Calls directed from a third party to the SLS number of a subscriber are routed to an SLS platform and redirected to the subscriber TD. Calls directed from the subscriber TD to the third party use the relationship number to route the call to the SLS platform. The combination of the SLS number and the relationship number identifies the third party calling number for call completion. Calls can be directed to and from an SLS number of a subscriber TD without having to coordinate the provisioning of a call through the subscriber's primary service provider.

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19-12-2013 дата публикации

On-Hold Message System

Номер: US20130336468A1
Автор: Hazenfield Joey C.
Принадлежит: INFO-HOLD, INC.

A programmable in-the-skin or intelligently connected message on hold delivery system is disclosed which comprises a message storage system and a processor for generating prompts which are transmitted to a control device. The prompts are received at the control device and allow an operator to select from a number of options in order to select, among other things, certain ones of the messages stored on said message storage system for playback, as well as the sequence in which said messages are to be played back. The message playback devices are each provided with one or more libraries of messages, and comprise at least one or more audio and/or visual advertising messages. Said system enables the user to start a pre-recorded audio or visual message at the beginning each time a caller is placed on hold in the telephone system. 1. An on-hold messaging system , comprising:a business telephone system comprising a plurality of lines or extensions on which a calling party from outside the system may be placed on hold;a storage device for storing a plurality of messages;a message playback device for playing said messages; anda control device for permitting a user of said telephone system to select one or more messages from among said plurality of stored messages and one or more lines or extensions from among said plurality of lines or extensions on which said one or more selected messages are to be played by the message playback device when a caller or called party is placed on hold by said telephone system;wherein the number of said one or more lines or extensions selected using said control device can be less than the total number of said plurality of lines or extensions.2. An on-hold messaging system as claimed in claim 1 , wherein said control device permits said user to select the order in which said one or more selected messages are to be played by said message playback device on said one or more selected lines or extensions.3. The on-hold messaging system of claim 1 , ...

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27-02-2014 дата публикации

Method and system for providing to a second party, computer-network related information about a first party

Номер: US20140056420A1
Автор: Arnold M. Lund
Принадлежит: RAKUTEN INC

A method and system for providing computer-network related information about a second party. First, the second party receives a telephone number of a first party. The second party's customer premises equipment (CPE) or elements in a telephone network then use the telephone number to index a database, which contains combinations of telephone numbers and computer-network addresses. Once the first party's computer-network address is retrieved, first-party-customized information present at the computer-network location specified by the computer-network address can be sent to and displayed on the second party's CPE. Also, the first party's telephone number can be sent to an application in the computer network, causing the first-party-customized information to be automatically displayed on the second party's CPE.

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27-02-2014 дата публикации

Automatic Contact Population

Номер: US20140057608A1
Принадлежит: Individual

Systems and methods for populating a contacts directory are disclosed. A method of populating a contacts directory associated with a telephonic device includes placing a call from a first telephonic device to a second telephonic device. Called party contact information associated with the second telephonic device can be received and stored automatically in the contacts directory as a contact.

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05-01-2017 дата публикации

APPARATUS AND METHOD FOR TELEPHONE CALL PROCESSING

Номер: US20170006163A1
Принадлежит:

A telephone call processing apparatus for a telephone system includes an exchange for receiving incoming telephone calls and switching said calls to selected respective agent workstations. The apparatus includes a routing function for connecting an incoming call from a caller to an agent workstation, and initiating a call from said apparatus to an external application platform, said call including data to enable a transaction call to be set up between said caller and said external application platform. A holding function maintains a call leg connection with said agent workstation whilst said call from said apparatus to said external application platform is initiated. A connection function connects a call from said external application platform to said agent workstation and establishing or maintaining a connection between said caller and said agent workstation for use whilst said transaction call is connected. 1. A telephone call processing apparatus for a telephone system comprising an exchange for receiving incoming telephone calls and switching said calls to selected respective agent workstations , the apparatus comprising:a routing function for connecting an incoming call from a caller to an agent workstation, and initiating a call from said apparatus to an external application platform, said call including data to enable a transaction call to be set up between said caller and said external application platform;a holding function for maintaining a call leg connection with said agent workstation whilst said call from said apparatus to said external application platform is initiated; anda connection function for connecting a call from said external application platform to said agent workstation and establishing or maintaining a connection between said caller and said agent workstation for use whilst said transaction call is connected.2. The apparatus according to claim 1 , wherein said routing function is provided by an on demand application module located remotely ...

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04-01-2018 дата публикации

MOBILE COMMUNICATION METHOD AND MOBILE COMMUNICATION SYSTEM

Номер: US20180007609A1
Принадлежит: NEC Corporation

A mobile communication method according to the present invention includes the steps of: sending “INVITE” from a UE# to a P-CSCF/VATF in a visited network of the UE# sending “INVITE” from the P-CSCF/VATF to an IMS; and allocating, by the P-CSCF/VATF, an MGW# to the path for voice communications. 1. A mobile communication method in a mobile communication system for SRVCC (Single Radio Voice Call Continuity) including a mobile station , a first radio access network that does not support circuit-switched communications , a second radio access network that supports the circuit-switched communications , a gateway within a serving network of the mobile station for switching a path for voice communications between the mobile station and a communication partner of the mobile station , from a path using the first radio access network , to a path using the second radio access network , and a control node within the serving network of the mobile station , the mobile communication method comprising:transmitting, by the mobile station, a registration request to the control node;transmitting, by the control node, a registration response to the mobile station;transmitting, by the mobile station, an INVITE signal to the control node; anddetermining, by the control node, whether to anchor the path for the voice communications based on SRVCC capability of the mobile station.2. A mobile communication system for SRVCC (Single Radio Voice Call Continuity) , the mobile communication system comprising:a mobile station;a first radio access network that does not support circuit-switched communications;a second radio access network that supports the circuit-switched communications;a gateway within a serving network of the mobile station for switching a path for voice communications between the mobile station and a communication partner of the mobile station, from a path using the first radio access network, to a path using the second radio access network; anda control node within the serving ...

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12-01-2017 дата публикации

EMERGENCY CALL SERVICE FOR GROUPS OF DEVICES WITH A SHARED NUMBER

Номер: US20170013113A1
Принадлежит:

Distinct user devices may sometimes share a telephone number. Techniques described herein may provide for the assigning of a temporary telephone number for a user device, which shares a telephone number with other user devices, in certain situations, such as when the user device places a call to an emergency call center. The temporary telephone number may allow the user device to be reached and/or located by the emergency call center. 1. A system , comprising:a memory device storing a set of processor-executable instructions; and receive a call notification indicating that a user device has placed a call;', 'determine that the user device is associated with at least one other user device, the user device and the at least one other user device being associated with a shared first telephone number;', 'assign, based on the determination, a second telephone number to the user device, the second telephone number being different from the first telephone number; and', 'route the call notification to a callee of the call, the call notification including the second telephone number assigned to the user device., 'a processor configured to execute the processor-executable instructions, wherein executing the processor-executable instructions causes the processor to2. The system of claim 1 , wherein the call notification is a first call notification and wherein the call is a first call claim 1 , wherein executing the processor-executable instructions further causes the processor to:receive a second call notification regarding a second call, the second call notification indicating that a callee of the second call is the second telephone number;determine a device identifier, of the user device associated with the second telephone number; androute the second call notification to the user device, the routing being performed based on the device identifier of the user device.3. The system of claim 2 , wherein the device identifier includes at least one of:an Internet Protocol address, ...

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11-01-2018 дата публикации

TELEPHONY WEB EVENT SYSTEM AND METHOD

Номер: US20180013895A1
Принадлежит:

An embodiment of the system for publishing events of a telephony application to a client includes a call router that generates events from the telephony application and an event router that manages the publication of events generated by the call router and that manages the subscription to events by clients. The system can be used with a telephony application that interfaces with a telephony device and an application server 1. A method comprising:a multi-tenant telephony system receiving a first application instruction of an application server external to the telephony system;a call router of the telephony system generating an event responsive to the first application instruction;the call router publishing the generated event to an event router of the telephony system;the event router sending the published event from the event router to at least one subscriber to the published event.2. The method of claim 1 , wherein the telephony system receives the first application instruction from the application server via an Application Programming Interface (API).3. The method of claim 1 , wherein the event router sending the published event comprises: sending the published event to the application server.4. The method of claim 1 , wherein the event router sends the published event to the application server via an HTTP connection.5. The method of claim 1 , wherein the first application instruction is an instruction to dial a phone number.6. The method of claim 1 , wherein the first application instruction controls interaction between a telephony device and the call router.7. The method of claim 1 , wherein the first application instruction is an instruction of a customer service application.8. The method of claim 1 , wherein the first application instruction is an instruction of a conference call application.9. The method of claim 1 , wherein the call router generates the published event based on the call router dialing a number.10. The method of claim 1 , wherein the call ...

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14-01-2016 дата публикации

Telecommunication call management and monitoring system with voiceprint verification

Номер: US20160014270A1
Автор: Stephen Lee Hodge
Принадлежит: Global Tel Link Corp

A secure telephone call management system is provided for authenticating users of a telephone system in an institutional facility. Authentication of the users of the telephone call management system is accomplished by using a personal identification number, biometric means, and/or radio frequency means. The secure telephone call management system includes accounting software capable of limiting access to the system based on funds in a user's account, and includes management software capable of implementing widespread or local changes to the system. The system monitors a conversation in the telephone call to detect a presence of a first characteristic in audio of the conversation, and terminates the telephone call if the first characteristic does not match a second characteristic of biometric information of a user or a called party.

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14-01-2016 дата публикации

Secure data entry system

Номер: US20160014278A1
Автор: Steven Defoort
Принадлежит: British Telecommunications plc

Data signals (e.g DTMF tones) transmitted on a telephone call between a customer terminal ( 1 ) and a call centre platform ( 4 ) are diverted at the call centre platform 4 to a secure payment system ( 3 ) such that the call centre operative cannot intercept them. In order to verify that this has been done, the security system provides access to the user over a connection ( 23,230 ) independant of the connection ( 34,41 ) to the call centre that allows the user to independently verify that the secure connection 34 has been made. This may take the form of returning the user's calling line identity (CLI) for the connection ( 34 ) over the independant connection ( 23/230 ) or, where CLI is not available, transmitting a one-time code over one of the links for the user to return over the other. The customer can continue to talk to the call centre operative as only DTMF tones are diverted. The call center operative can confirm that the customer's details have been entered and verified by the security system (but is not told what the verification details are) over a separate link ( 43 ) between the call centre platform ( 4 ) and security system ( 3 ).

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14-01-2016 дата публикации

SYSTEM AND METHOD FOR MOBILE NUMBER VERIFICATION

Номер: US20160014603A1
Принадлежит: EARLY WARNING SERVICES, LLC

A mobile phone number that is been provided by a customer during enrollment is verified for use. SS7 protocol data associated with the mobile phone number is collected and evaluated, and if the condition of the subscriber account associated with phone number has changed or is inconsistent with the circumstances surrounding the enrollment of the phone number, the phone number is designated as one that is not to be called. 1. A method for verifying that a phone number is authorized for use , comprising:receiving an identified phone number that has been authorized by customer for use in contacting the customer;sending a request, including the identified phone number, for status data relating to the identified phone number, the status data collected from out-of-band protocol data generated when communications are made using the identified phone number;evaluating the status data for an eligible condition for the identified phone number, the eligible condition indicative that the authorized number is associated with the customer for receiving communications; anddetermining whether the identified phone number may be used in contacting the customer, based on the eligible condition.2. The method of claim 1 , wherein the out-of-band protocol data comprises signaling system 7 (SS7) protocol data.3. The method of claim 3 , wherein the SS7 protocol data comprises one or more of (a) a mobile identity creation date claim 3 , indicating a date the identified phone number has been established for use by the customer claim 3 , (b) a network change event date claim 3 , indicating a date that the status of the identified phone number regarding its availability for use has changed claim 3 , and (c) a network status that indicates the availability for use of the identified phone number.4. The method of claim 3 , wherein the method further comprises:determining an enrollment date on which the identified phone number has been provided by the customer;comparing the determined enrollment ...

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21-01-2016 дата публикации

Telecommunication call management and monitoring system with voiceprint verification

Номер: US20160021243A1
Автор: Stephen Lee Hodge
Принадлежит: Global Tel Link Corp

A secure telephone call management system is provided for authenticating users of a telephone system in an institutional facility. Authentication of the users of the telephone call management system is accomplished by using a personal identification number, biometric means, and/or radio frequency means. The secure telephone call management system includes accounting software capable of limiting access to the system based on funds in a user's account, and includes management software capable of implementing widespread or local changes to the system. The system monitors a conversation in the telephone call to detect a presence of a first characteristic in audio of the conversation, and terminates the telephone call if the first characteristic does not match a second characteristic of biometric information of a user or a called party.

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22-01-2015 дата публикации

Techniques For Providing Multimedia Communication Services To A Subscriber

Номер: US20150023220A1
Принадлежит:

A technique for providing multimedia communication services to a subscriber includes receiving a communication query for the subscriber, the communication query having an associated requested communication mode. The technique also includes servicing the communication query for the subscriber using the requested communication mode when the requested communication mode corresponds to one of one or more selected communication modes. 1. A method , comprising:receiving, by a system comprising a processor, a communication query from a network element of a communication network to access a subscriber device by a requested communication mode at a requested time period;retrieving, by the system, an address pointer from a telephone number mapping server corresponding to the subscriber device and the requested communication mode to identify a retrieved address pointer;determining, by the system, whether the retrieved address pointer indicates that the requested communication mode for the subscriber device is enabled or disabled for the requested time period;transmitting, by the system, a rejection of the communication query to the network element responsive to determining that the requested communication mode is disabled for the request time period; andservicing, by the system, the communication query for the subscriber device using the requested communication mode responsive to determining that the requested communication mode is enabled for the requested time period.2. The method of claim 1 , further comprising modifying claim 1 , by the system claim 1 , the address pointer associated with the subscriber device at the telephone number mapping server to disable the requested communication mode.3. The method of claim 2 , further comprising receiving claim 2 , by the system claim 2 , a request from the subscriber device to disable the requested communication mode.4. The method of claim 2 , wherein the request is generated according to a personal profile of a subscriber ...

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17-04-2014 дата публикации

System, method, and apparatus for using alternative numbers for routing voice calls and short messages in a communications network

Номер: US20140106701A1
Принадлежит: Tango Networks Inc

A system, method and computer-readable medium for allowing the use of an alternative numbering plan for delivering short messages to mobile subscribers using the public mobile telephone network is provided. Users of an enterprise or other closed networks as well as users that are not part of a closed network may send short messages destined to members of an enterprise or closed network from their mobile or other devices associated with their subscription using an alternative enterprise directory number, such as the office number, instead of the mobile number, such that the alternative number will be presented at the destination device as the originating number. The recipient may respond to the message by addressing the response to the enterprise or closed network number, and the response may be delivered to the originator's mobile or other device associated with their subscription.

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28-01-2016 дата публикации

CLASS 4 LONG DISTANCE SOFTSWITCH NETWORK WITH INTEGRATED CLASS 5 APPLICATION SERVICES

Номер: US20160028778A1
Принадлежит:

A telecommunication system including a class 4 long distance softswitch network with one or more a core routing engines and one or more class 5 application servers. The class 4 long distance softswitch network further includes at least one edge device, which may be in the form of a session border controller or media gateway, with at least one connection, including PRI, SS7 and TDM connections, to at least one customer premise equipment of at least one retail customer, which may be an enterprise customer. The class 5 application server is configured to provide the customer with class 5 services within the class 4 network. 1. A telecommunication system comprising:a class 4 long distance softswitch network comprising a core routing engine and a class 5 application server, the class 4 long distance softswitch network further comprising at least one edge device with at least one direct connection to at least one customer premise equipment of at least one retail customer, the class 5 application server configured to provide the retail customer with class 5 services within the class 4 network, and the core routing engine configured to determine whether to process a call with class 5 services.2. The telecommunication system of wherein the class 4 network provides enterprise session initiation protocol trunking services.3. The telecommunication system of wherein the class 5 application server is configured to perform call admission control.4. The telecommunication system of wherein the at least one edge device includes at least one session border controller and at least one media gateway.5. The telecommunication system of wherein the at least one direct connection includes a first direction connection between the at least one session border controller providing session initiation protocol messaging services for a first enterprise retail customer of the at least one retail customer6. The telecommunication system of wherein the at least one direct connection includes a second ...

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26-01-2017 дата публикации

SYSTEM AND METHOD FOR SECURE TRANSMISSION OF DATA SIGNALS

Номер: US20170026516A1
Автор: WESTLAKE Colin Philip
Принадлежит:

The present invention provides systems and methods for controlling signaling data transmitted over a communication system between a first party and a second party. The system comprises a first communication channel configured to carry communication data, wherein the communication data comprises the content of the communication between the first party and the second party; a second communication channel configured to carry signaling data, wherein the signaling data comprises data relating to the first communication channel and sensitive data transmitted by the first party; a signaling processor configured to receive the signaling data from the first party via the second communication channel, modify the signaling data to remove or replace at least some of the sensitive data in the signaling data, and transmit the modified signaling data via the second communication channel to the second party. 1. A system for concurrently transmitting communication data and signaling data over a communication system from a first party to a second party comprising:a communication channel configured to carry the communication data, wherein the communication data comprises content of the communication between the first party and the second party;a signaling channel configured to carry the signaling data, wherein the signaling data comprises data relating to the communication channel and sensitive data transmitted by the first party; anda signaling processor configured to receive the signaling data from the first party via the signaling channel, modify the signaling data to remove or replace at least some of the sensitive data in the signaling data, and transmit the modified signaling data via the signaling channel to the second party.2. The system of claim 1 , wherein the signaling processor is further configured to analyze the signaling data to detect the sensitive data within the signaling data.3. The system of claim 2 , wherein the signaling processor is configured to modify the ...

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28-01-2016 дата публикации

ENABLING ECSFB IN HETNETS

Номер: US20160029254A1
Принадлежит: Intel Corporation

An enhanced Circuit Switch Fallback enabled Heterogeneous Network (HETNET) is provided by the present invention in which the 1×IWS functionality is co-located with LTB eNB. It also tunnels 1×RTT over LTE messages directly to the Convergence Server over SIP, Further, it enables distributed PN-FAP identification determination. A mobile management, entity is configured to maintain multiple 1×CS IWS tunnels. Multiple 1×CS IWS tunnels are established, by using the same S1 tunnel end point used in establishing a Borne evolved Node B Gateway of the HETNET. The HETNET is configured to provide a correct FAP identification to the convergence server so that handover preparation can be done. 1. An enhanced Circuit Switch Fallback (eCSFB) enabled Heterogeneous Network (HETNET) comprising:a. a macro access network;b. a femtocell access network including a 1×FAP and a co-located 1×IWS arranged in an enabled Node B; andc. a mobile management entity coupled to the macro access network and the femtocell access network.wherein the mobile management entity is configured to tunnel 1×RTT directly to a convergence server over SIP.2. The HETNET of wherein the mobile management entity is configured to maintain multiple 1×CS IWS tunnels.3. The HETNET of wherein multiple 1×CS IWS tunnels are established by using the same S1 tunnel end point used in establishing a Home evolved Node B Gateway of the HETNET.4. The HETNET of configured to provide a correct FAP identification to the convergence server so that handover preparation can be done.5. The HETNET of wherein the correct FAP identification is provided using FSM/EMS assisted inter-RAT NRT.6. The HETNET of wherein the correct FAP identification is provided using automated NET discovery based on user equipment tracking.7. The HETNET of wherein fee correct FAP identification is provided using 1×RTT FAP based measurements.8. The HETNET of wherein the correct FAP identification is provided using multiple target preparation and NRT optimization.9. ...

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29-01-2015 дата публикации

Integrated information communication system

Номер: US20150030013A1

To provide an integrated information communication system without using dedicated lines or the Internet, ensuring communication speed, communication quality, communication trouble countermeasures in a unified manner, wherein security and reliability in communication is ensured. The system is comprised of an access control apparatus for connecting a plurality of computer communication networks or information communication equipment to each, and a relay device for networking the aforementioned access control apparatus, the system having functions for performing routing by transferring information by a unified address system, and is configured such that the aforementioned plurality of computer communication networks or information communication equipment can perform communications in an interactive manner.

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28-01-2021 дата публикации

SYSTEM AND METHOD FOR FILTERING ACCESS POINTS PRESENTED TO A USER AND LOCKING ONTO AN ACCESS POINT

Номер: US20210029547A1
Принадлежит:

The present invention filters access points presented to a user and locks onto an access point. The present invention includes an access point filtering unit and an access point locking unit. The access point filtering unit determines the access points that are accessible by a client device and then filters them to present only the access points that are acceptable to under a security policy in force. The access point locking unit has a plurality of operating modes and can lock onto a user selected access point, a security policy prescribed access point, or the access point with the best signal profile. The present invention also includes several methods such as: a method for filtering access points for presentation to the user, a method for locking onto an access point selected by the user, a method for locking onto an access point with the best signal profile, and a method for locking onto an access point prescribed by a security policy for a given location. 1. (canceled)2. A method comprising: determining that one or more first criteria are satisfied, including a criterion that is satisfied when a first security feature associated with the electronic device is present;', 'in response to determining that the one or more first criteria are satisfied, automatically applying a first security policy to a communication session associated with the electronic device;', 'determining that one or more second criteria are satisfied, including a criterion that is satisfied when a second security feature associated with the electronic device is present; and', 'in response to determining that the one or more second criteria are satisfied, automatically applying a second security policy to the communication session associated the electronic device., 'at an electronic device with a processor and memory3. The method of claim 2 , wherein the one or more first criteria include a criterion that is satisfied based on a location of the electronic device.4. The method of claim 2 , ...

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05-02-2015 дата публикации

SYSTEMS AND METHODS FOR PROVIDING ENHANCED TELEPHONE SERVICES

Номер: US20150036553A1
Автор: Vagelos Ted
Принадлежит: Frontier Communications Corporation

A switching platform comprises a first, second, and third circuit. The first circuit is adapted for recognizing a first set of tones generated by art originating telephone line. The originating telephone line is associated with a call origin telephone number. The first set of tones represents a request for a telephone conference over a conference line. The second circuit is adapted for determining whether the call origin telephone number is included in a set of defined telephone numbers permitted to connect to the conference line without further authentication. The third circuit is adapted for establishing a telephone connection between the originating telephone line and the conference line if the set of defined telephone numbers includes the call origin telephone number. 1. A method , executed by a computerized telephone services device , for providing enhanced telephone services comprising:recognizing, by the computerized telephone services device, tones generated by an originating telephone line, the originating telephone line being associated with a calling telephone number, wherein the tones are associated with a common number assigned to two or more telephone lines of the called party;determining, by the computerized telephone services device, whether one of the two or more telephone lines of the called party is in use;determining, by the computerized telephone services device, whether the calling telephone number associated with the originating telephone line is included in a set of defined telephone numbers permitted to connect to the in-use telephone line of the called party without requiring authentication of the calling party; andestablishing, by the computerized telephone services device, a telephone connection between the originating telephone line and the in-use telephone line of the called party if the set of defined telephone numbers includes the calling telephone number.2. The method of claim 1 , wherein determining whether the calling telephone ...

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02-02-2017 дата публикации

METHODS, SYSTEMS, AND COMPUTER READABLE STORAGE DEVICES FOR DETERMINING WHETHER TO FORWARD REQUESTS FROM A PHYSICAL TELEPHONE NUMBER MAPPING SERVICE SERVER TO A VIRTUAL TELEPHONE NUMBER MAPPING SERVICE SERVER

Номер: US20170034117A1
Принадлежит: AT&T Intellectual Property I, L.P.

A determination is made whether to forward a request for communication services associated with a specific number range from a physical telephone number mapping service server to a virtual telephone number mapping service server. Responsive to determining to forward the request, a determination is made whether a virtual telephone number mapping service instance has been provisioned to handle requests associated with the specific number range. 1. A method , comprising:determining, by a processor in a physical telephone number mapping service server, whether to forward a request for communication services associated with a specific number range to a virtual telephone number mapping service server; andresponsive to a determining to forward the request, determining, by the processor, whether a virtual telephone number mapping service instance has been provisioned to handle requests associated with the specific number range.2. The method of claim 1 , further comprising claim 1 , responsive to determining not to forward the request claim 1 , handling the request by the physical telephone number mapping service server.3. The method of claim 1 , further comprising claim 1 , responsive to determining that the virtual telephone number mapping service instance has been provisioned to handle the requests associated with the specific number range claim 1 , forwarding the request to the virtual telephone number mapping service instance.4. The method of claim 1 , further comprising claim 1 , responsive to determining that the virtual telephone mapping service instance has not been provisioned to handle the requests associated with the specific number range claim 1 , handling the request by the physical telephone number mapping service server.5. The method of claim 1 , wherein determining whether to forward the request is based on configuration data associated with the physical telephone number mapping service server indicating whether the whether the physical telephone number ...

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02-02-2017 дата публикации

Method and Device to Operate Phone with a Single Key

Номер: US20170034330A1
Автор: Zhang Rui, Zhang Yimin
Принадлежит:

The present invention discloses a method to operate phone using a single key, first provide method to form ‘key value’: (1) When the key is just closed, plus 1 to the ‘key value’, (2) When the key closing time is greater than time t1, form ‘closing key value’, (3) When the key closing time is smaller than t1 and the immediate key releasing time is greater than t2, store the key value as ‘releasing key value’, (4) When the key closing time is smaller than t1 and the immediate key releasing time is smaller than t2, skip to step (1). The ‘releasing key values’ can be used to represent the numerical and alphabetical keys on typical keypad of phone, while the ‘closing key values’ can be used to represent the function key on typical keypad of phone, so that the complete function of a typical phone keypad can be realized using only one key. Therefore, a phone can be made very small, which can be especially useful when installed on small Bluetooth device that connects to phone, since not only can it answer incoming phone calls, but also making outgoing phone calls. 1. A method to operate phone using a single key , characterized in that: the method is realized with a single key through the following steps:(1) when the key is just closed, plus 1 to the ‘key value’,(2) judging if the key closing time is less than t1, if yes, skip to step (3); otherwise, skip to step (4);(3) judging if the key releasing time is less than t2, if yes, skip to step (1); otherwise, store the key value as the ‘releasing key value’, exit the key value forming process, and trigger the ‘releasing key value processing procedure’;(4) store the ‘key value’ as the ‘closing key value’, exit the key value forming process, and trigger the ‘closing key value processing procedure’;(5) the key statues enters idle mode and the key value is set to zero;wherein the said step (3), when dialing the phone number associated with the ‘releasing key value’, each time when the key is pressed, a prompt tone corresponding ...

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02-02-2017 дата публикации

Systems and Methods for Making Low-Cost International Phone Calls

Номер: US20170034357A1
Принадлежит:

Various embodiments are described herein that relate to computer hardware and programs for facilitating international calls between individuals. More specifically, an international call system can route voice traffic between multiple telephone network endpoints. For example, a client application (also referred to as a “client”) may be accessed by a traveler on a mobile phone. The client can be configured to connect to a network-accessible application server, which can communicate with a call processing system that routes voice traffic from the traveler's mobile phone to the mobile phone of a desired contact (and, in some embodiments, vice versa). The international call system described herein makes use of the local providers in each country traversed by the traveler. Consequently, low-cost calling can be enabled in both directions, while predictable reachability is guaranteed as the traveler moves between different local providers. 1. A computer-implemented method for routing calls between users of an international call system , the method comprising:receiving, at a client residing on a first computing device associated with a user located in a first country, a first user input indicative of a request to place a call to a second computing device associated with a contact located in a second country; 'wherein the call is routed to a first gateway over a Public Switched Telephone Network (PSTN);', 'sending, by the client to a phone application residing on the first computing device, an inter-process message that causes the phone application to place a call to an access number associated with the contact,'}causing the first gateway to send a first message to a call processing system across an Internet Protocol (IP) network; retrieving a destination number for the contact from an application server;', 'sending the first message to a second gateway over the IP network; and', 'causing the second gateway to place a call to the destination number that is received by a phone ...

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02-02-2017 дата публикации

METHODS, SYSTEMS, AND COMPUTER READABLE STORAGE DEVICES FOR ADJUSTING THE USE OF VIRTUAL RESOURCES PROVIDING COMMUNICATION SERVICES BASED ON LOAD

Номер: US20170034359A1
Принадлежит: AT&T Intellectual Property I, L.P.

The use of virtual resources providing communication services is adjusted based on load. A determination is made, for a specific number range, whether the virtual telephone number mapping service instances dedicated to the specific number range are being. If the virtual telephone number mapping service instances dedicated to the specific number range are being underutilized, a virtual telephone number mapping service instance dedicated to the specific number range is selected for removal. A determination is also made, for the specific number range, whether a number of virtual telephone number mapping service instances dedicated to the specific number range is sufficient to handle requests for communication services. If the number of virtual telephone number mapping service instances dedicated to the specific number range is not sufficient to handle the requests, instantiation of a virtual telephone mapping instance is initiated. 1. A method , comprising:receiving, by a processor, information indicative of performance of virtual telephone number mapping service instances dedicated to a specific number range associated with requests for communication services;determining, by the processor, for the specific number range, whether the virtual telephone number mapping service instances dedicated to the specific number range are being underutilized based on the received information;responsive to determining that the virtual telephone number mapping service instances dedicated to the specific number range are being underutilized, selecting, by the processor, a virtual telephone number mapping service instance dedicated to the specific number range for removal; andinitiating, by the processor, removal of the selected virtual telephone number mapping service instance for handling the requests.2. The method of claim 1 , further comprising claim 1 , reserving the selected virtual telephone mapping service instance as a resource for handling the requests.3. The method of claim 1 ...

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04-02-2016 дата публикации

CALL EXTENDER FOR INTER-CARRIER NETWORK SWITCH

Номер: US20160036989A1
Принадлежит:

A method of extending calls includes receiving a request to establish a call session with a called party, and establishing the call session between the calling party and the called party. The method further includes monitoring the length of the established call session, and receiving a message requesting a termination of the call session. Still further, the method includes, in response to receiving the message requesting the termination of the call session, maintaining the established call session for an additional time period, wherein the additional time period plus the monitored length of the established call session is equal to or greater than a threshold of time. 1. A method of extending calls , the method comprising:receiving, at a call router of an inter-carrier network switch from a first provider corresponding to a calling party, a request to establish a call session with a called party;establishing, by the call router, the call session between the calling party and the called party through a Private Packet Network Backbone Exchange (PPNBE) of the inter-carrier network switch to an exchange connected to a second provider corresponding to the called party;monitoring, by a call extender of the inter-carrier network switch, the length of the established call session;receiving, at the call router of an inter-carrier network switch from the first provider, a message requesting a termination of the call session;in response to receiving the message requesting the termination of the call session, maintaining, by the call extender, the established call session for an additional time period, wherein the additional time period plus the monitored length of the established call session is equal to or greater than a threshold of time associated with surcharges assessed by an operator of the exchange connected to the second provider.2. The method of claim 1 , further comprising:after maintaining the established call session for the additional time period, forwarding the ...

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04-02-2016 дата публикации

SELF-HEALING INTER-CARRIER NETWORK SWITCH

Номер: US20160036990A1
Принадлежит:

A vendor evaluator system enables real-time or near real-time efficiencies of an inter-carrier switch. At a nominal periodic rate (e.g., every fifteen minutes or less), the vendor evaluator system updates a performance map of vendor exchanges to which calls may be routed from the switch. Specifically, in real-time or in near real-time, an updated performance score is generated for each vendor exchange and/or for each NPA-NXX code that is connected to the switch based on a weighted average of post dial delay, average call hold time, and attempt-seizure ratio. The score is compared against performance thresholds to determine whether or not the respective exchange and/or code should be included in a pool of acceptably-performing, candidate vendor exchanges from which a particular exchange is selected to service a call. Such techniques prevent poorly performing vendors from servicing calls routed by the inter-carrier network switch in real time or near real-time. 1. A method for delivering calls based on vendor performance , the method comprising:determining, for each vendor exchange of a plurality of vendor exchanges connected to a private packet network backbone exchange (PPNBE), a respective performance score based on post-dial delay, attempt-seizure ratio, and average call hold time of a set of calls routed by the PPNBE to the each vendor exchange during a time period occurring immediately prior to an initiation of an execution of the method;comparing each respective performance score to one or more thresholds, and determining whether or not the each respective performance score is acceptable based on the comparison;when the each respective performance score is determined to be acceptable, including the corresponding vendor exchange in a pool of candidate vendor exchanges from which a terminating or forwarding exchange to service a call is obtained;when the each respective performance score is determined to be unacceptable, excluding the corresponding vendor ...

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04-02-2016 дата публикации

AUTO-DIALER DETECTOR FOR INTER-CARRIER NETWORK SWITCH

Номер: US20160036991A1
Принадлежит:

To maximize efficiencies and to reduce termination costs of inter-carrier exchanges, an auto-dialer detection system enables an inter-carrier network switch to detect, in real-time or in near real-time, calls that are originated by auto-dialers. A call router of the switch may receive an incoming call attempt that includes a particular Automatic Number Identification (ANI). The auto-dialer detection system allows for a real-time or near-real time determination, based on the ANI and contents of a cache during a sliding window of time coincident with the reception of the origination, whether or not the call should be routed through the switch. Further, the auto-dialer detection system provides a real-time or a near real-time update to the cache contents to enable further real-time or near-real time detection and blocking of auto-dialed calls. Overrides to the cache (e.g., to always allow and/or to always block calls that include certain ANIs) may be provided. 1. A method for detecting auto-dialed calls in real-time or near real-time , the method comprising:receiving, at a call router of an inter-carrier network switch communicatively connected to a data storage device storing call data records (CDRs), an indication of an incoming call received at the inter-carrier network switch and including an Automatic Number Identification (ANI);based on the reception of the indication of the incoming call, causing an indication of a call attempt corresponding to the ANI to be stored in a cache;determining, by the call router, whether to allow or to deny the incoming call based on contents of the cache;when the incoming call is determined to be allowed, processing the incoming call through a Private Packet Network Backbone Exchange (PPNBE) of the inter-carrier network switch to a terminating exchange, and causing a call data record for the incoming call to be stored in the CDR data storage device; andwhen the incoming call is determined to be denied, not processing the incoming ...

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01-02-2018 дата публикации

Telecommunications Addressing System and Method

Номер: US20180034959A1
Автор: THANGE Maqsood A.
Принадлежит:

A telecommunications addressing system/method allowing selection of a telephone instrument device (TID) using arbitrary identifiers is disclosed. The system/method allows a source TID (STD) to select a target TID (TTD) by the use of a Target Telephone Identifier (TTI) data string rather than a traditional numeric telephone identification (NTI). This TTI is then indexed within a TTI/NTI mapping server (TMS) that functions as a hierarchical repository of TTI/NTI mappings. STD/TTD communication is established by first performing a lookup of the STD-selected TTI within the TMS to identify the NTI of the TTD. Once the NTI of the STD has been identified by the TMS, communication between the STD and TTD is established using the NTI via the normal public switched telephone network (PSTN). TMS TTI lookup may be performed via STD TID web application and/or via PSTN infrastructure interface. 1. A telecommunications addressing method , said method operating in conjunction with a telecommunications addressing system , said system comprising:(a) telephone mapping server (TMS);(b) telephone mapping database (TMD);(c) source telephone instrument device (STD);(d) target telephone instrument device (TTD); and(e) computer communication network (CCN);whereinsaid TMS is configured to store information that identifies a telephone instrument device (TID) in said TMD;said TMS is configured to locate a numeric telephone identifier (NTI) within said TMD using a target telephone identifier (TTI) data string as the locating index;said NTI permits said TID to be accessed via a public switched telephone network (PSTN); andsaid TMS is configured to initiate a telephone call via said PSTN between said STD and said TTD using said NTI;wherein said method comprises the steps of:(1) entering a TTI via a user interface on said STD;(2) transmitting said TTI to said TMS via said CCN;(3) indexing said TTI within said TMD to retrieve said NTI associated with said TTI;(4) transmitting said NTI to said STD ...

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01-02-2018 дата публикации

System and method for call termination based on an over-the-top (OTT) call service of a calling party

Номер: US20180034972A1
Автор: Pande Avanish
Принадлежит:

A system and method for connecting a call from an originating endpoint of a calling party to a terminating endpoint of a called party while accounting for whether the calling party is using an over-the-top (OTT) call service. In response to receiving a dialed number, or other call information, a computer system determines whether to connect the call to the terminating endpoint of the called party via an OTT call service, or not, based on the geolocation of the originating endpoint. For example and without limitation, based on the originating endpoint currently being located within a country that is on a list of developed countries or not currently being located within a country that is on a list of developing countries, the computer system might deem a particular OTT call service to be acceptable for handling the call for the terminating endpoint. 1. A system for connecting a call from an originating endpoint of a calling party to a terminating endpoint of a called party , comprising: receive a dialed number for the call when placed from the originating endpoint,', 'determine an over-the-top (OTT) call service subscribed to by the called party, wherein the OTT call service of the called party is accessible from the terminating endpoint via the Internet, and', "generate a signal for controlling the call, wherein the signal indicates to route the call i) via the dialed number such that the call is switched via the terminating endpoint's local telecommunications service provider and access network or ii) via the OTT call service of the called party, based on whether the call is placed from the originating endpoint through an OTT call service of the calling party; and"], 'a computer system configured toa networking device configured to route the call to the terminating endpoint, based on the signal.2. The system of wherein the OTT call service of the calling party is delivered through a first access network claim 1 , and wherein the signal is further based on a first ...

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31-01-2019 дата публикации

SYSTEM AND METHOD FOR ANALYTICS WITH AUTOMATED WHISPER MODE

Номер: US20190037076A1
Принадлежит: AT&T Intellectual Property I, L.P.

A service session is facilitated via a packet switched network; in the service session, user equipment participates in an interactive communication exchange with an agent via a first interaction mode, and the interactive communication exchange is based on a user inquiry. The interactive communication exchange is monitored and a determination is made that a consultation service would facilitate resolution of the user inquiry. A service resource is associated with the service session responsive to determining that the consultation service would facilitate the resolution; the service resource provides consultation to the agent via a second interaction mode without exposing the consultation to the user equipment. The consultation elevates an experience level employed in the first service session towards resolution of the user inquiry. 1. A method , comprising:facilitating, by a processing system including a processor, a customer service session via a communication network, wherein customer equipment participates in a first interactive communication exchange with a first customer service agent via a network connection using a first interaction mode;monitoring, by the processing system, the first interactive communication exchange;associating, by the processing system based on the monitoring, a second customer service agent with the customer service session, wherein the second customer service agent provides a consultation service to the first customer service agent in a second interactive communication exchange using a second interaction mode different from the first interaction mode, without exposing the consultation service to the customer equipment, wherein the first customer service agent provides, in accordance with the monitoring, an assessment of a customer mood to the second customer service agent; anddetermining, by the processing system based on the assessment, whether to associate an additional customer service resource with the customer service session.2. The ...

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30-01-2020 дата публикации

SYSTEM AND METHOD FOR DELIVERY OF VOICEMAILS TO HANDHELD DEVICES

Номер: US20200036835A1
Принадлежит:

Briefly, a variety of embodiments, including the following, are described: a system embodiment and methods that allow random access to voice messages, in contrast to sequential access in existing system embodiments; a system embodiment and methods that allow for the optional use of voice recognition to enhance usability; and a system embodiment and methods that apply to the area of voicemail. 129-. (canceled)30. An apparatus comprising:a mobile device able to engage in cellular communications comprising an integrated voicemail system in which visual voicemail access capability is provided for the integrated voicemail system;wherein the mobile device includes a capability to transcribe voicemail messages to text messages so as to provide the visual voicemail access capability.31. The apparatus of claim 30 , wherein the mobile device comprises a capability to display a voicemail inbox of the integrated voicemail system claim 30 , the voicemail inbox to comprise a visual voicemail inbox to provide the visual voicemail access capability.32. The apparatus of claim 31 , wherein the visual voicemail inbox also to include a capability to display one or more visual voicemail indicators.33. The apparatus of claim 32 , wherein the one or more visual voicemail indicators to comprise at least one of the following: time of a telephone call associated with a voicemail message; date of a telephone call associated with a voicemail message; and/or identification of a telephone number associated with a voicemail message.34. The apparatus of claim 33 , wherein claim 33 , for the one or more visual voicemail indicators to comprise identification of the telephone number associated with the voicemail message claim 33 , the visual voicemail inbox further includes the capability to identify and display the caller of the telephone number in place of displaying the telephone number in at least some cases.35. The apparatus of claim 32 , wherein the capability to display one or more visual ...

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12-02-2015 дата публикации

TELEPHONE NETWORK ACCESS DEVICE

Номер: US20150043573A1
Принадлежит:

A telephone network access device switchable between a VOIP service and a PSTN service includes an Rj-11 port, a voice splitter, a first relay, a second relay, a SLIC chip, a CPU, and a ring detection unit. The ring detection unit outputs a first switch signal in response to detecting a ring signal of a PSTN communication. The second relay connects the telephone to the VOIP network in response to a power on event. The CPU controls the first relay and the second relay to connect the telephone to the PSTN network in response to the CPU receiving the first switch signal, and controls the second relay to reconnect the telephone to the VOIP network when the PSTN communication ends. 1. A telephone network access device comprising:an Rj-11 port coupled to a public switched telephone network (PSTN) network;a voice splitter coupled to the Rj-11 port;a subscriber line interface circuit (SLIC) chip coupled to a voice over Internet protocol (VOIP) network;a first relay coupled to the voice splitter;a second relay coupled to the first relay, the SLIC chip, and a telephone;a ring detection unit coupled between the first relay and the voice splitter, wherein the ring detection unit outputs a first switch signal in response to detecting a ring signal from a PSTN communication; anda central processing unit (CPU) coupled to the first relay, the second relay, the SLIC chip, and the ring detection unit, and receiving the first switch signal from the ring detection unit;wherein the second relay connects the telephone to the VOIP network in response to a power on event, the CPU controls the first relay and the second relay connecting the telephone to the PSTN network in response to the CPU receiving the first switch signal from the ring detection unit, and controls the second relay reconnecting the telephone to the VOIP network in response to an ending of the PSTN communication.2. The telephone network access device of claim 1 , wherein the first relay comprises two first control ...

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04-02-2021 дата публикации

Apparatus and method for using an intelligent network for analyzing an event external to a signaling network

Номер: US20210037143A1
Автор: Leo Anthony Wrobel
Принадлежит: Individual

A system and method in an intelligent network are provided for analyzing an event external to the intelligent network. A set of intelligent network subscribers is selected in a selected geographic area. Next, signal message data is obtained from a signaling network in the intelligent network, wherein the signal message data indicates status of subscribers in the set of intelligent network subscribers. External data related to the selected geographic area is obtained, wherein the external data is external to the intelligent network. The signal message data and the external data are then fused to produce synthetic data. In response to the synthetic data, an alert of an external event is produced. The alert can be graphically represented on a display.

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11-02-2016 дата публикации

TONING CONTROL METHOD OF WIFI DEVICE SETTING BY SMART DEVICE

Номер: US20160043895A1
Принадлежит:

The present invention provides a toning control method for setting a WiFi device by a smart device so that the WiFi device can link with Internet through a WiFi AP. The present invention is different from the prior art that a smart phone uses WiFi signal to set an IP camera, and is also different from the prior art that a smart phone generates QR Code for setting an IP camera. The present invention adopts methods of DTMF (Dual-Tone Multi-Frequency) or FSK (Frequency-Shift Keying) for toning encoding, and sends the toning encoding through a loudspeaker of the smart device to a toning decoder of the WiFi device for toning decoding, then performs the setting and linking of the WiFi device. 1. A toning control method of a WiFi device setting by a smart device , in a system of linking a WiFi device to Internet through a WiFi AP , a smart device is used for setting the WiFi device by the toning control method , comprising steps as below:a. a toning decoder of the WiFi device begins recording;b. an application program is clicked out from a screen of the smart device, an SSID and a password of the WiFi AP are filled in;c. encoding the SSID and the password by the smart device sequentially to generate a series of toning signals; a loudspeaker of the smart device sends the series of toning signals to the toning decoder for decoding;d. after decoding by the toning decoder successfully, the WiFi device is set with the SSID and the password of the WiFi AP; the WiFi device sends out the SSID and the password of the WiFi AP to the WiFi AP;e. after an authentication by the WiFi AP successfully, the WiFi device can then link with Internet through the WiFi AP, and generate a correct lamp signal;f. if the authentication by the WiFi AP is failed, then the WiFi device generates an abnormal lamp signal, go back to step a for resetting;g. in step d, if the decoding is failed, then the WiFi device generates a wrong lamp signal, go back to step c, the smart device encodes again the SSID and ...

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11-02-2016 дата публикации

Telecommunications Addressing System and Method

Номер: US20160044166A1
Автор: THANGE Maqsood A.
Принадлежит:

A telecommunications addressing system/method allowing selection of a telephone instrument device (TID) using arbitrary identifiers is disclosed. The system/method allows a source TID (STD) to select a target TID (TTD) by the use of a Target Telephone Identifier (TTI) data string rather than a traditional numeric telephone identification (NTI). This TTI is then indexed within a TTI/NTI mapping server (TMS) that functions as a hierarchical repository of TTI/NTI mappings. STD/TTD communication is established by first performing a lookup of the STD-selected TTI within the TMS to identify the NTI of the TTD. Once the NTI of the STD has been identified by the TMS, communication between the STD and TTD is established using the NTI via the normal public switched telephone network (PSTN). TMS TTI lookup may be performed via STD TID web application and/or via PSTN infrastructure interface. 1. A telecommunications addressing method , said method operating in conjunction with a telecommunications addressing system , said system comprising:(a) telephone mapping server (TMS);(b) telephone mapping database (TMD);(c) source telephone instrument device (STD);(d) target telephone instrument device (TTD); and(e) computer communication network (CCN);whereinsaid TMS is configured to store information that identifies a telephone instrument device (TID) in said TMD;said TMS is configured to locate a numeric telephone identifier (NTI) within said TMD using a target telephone identifier (TTI) data string as the locating index;said NTI permits said TID to be accessed via a public switched telephone network (PSTN); andsaid TMS is configured to initiate a telephone call via said PSTN between said STD and said TTD using said NTI;wherein said method comprises the steps of:(1) entering a TTI via a user interface on said STD;(2) transmitting said TTI to said TMS via said CCN;(3) indexing said TTI within said TMD to retrieve said NTI associated with said TTI;(4) transmitting said NTI to said STD ...

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08-02-2018 дата публикации

METHOD FOR COLLECT CALL SERVICE BASED ON VOIP TECHNOLOGY AND SYSTEM THEREOF

Номер: US20180041642A1
Автор: KWON Ey-Taeg
Принадлежит:

One embodiment of the present invention provides a collect call method and system thereof, more particularly, in order to charge the called party with a uniform toll for collect call, which is determined by only the type and location of called party terminal. In one embodiment, the collect call method, system and a counsel service providing method use a free VoIP network for part of the voice call link and a charge PSTN network for the rest of the voice call link. In one embodiment, if the first link corresponding to the collect call request is established, the collect call switch calls the called party terminal to establish the second link, and billing on the second link is initiated. 1. A post-paid collect call system based on voice over Internet protocol (VoIP) , comprising:a calling gateway configured to receive a collect call request generated by a caller telephone, the collect call request including a collect call identifier and a phone number of a called terminal;a called gateway configured to receive the collect call request via a VoIP network; and a) establish a first communication link only between the collect call switch and the called gateway upon receiving the collect call request from the caller telephone;', 'b) subsequent to establishing the first communication link, establish a second communication link via the PSTN between the collect call switch and the called terminal;', 'c) transmit a recorded caller identification to the called terminal when the second communication link is established; and', 'd) connect the first communication link and the second communication link based on a collect call acceptance from the called terminal to establish a voice call link between the caller-telephone and the called terminal;, 'a collect call switch, wherein the collect call switch is in a Public Switched Telephone Network (PSTN), and the collect call switch is configured towherein the collect call switch is further configured to determine a collect call toll ...

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16-02-2017 дата публикации

Command and Control of Devices and Applications by Voice Using a Communication Base System

Номер: US20170047068A1

A first communication path for receiving a communication is established. The communication includes speech, which is processed. A speech pattern is identified as including a voice-command. A portion of the speech pattern is determined as including the voice-command. That portion of the speech pattern is separated from the speech pattern and compared with a second speech pattern. If the two speech patterns match or resemble each other, the portion of the speech pattern is accepted as the voice-command. An operation corresponding to the voice-command is determined and performed. The operation may perform an operation on a remote device, forward the voice-command to a remote device, or notify a user. The operation may create a second communication path that may allow a headset to join in a communication between another headset and a communication device, several headsets to communicate with each other, or a headset to communicate with several communication devices.

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06-02-2020 дата публикации

Method and program product for electronic communication based on user action

Номер: US20200045183A1
Автор: Timur Voloshin
Принадлежит: Individual

A system, method and program product for electronic communication includes checking if a current called number of a subscriber is on a list of unsuccessful calls made by an onward caller. A determination is made that the current called number is on the list of unsuccessful calls. A calling route used to establish a connection between onward caller and subscriber is concluded as a low-quality route. A subscriber ID is extracted and included on the list of low quality routes. A request is made for a list of all possible routes for communication between onward caller and receiver user. A comparison list of all possible routes and low quality routes is made. A route not earlier marked as a low-quality route is identified and selected as an alternative route. An electronic communication between onward caller and receiving user is established using the alternative route.

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18-02-2021 дата публикации

MOBILE APPLICATION FOR PROVIDING MULTIPLE SECOND LINE NUMBERS ON SINGLE MOBILE DEVICE

Номер: US20210051452A1
Принадлежит: Movius Interactive Corporation

A mobile application and a method are described for servicing a second line service (“SLS”) based communication request originating from a subscriber's telecommunications device (“TD”) even if the call signal does not include sufficient information to identify the phone number from which the subscriber initiated the call. The method involves associating the SLS phone number of the subscriber, the primary number of the subscriber and the primary number of a third party via a special relationship number. 2. The method of claim 1 , further comprising the action of:presenting on a display of the mobile communications device, an indication that an incoming call from the particular third party is being received for the particular additional network accessible number.3. The method of claim 2 , further comprising the actions of:receiving a call accept actuation of a user interface of the mobile communications device; andterminating the incoming call at the mobile communications device, wherein the incoming call is established over a cellular network.4. The method of claim 3 , further comprising:receiving a call origination actuation of a user interface of the mobile communications device, wherein the call origination actuation includes the particular relationship number as the dialed number;initiating an outgoing call over a telecommunications network to the particular relationship number and setting the calling line identifier of the outgoing call to the cellular service provider number, whereby the telecommunications network will recognize the particular relationship number as being serviced by the second line service platform and route the outgoing call to the second line service platform, whereby the second line service platform will direct the outgoing call to the particular third party calling number associated with the particular third party with the calling line identifier set to the particular additional network number associated with the particular relationship ...

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15-02-2018 дата публикации

NETWORK DOMAIN SELECTION

Номер: US20180048766A1
Принадлежит:

For an incoming session intended for a user element, a domain selection function (DSF) is provided in a multimedia subsystem (MS) to select a circuit-switched subsystem (CS) or MS domain to use when routing the incoming session to the user element. Regardless of the domain in which incoming sessions are originated, the incoming sessions intended for the user element are routed to an S-CSCF in the MS. The S-CSCF directly or indirectly employs the DSF to determine whether to use the MS or the CS for terminating the incoming session. Based on available domain selection criteria, the DSF will select an appropriate domain, such as the MS or CS, to use for routing the incoming session to the user element. The domain selection decision of the DSF is provided to the S-SCSF, which will proceed by routing the incoming session to the user element via the selected domain. 1. A method comprising:providing an application server configured to provide a plurality of services for a call; andproviding a continuity control function (CCF) that is invoked as a service in a chain of the services provided by the application server, wherein the CCF is configured to effect a transfers of the call from a first client to a second client.2. The method of comprising passing all call signaling messages for the call through the application server.3. The method of wherein a circuit switched subsystem (CS) provides an access signaling leg for the call claim 1 , and the access signaling leg and a remote signaling leg towards a remote endpoint for the call are anchored at the CCF in a multimedia subsystem claim 1 , wherein the CCF controls the call.4. The method of claim 3 , wherein claim 3 , if a call transfer is required claim 3 , the CCF maintains the remote signaling leg and establishes a new access signaling leg.5. The method of wherein the remote signaling leg extends through the application server.6. The method of wherein the CCF maintains control of the call and provides necessary call ...

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26-02-2015 дата публикации

METHOD AND APPARATUS FOR PROVIDING SERVICES ACROSS SERVICE DOMAINS

Номер: US20150055561A1
Принадлежит: Alcatel-Lucent USA Inc.

The present invention supports services that span multiple service domains. In one embodiment, the present invention provides service-level interaction between an enterprise network and a non-enterprise network (or networks), thereby enabling enterprise users at remote locations (i.e., at home, on vacation, or at other locations remote from the enterprise location) to use services typically only available to the enterprise users while at the enterprise location (e.g., while in the office). The present invention provides a calendar notification service, whereby a calendar notification originating in an enterprise network is provided to a remote enterprise user via a non-enterprise network. The present invention provides an enterprise dialing plan service, whereby a remote enterprise user may register a remote user device to be able to use an enterprise dialing plan. Once registered, the remote enterprise user may use the enterprise dialing plan from the remote user device to place calls to local enterprise users at local user devices connected to the enterprise network and, similarly, local enterprise users may use the enterprise dialing plan to place calls to the remote enterprise user at the remote user device. 1. A method , comprising: receiving, at an element of a non-enterprise network, a request to register a remote user device of a user to use an enterprise dialing plan of an enterprise network of an enterprise;', 'obtaining enterprise dialing plan information for the enterprise dialing plan of the enterprise network; and', 'storing the enterprise dialing plan information for the enterprise dialing plan of the enterprise network., 'using a processor and a memory for2. The method of claim 1 , wherein obtaining the enterprise dialing plan information for the enterprise dialing plan of the enterprise network comprises:identifying the enterprise associated with the request to register the remote user device to use the enterprise dialing plan of the enterprise ...

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26-02-2015 дата публикации

SINGLE NUMBER SERVICES FOR FIXED MOBILE TELEPHONY DEVICES

Номер: US20150056984A1
Принадлежит: NET2PHONE, INC.

A special-purpose Service Control Point includes a customized application that provides Fixed Mobile Convergence services and interoperates with a Voice Over Internet Protocol (VoIP) network to achieve a single-number fixed mobile convergence overlay network. By determining if a dual-mode cellular/WiFi handset is reachable via a WiFi network, the Service Control Point may redirect incoming and outgoing calls off of the cellular network and onto the WiFi network, thereby reducing cost and providing greater coverage. 1. A system for off-loading telephony connections for a multi-mode phone from a cellular network to a computer data network , the system comprising:a receiver receiving a request for an outgoing call from the multi-mode phone;a controller for determining if the multi-mode phone is reachable via the computer data network; anda bridge being addressed by a first Mobile Station Roaming Number from a pool of available Mobile Station Roaming Numbers for bridging a call between the multi-mode phone and a destination of the request for the outgoing call, wherein the multi-mode phone communicates via the computer data network and the destination of the request for the outgoing call communicates via a public switched telephone network, wherein the first Mobile Station Roaming Number is received from a Service Control Point during call setup.2. The system as claimed in claim 1 , wherein the multi-mode phone is a dual-mode phone where a first mode is a cellular mode and a second mode is a computer data network mode.3. The system as claimed in claim 2 , wherein the computer data network mode is a WiFi mode.4. The system as claimed in claim 1 , wherein the multi-mode phone is a tri-mode phone where a first mode is a cellular mode and second and third modes are two different computer data network modes.5. The system as claimed in claim 4 , wherein the second mode is a WiFi mode.6. The system as claimed in claim 1 , wherein the request for an outgoing call is received ...

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25-02-2016 дата публикации

SS7 MAP/Lg+ TO SIP BASED CALL SIGNALING CONVERSION GATEWAY FOR WIRELESS VoIP E911

Номер: US20160057287A1
Автор: Donald Mitchell
Принадлежит: TeleCommunication Systems Inc

An SS7-based call protocol conversion gateway that translates between circuit-switched SS7 protocols and session initiation protocol (SIP) oriented protocol, allowing an E911 call initiated over a switched network to be routed by a VoIP network. The SS7-based call protocol conversion gateway provides a PSAP with MSAG quality (street address) information about a VoIP dual mode phone user without the need for a wireless carrier to invest in building out an entire VoIP core. Thus, wireless carriers may continue signaling the way they are today, i.e., using the J-STD-036 standard for CDMA and GSM in North America, yet see benefits of a VoIP network core, i.e., provision of MSAG quality location data to a PSAP.

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10-03-2022 дата публикации

SIGNALING GATEWAY APPARATUS, PROTOCOL CONVERSION METHOD, AND PROGRAM

Номер: US20220078287A1
Принадлежит: NEC Corporation

A signaling gateway apparatus (SG) in an IP network, is directly connected to a group unit center (GC) in a PSTN in a layer of MTP level 3 without via a signaling transfer point (STP), converts a M2PA sequence number included in an XCO or an XCA which is a response to the XCO, to a sequence number with a value in a range from 0 to 127 which is a maximum value of a 7-bit unsigned integer and transmits the XCO or the XCA to an opposite apparatus. 1. A signaling gateway apparatus , disposed in an IP (Internet protocol) network and connected by associated network connection in a layer of MTP (Message Transport Part) level 3 with a group unit center in a PSTN (Public Switched Telephone Network) , the signaling gateway apparatus comprising:a receiver;a transmitter;a processor; anda memory storing program instructions executable by the processor to perform:receiving, via the receiver, an extended changeover order signal XCO (eXtended Changeover Order) of M2PA (MTP2 User Peer-to-peer Adaptation Layer) or an extended changeover acknowledgement signal XCA (eXtended Changeover Acknowledgement) which is a response to the XCO;converting a 24-bit M2PA sequence number included in the XCO or the XCA to a value in a range from 0 to 127 which is a maximum value of a 7-bit unsigned integer; andcausing the transmitter to transmit, to an opposite apparatus, the XCO or the XCA including the 24-bit M2PA sequence number with a value in the range from 0 to 127.2. The signaling gateway apparatus according to claim 1 , wherein the processor is configured to perform:checking the 24-bit M2PA sequence number included in the XCO or the XCA,setting the 24-bit M2PA sequence number in the XCO or the XCA to 0, if the 24-bit M2PA sequence number is over 127, andleaving the 24-bit M2PA sequence number included in the XCO or the XCA as it is, if the 24-bit M2PA sequence number is in the range from 0 to 127.3. The signaling gateway apparatus according to claim 1 , wherein the transmitter transmits claim ...

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10-03-2022 дата публикации

CONTROL METHOD CAPABLE OF CONTROLLING MCU AND VIDEO CONFERENCE TERMINAL BY USING USER TERMINAL, AND VIDEO CONFERENCE SYSTEM FOR SAME

Номер: US20220078375A1
Автор: CHA Min Soo
Принадлежит: UPRISM CO., LTD.

A control method that can control an MCU and a video conference terminal using a user terminal, and a video conference system for the control method are proposed. The method of controlling a video conference of the present disclosure includes: a pairing request step in which the user terminal connects to the MCU in a wired or wireless type and then requests pairing with a target video conference terminal to the MCU; a pairing step in which the user terminal forms a control channel between the user terminal and the MCU in accordance with approval of the pairing request by the MCU; and a control step in which the user terminal requests one of control items allowed by the MCU to the MCU after the pairing step. 1. A method of controlling a video conference of a user terminal that can connect to a video conference system including an MCU and at least one video conference terminal connected to the MCU , the method comprising:a pairing request step in which the user terminal connects to the MCU in a wired or wireless type and then requests pairing with a target video conference terminal to the MCU;a pairing step in which the user terminal forms a control channel between the user terminal and the MCU in accordance with approval of the pairing request by the MCU; anda control step in which the user terminal requests one of control items allowed by the MCU to the MCU after the pairing step.2. The method of claim 1 , further comprising a step in which the user terminal claim 1 , after the pairing step claim 1 , uploads sharing data to be provided to the video conference by transmitting at least one of a document file claim 1 , a video file claim 1 , and a screen of the user terminal to be provided to the video conference to the MCU through the control channel claim 1 , instead of the target video conference terminal.3. The method of claim 1 , wherein the pairing request step includes:a step in which the user terminal connects to the MCU in a wired or wireless type;a step in ...

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03-03-2016 дата публикации

A METHOD OF RESOLVING A PORTED TELEPHONE NUMBER INTO A NETWORK RESOURCE IDENTIFIER

Номер: US20160065747A1
Принадлежит:

A method of resolving by an original network/domain of a telephone number of a called party, belonging to a recipient network/domain. The method includes: a) a calling party, belonging to the original domain dispatches a message using the telephone number of the called party as routing identifier; b) a server of the original domain in charge of routing the message, or a DNS server, produces an interrogation key dependent on the telephone number and on a penultimate domain name; c) a DNS request is sent to a DNS server associated with the penultimate domain; d) the DNS server associated with the penultimate domain performs, by using the interrogation key, a search in a database ENUM associated therewith; e) the search provides at least one record containing a return network/domain including the recipient domain; and f) a DNS request is sent to the return domain. 1. An ENUM database stored on a non-transitory computer-readable medium and comprising:at least one CNAME or DNAME record providing a referral domain in response to an interrogation key that is a function of a telephone number of a “called” user, wherein said called user belongs to said referral domain and said telephone number has been ported in with relation to a change in the network/domain of the called user.2. An ENUM database according to claim 1 , wherein the database is associated with a fixed or mobile circuit-switched telephone network.3. An ENUM database according to claim 1 , wherein the database is dedicated entirely to ported telephone numbers.4. A DNS server associated with an ENUM database according to .5. A method enabling a network/domain claim 1 , referred to as an “originating” domain to resolve a telephone number of a user referred to as a “called” user belonging to a network/domain referred to as a “destination domain” claim 1 , the method comprising:a) a “calling” user belonging to said originating domain sending a message using said telephone number of the called user as a routing ...

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20-02-2020 дата публикации

High Availability Voice Over Internet Protocol Telephony

Номер: US20200059508A1
Автор: Escude Luke, Hull Jared
Принадлежит:

The present invention is directed to processes and systems for high availability Voice Over Internet Protocol telephony. Exemplary embodiments comprise a VOIP proxy cluster in communication with an IP private branch exchange cluster in communication with a PSTN VOIP gateway cluster, all at a first call cluster locale. In exemplary configuration, nodes within each of the clusters mount a database node from the database cluster, and store and retrieve persistent telephony operation data within the database cluster. Embodiments employ the clustering mechanisms to separate telephony elements in a cluster to cluster topology. In exemplary usage, multiple physically separate call cluster locales are deployed for higher availability. 1. A system for voice over internet protocol (VOIP) to public switched telephone network (PSTN) at a call locale , said system comprising:a VOIP proxy cluster in communication with a IP private branch exchange (PBX) cluster in communication with a PSTN VOIP gateway cluster;a database cluster, said database cluster comprised of a plurality of database nodes operable to store and retrieve state of telephony operations, said database nodes in communication with each other over an applied cluster mechanism;said VOIP proxy cluster comprised of a plurality of VOIP proxy nodes, each VOIP proxy node being an instance operable to proxy telephony from VOIP telephones, said VOIP proxy nodes in communication with each other and a clustering mechanism applied to said VOIP proxy cluster, replicating the data in each of said VOIP proxy nodes across said VOIP proxy nodes in said VOIP proxy cluster;said PBX cluster comprised of a plurality of IP PBX nodes, each PBX node being an instance operable to act as central switching system for telephony within said call locale, said PBX nodes in communication with each other and a clustering mechanism applied to said PBX node cluster, replicating the data in each of said PBX nodes across said PBX nodes in said PBX ...

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04-03-2021 дата публикации

Service System Supporting Voice Call Using Digital Assistant Device, Method Thereof, And Non-Transitory Computer Readable Medium Having Computer Program Recorded Thereon

Номер: US20210067633A1
Автор: KIM DongKeun, Kim Namhoon
Принадлежит:

The present invention relates to a service system supporting a voice call using a digital assistant device, a method thereof, and a non-transitory computer readable medium having a computer program recorded thereon, and more particularly, to a service system which performs a call between a digital assistant device and a terminal corresponding to a call opponent through a voice command of a user and supports call conversion between the digital assistant device and a user terminal through the voice command of the user, a method thereof, and a non-transitory computer readable medium having a computer program recorded thereon. 1. A service method supporting a voice call using a digital assistant device , the service method comprising:a checking step of checking, by a service server, a digital assistant device associated with a receiving number and receiver information included in a call processing request signal transmitted from a communication company server;a call connecting step of attempting, by the service server, a call connection to the digital assistant device checked based on the call processing request signal while attempting a call connection to a receiving terminal corresponding the receiving number checked based on the call processing request signal through the communication company server;a response step of requesting, by the service server, a call connection between a transmitting terminal and the digital assistant device to the communication company server based on a response signal received from the digital assistant device which recognizes a voice command for a call connection request of the receiver in response to the call connection attempt; anda call connection step of maintaining, by the service server, a call channel between the digital assistant device and the transmitting terminal formed by the communication company server in response to the requested call connection.2. The service method of claim 1 , further comprising:before the checking step, ...

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12-03-2015 дата публикации

Method and system for on-hold messaging for off network calls

Номер: US20150071124A1
Автор: Jaya MEGHANI, Tzahi Efrati
Принадлежит: VONAGE NETWORK LLC

An Off Net scenario may occur during primary communications between a first and second caller, in which the second caller is connected to a network over a data channel and receives a second call over a voice channel resulting in an interruption of the primary communications between the original callers. In the Off Net scenario, a message indicating the original communication has been interrupted is received by a carrier. The carrier substitutes another communication device in place of the second caller's communication device to establish a secondary communication between the first caller's communication device and the third communication device. The secondary communication may inform the first caller of the interruption of the primary communication and may provide options regarding the primary communication to the first caller.

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17-03-2022 дата публикации

DYNAMIC COMMUNICATION MANAGEMENT SYSTEM

Номер: US20220086283A1
Автор: Collard Michael H.
Принадлежит:

A dynamic communication management system manages the exchange of phone-based communications between devices using dynamically determined routing data. The routing for the phone-based communications can be dynamically determined such that certain identifying information is provided to destination devices instead of the identifying information that would be provided without the services of the dynamic communication management system. 120-. (canceled)21. A computer-implemented method comprising: receiving a request to transmit a phone-based communication to a destination device, wherein the destination device is associated with a first phone number, wherein the request comprises a selection method identifier, and wherein the selection method identifier identifies a selection method of a plurality of selection methods available to be used to select routing data;', 'identifying a pool of routing data associated with the request, wherein the pool of routing data comprises a plurality of phone numbers;', 'determining, based at least partly on the selection method identifier, the selection method from the plurality of selection methods available to be used to select routing data;', 'analyzing the first phone number to determine a property of the first phone number;', 'determining that the plurality of phone numbers does not include a phone number associated with the property of the first phone number;', 'determining, for individual phone numbers of the plurality of phone numbers, corresponding geographic regions of a plurality of geographic regions;', 'selecting a second phone number from the plurality of phone numbers based at least partly on a geographic region associated with the second phone number being geographically closest, out of the plurality of geographic regions, to a geographic region associated with the first phone number; and', 'sending the phone-based communication to the destination device using the second phone number, wherein sending the phone-based ...

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17-03-2022 дата публикации

Method for processing voice messages, method for deactivating dtmf coding and method for processing a request to deactivate dtmf coding

Номер: US20220086284A1
Автор: Bertrand Bouvet
Принадлежит: ORANGE SA

Methods are described for processing voice messages, for deactivating DTMF coding, and for processing a request to deactivate DTMF coding. The method for processing voice messages using a terminal includes sending a request to a voice server to be interpreted by the server to deactivate DTMF coding on a communication channel between the server and the terminal, receiving, from the server, a datum relating to the terminal's configuration, interpreting the configuration datum and applying to the terminal a configuration mode obtained from the interpretation, and processing voice messages using the terminal according to the configuration mode.

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10-03-2016 дата публикации

Device and method for the recognition of call numbers for voice-over-ip telephony

Номер: US20160072760A1
Автор: Heinrich Diebel

Call numbers are recognized in order to establish a connection from a lie-switched network to a packet-switched network. In one aspect, a device comprises a unit for detecting a selected string of digits as a selected call number, a unit for storing a plurality of authorized call numbers, a comparator unit for comparing the selected all number to the plurality of stored call numbers, and a unit for converting the selected call number into an associated IP address as soon as the comparator unit detest that the selected call number matches one of the stored all numbers.

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08-03-2018 дата публикации

System and method for call termination based on one or more over-the-top (OTT) call services

Номер: US20180069729A1
Автор: Avanish Pande
Принадлежит: Tata Communications America Inc

A system and method for connecting a call from an originating endpoint of a calling party to a terminating endpoint of a called party while accounting for one or more over-the-top (OTT) call services that are subscribed to by the called party. In response to receiving a dialed number, or other call information, a computer system determines whether to connect the call to the terminating endpoint of the called party via an OTT call service, based on the geolocation of the terminating endpoint. For example and without limitation, based on the terminating endpoint currently being located within a country that is on a list of developed countries or not currently being located within a country that is on a list of developing countries, the computer system might deem a particular OTT call service to be acceptable for handling the call for the terminating endpoint.

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10-03-2016 дата публикации

METHOD AND SYSTEM FOR IP COMMUNICATION COMPLETION VIA A WIRELESS NETWORK

Номер: US20160072959A1
Принадлежит:

Methods and systems for internet protocol (IP) communication completion via a wireless network include receiving a request from a first device to setup an IP telephony communications session with a second device, generating, based on user device data retrieved for the second device, a first message based on a first communication protocol including telecommunication invitation data to complete the IP telephony communications session, sending the first message to the second device, receiving a response message based on a second communication protocol from the second device, and connecting the second device with the first device to establish the IP telephony communications session. 1. A method for Internet Protocol (IP) communication completion via a wireless network , comprising:receiving a request from a first device to setup an IP telephony communications session with a second device;generating, based on user device data retrieved for the second device, a first message based on a first communication protocol including telecommunication invitation data to complete the IP telephony communications session;sending the first message to the second device;receiving a response message based on a second communication protocol from the second device; andconnecting the second device with the first device to establish the IP telephony communications session.2. The method of claim 1 , wherein the first message is a Short Message Service (SMS) message and wherein the first communication protocol is SMS.3. The method of claim 2 , wherein the SMS message is sent to the second device via a wireless gateway coupled to a wireless network.4. The method of claim 1 , wherein the telecommunication invitation data includes information included in an Session Initiation Protocol (SIP) INVITE message.5. The method of claim 1 , wherein the user device data includes at least one of a subscriber identity or an integrated circuit card identifier (ICCID) associated with the second device.6. The ...

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08-03-2018 дата публикации

METHODS AND APPARATUS FOR PROVIDING VOICE MAIL SERVICES

Номер: US20180070139A1
Принадлежит:

Methods and apparatus for retrieving and providing voice mail messages from a server are described. In accordance with the invention voice mail messages may be retrieved via requests made via a set top box. Voice mail messages are retrieved in response to the request from a voice mail server which is also accessible via the telephone network. Retrieved voice mail is transcoded and included in a video on demand (VOD) file. Text, call ID information and/or other information as, e.g., an automatically generated transcript of the voice mail message, may be included in the VOD file. The VOD file is supplied to a VOD server which provides the file including the message to the set top box for display. A user can view the messages and switch from viewing one message to the next by using video play back commands. 120-. (canceled)21. A method of operating a device , the method comprising:receiving voice mail content from a voice mail server; 'including a customer premise device executable application along with said voice mail content in said VOD file; and', 'generating a video on demand (VOD) file including the received voice mail content, said step of generating a VOD file includingcommunicating said VOD file including said customer premise device executable application and voice mail content to another device.22. The method of claim 21 , wherein said customer premise device executable application is an application which when executed by a processor controls the customer premise device to provide a user with voice mail functionality which was otherwise not available.23. The method of claim 22 , wherein said customer premise device executable application is an application that allows the customer premise device to perform at least one of: play said voice mail content or display information communicated in the VOD file.24. The method of claim 21 , wherein said voice mail content is received in a first format claim 21 , the method further comprising:transcoding the voice mail ...

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28-02-2019 дата публикации

DOOR ENTRY SYSTEMS AND METHODS

Номер: US20190068792A1
Принадлежит:

Various implementations include systems and methods for unlocking a door to a building on behalf of a tenant of the building that addresses the shortcomings of legacy telephone entry systems and allows for more efficient and safer management of building entry requests from third parties. In some implementations, the system taps into the legacy telephone entry systems to communicate with third parties seeking entry into the building. In some implementations, the system communicates with third party computing devices (e.g., stationary or mobile devices) to process entry requests. And, in some implementations, the system communicates entry preferences (or parameters) associated with a tenant with a door lock system to allow for entry during certain time windows or under certain conditions. 1102-. (canceled)103. A door entry system comprising a processor in communication with a memory and an intercom , wherein the processor executes computer-readable instructions stored on the memory , said instructions cause the processor to:receive a signal from a mobile computing device, the signal identifying a guest associated with the mobile computing device;receive a request to enter a building from the intercom;in response to receiving the request to enter the building, compare the identified guest with a list of approved guests associated with the building; andin response to the identified guest matching one of the approved guests, cause a lock on the door of the building to move into an unlocked position, wherein the lock and the door are associated with the intercom,wherein causing the lock on the door to move into the unlocked position comprises generating and communicating a telephone signal to the intercom.104. The door entry system of claim 103 , wherein the signal is a first signal claim 103 , and the instructions further cause the processor to receive a second signal from the mobile computing device claim 103 , the second signal identifying a location of the mobile ...

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09-03-2017 дата публикации

CORDLESS TELEPHONE EQUIPMENT, CORDLESS TELEPHONE SYSTEM, AND CORDLESS TELEPHONE COMMUNICATION METHOD

Номер: US20170070604A1
Принадлежит:

Cordless telephone equipment includes a portable unit and a base unit. The base unit includes: a portable unit control section controlling a communication protocol with the portable unit through a wireless interface wirelessly connecting with the non-IP terminal type portable unit; a wireless LAN interface section transmitting and receiving packet data to and from a wireless IP terminal and wireless LAN access point; a wireless LAN bridge control section forwarding the received packet data to the wireless IP terminal, or the wireless LAN access point, or within the base unit in accordance with a destination address of the packet data received through the wireless LAN interface section; and a base unit control section processing the packet data received from the wireless IP terminal and forwarded within the base unit, as data received from a portable unit under control of the base unit. 1. Cordless telephone equipment comprising:a handset that is a non IP terminal; anda base unit that connects with the handset by radio, wherein a handset controller that controls a communication protocol between the base unit and the handset through a wireless interface that connects with the handset by radio;', 'a wireless LAN interface that transmits and receives packet data to and from a wireless IP terminal;', 'a base unit controller that processes a message received from the handset controller and executes a call control, and receives, from the wireless IP terminal through the wireless LAN interface, packet data having address information of the base unit as a destination address, wherein, 'the base unit comprises an IP telephone controller that includes an IP call control server and that performs, when the packet data received from the wireless IP terminal is an IP telephone message, processing to convert the IP telephone message to an internal message format to be processed within the base unit controller and processing to convert an internal message within the base unit ...

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19-03-2015 дата публикации

Systems and Methods of Conducting Conference Calls

Номер: US20150078209A1
Автор: ARORA NEHAR
Принадлежит:

Systems and methods performed by an IP telephony system allow a user to request that his IP telephony device be set into a conference calling mode for an indefinite period of time. When the user's IP telephony device is in the conference calling mode, all parties that attempt to call the user's IP telephony device are added to a conference bridge tied to the user's IP telephony device. All incoming calls are treated in this fashion until the user cancels the conference calling mode and returns to a normal calling mode. 1. A method of managing telephony communications for an IP telephony device , the method being performed by at least an IP telephony system , the method comprising:setting an account of an IP telephony device into a first calling mode; andhandling an incoming telephony communication setup request directed to the account by directing each telephony communication setup request as a request to display a first call behavior in association with a first calling mode with the IP telephony device until the first calling mode is changed.2. The method of claim 1 , wherein each time that a telephony communication setup request directed to the account is received claim 1 , said handling comprises:handling the incoming telephony communication setup request as a request to join a first call mode with the IP telephony device if the account is set into the first calling mode; andhandling the incoming telephony communication setup request as a second call mode communication setup request if the account is set into a second calling mode.3. The method of claim 2 , wherein handling the incoming telephony communication setup request as a request to join the first call mode comprises:determining whether the IP telephony device is currently conducting a telephony communication;setting up a new telephony communication with the IP telephony device if the IP telephony device is not currently conducting a telephony communication; andplacing the party that sent the setup request ...

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19-03-2015 дата публикации

Computer, Internet and Telecommunications Based Network

Номер: US20150079955A1
Автор: Alex Kurganov
Принадлежит: Parus Holdings Inc

A method and apparatus for a computer and telecommunication network which can receive, send and manage information from or to a subscriber of the network, based on the subscriber's configuration. The network is made up of at least one cluster containing voice servers which allow for telephony, speech recognition, text-to-speech and conferencing functions, and is accessible by the subscriber through standard telephone connections or through internet connections. The network also utilizes a database and file server allowing the subscriber to maintain and manage certain contact lists and administrative information. A web server is also connected to the cluster thereby allowing access to all functions through internet connections.

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17-03-2016 дата публикации

Routing of Sessions to Other Communication Networks

Номер: US20160080429A1
Принадлежит: Telefonaktiebolaget LM Ericsson AB

System, methods, nodes, and instruction set for routing a session invitation from a first network ( 101 ) to a second network ( 112 ) are described. The first network ( 101 ) and the second network ( 112 ) are interconnected via at least two points of interconnect ( 1 14, 120 ). A first control node ( 124 ) receives a session invitation to a first user equipment ( 106 ), both are part of the first network ( 101 ). The first control node ( 124 ) determines whether a breakout condition for routing of the session invitation to the second network ( 112 ) is fulfilled. If so, a second control node ( 128 ) of the first network ( 101 ) selects a point of interconnect ( 114, 120 ) to the second network ( 112 ), considering a capability information characterizing the first control node ( 124 ). Therefore the session invitation may be routed in an efficient way from the first network ( 101 ) to the second network ( 112 ).

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15-03-2018 дата публикации

Verifying an application identifier on a mobile device through a telecommunication network

Номер: US20180077155A1
Автор: Elad Pinhas BARKAN
Принадлежит: Vokee Applications Ltd

A system for verifying an association between an application and a mobile computing device. The application is identified by an application instance identifier, which uniquely identifies the instance of the application on the mobile computing device. The system receives a request to verify the application, including the application instance identifier. The system also obtains a mobile device identifier used to communicate with the mobile computing device. The system initiates one or more telephone call setup messages, which signal a voice call request, to the mobile computing device using the mobile device identifier. Based on receiving an indication that the one or more telephone call setup messages were detected at the mobile computing device, the system associates the application instance identifier with the mobile device identifier.

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17-03-2016 дата публикации

Methods and apparatus to dynamically select a peered voice over internet protocol (voip) border element

Номер: US20160080576A1
Принадлежит: AT&T INTELLECTUAL PROPERTY I LP

Methods and apparatus to select a dynamically peered voice over Internet protocol (VoIP) border element are disclosed. An example method comprises collecting data representative of a dynamic performance of a voice over Internet protocol network, prioritizing a selection of a peered border element based on the collected data, and modifying a telephone number mapping (ENUM) database based on the prioritized selection.

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15-03-2018 дата публикации

SYSTEMS AND METHODS FOR CONDUCTING CONFERENCE CALLS

Номер: US20180077289A1
Автор: ARORA NEHAR
Принадлежит:

Systems and methods performed by an IP telephony system allow a user to request that his IP telephony device be set into a conference calling mode for an indefinite period of time. When the user's IP telephony device is in the conference calling mode, all parties that attempt to call the user's IP telephony device are added to a conference bridge tied to the user's IP telephony device. All incoming calls are treated in this fashion until the user cancels the conference calling mode and returns to a normal calling mode. 1. A method of managing telephony communications for an IP telephony device , the method being performed by at least an IP telephony system , the method comprising:sending invitations to at least one invitee of a future communication;setting an account of an the IP telephony device into a first calling mode; andhandling an incoming telephony communication setup request directed to the account by directing each telephony communication setup request as a request to conduct a call, in association with the first calling mode, with the IP telephony device until the first calling mode is changed.2. The method of claim 1 , wherein the invitations are sent to the at least one invitee by a means selected from the group consisting of a voice message claim 1 , an email claim 1 , an SMS claim 1 , an MMS claim 1 , contacting the invitee directly and an interactive voice response application.3. The method of claim 1 , wherein the delivery of the invitations are scheduled in advance of the future communication.4. The method of claim 1 , wherein the at least one invitee is associated with an access code.5. The method of claim 4 , wherein the at least one invitee uses the access code to join the future communication.6. The method of claim 4 , wherein information regarding the access code is included in the invitation sent to the at least one invitee.7. The method of claim 1 , further comprising establishing a list of contacts associated with the future communication.8 ...

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24-03-2022 дата публикации

Enabling Internet Protocol Carrier Peering to Foreign Network Domains

Номер: US20220094660A1
Принадлежит: AT&T Intellectual Property I, L.P.

Concepts and technologies pertaining to enabling internet protocol carrier peering to foreign network domains are provided. A method includes identifying, by a computer system executing within an originating carrier network, a plurality of numbering plan area identifiers corresponding to a receiving carrier network. The method further includes accessing, by the computer system executing within the originating carrier network based on the plurality of numbering plan area identifiers, a plurality of numbering plan area zone file records stored on a private enabled telephone number mapping server. The method also includes creating, by the computer system executing within the originating carrier network, a single instance of a name authority pointer record placeholder within each of the plurality of numbering plan area zone file records stored on the private enabled telephone number mapping server. 1. A method comprising:identifying, by a computer system executing within an originating carrier network, a plurality of numbering plan area identifiers corresponding to a receiving carrier network, wherein the originating carrier network is associated with a domestic carrier and the receiving carrier network is associated with a foreign carrier;accessing, by the computer system, a plurality of numbering plan area zone file records corresponding to the plurality of numbering plan area identifiers, wherein the plurality of numbering plan area zone file records are stored on a private enabled telephone number mapping server, and wherein at least a portion of the plurality of numbering plan area zone file records comprises a plurality of name authority pointer records;creating, by the computer system, a single instance of a name authority pointer record placeholder within each of the plurality of numbering plan area zone file records stored on the private enabled telephone number mapping server; andreplacing, by the computer system, the plurality of name authority pointer ...

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12-06-2014 дата публикации

Method and Arrangement for Making a Call-Setup

Номер: US20140160992A1
Принадлежит: TELEFONAKTIEBOLAGET L M ERICSSON (PUBL)

A method and arrangement for providing a ring-back presentation to a calling terminal after receiving a call setup request from the calling terminal for a first communication session with a called terminal. A second packet-based communication session is established with the calling terminal, independent of the first session, and pre-defined media content is provided as said ring-back presentation 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. 1. A method of 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 with a called terminal (B) , comprising the following steps:executing a call-setup for the first communication session,establishing a packet-based second communication session with the calling terminal, independent of the call-setup for the first session, by the called terminal sending a session initiating message directed to the calling terminal, andproviding pre-defined media content representing said ring-back presentation to the calling terminal, by means of said second session, wherein said pre-defined media content has been stored in a server and is retrieved therefrom for delivery to the calling terminal by means of the second session;wherein the called party provides the calling party with a server address and a suitable reference to said media content.2. (canceled)3. (canceled)4. (canceled)5. (canceled)6. The method according to claim 1 , wherein said server is an RTSP server and the media content is downloaded directly therefrom to the calling terminal.7. The method according to claim 6 , wherein the network address of the RTSP server and a suitable reference to said media content is sent to the ...

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05-03-2020 дата публикации

SYSTEM AND METHOD FOR DELIVERY OF VOICEMAILS TO HANDHELD DEVICES

Номер: US20200076950A1
Принадлежит:

Briefly, a variety of embodiments, including the following, are described: a system embodiment and methods that allow random access to voice messages, in contrast to sequential access in existing system embodiments; a system embodiment and methods that allow for the optional use of voice recognition to enhance usability; and a system embodiment and methods that apply to the area of voicemail. 1. A system comprising:a server adapted to record a received voicemail message in an audio format, convert the recorded audio formatted message to a text representation of the message, and communicate the audio formatted message and text representation of the message to an output device.2. The system of claim 1 , wherein the audio format comprises at least one of a WAV audio file claim 1 , or an MP3 audio file.3. The system of claim 1 , wherein the output device comprises at least one of a mobile phone or interface device.4. The system of claim 1 , wherein the output device comprises a mobile phone with a message detection module implemented in code native to the mobile phone.5. (canceled)6. A method of operating a server comprising:capturing a received voicemail message in an audio format;converting the audio formatted message to a text representation of the message; andinitiating communication of the text representation of the message and the audio formatted message to an output device from the server.7. The method of claim 6 , and further comprising: receiving claim 6 , at the output device for display on the output device as a voicemail message claim 6 , the text representation of the message claim 6 , as well as receiving the audio formatted message of the received voicemail message.8. The method of claim 7 , and further comprising: processing for display on the output device as a voicemail message claim 7 , the text representation of the message; and storing the audio formatted message.9. The method of claim 6 , and further comprising: retrieving the text representation ...

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05-03-2020 дата публикации

TELEPHONY WEB EVENT SYSTEM AND METHOD

Номер: US20200076952A1
Принадлежит:

An embodiment of the system for publishing events of a telephony application to a client includes a call router that generates events from the telephony application and an event router that manages the publication of events generated by the call router and that manages the subscription to events by clients. The system can be used with a telephony application that interfaces with a telephony device and an application server 1. A method comprising:receiving at a telephony system, an application instruction of an application server external to the telephony system, the application instruction received using an application programming interface (API) via a network;in response to the application instruction, performing a telephony action,publishing an event corresponding to the telephony action to an event router of the telephony system; andsending the published event from the event router to a subscriber.2. The method of claim 1 , wherein:the application instruction is a telephony instruction;the published event is a telephony event that corresponds to the telephony instruction;the telephony instruction is an instruction of an account of a plurality of accounts of the telephony system; andthe telephony event is an event of the account.3. The method of claim 1 , further comprising:before receiving the application instruction, establishing a subscription by the subscriber for events associated with a type of the published event.4. The method of claim 1 , further comprising:sending the published event to the application server.5. The method of claim 1 , wherein the application instruction is an instruction to dial a phone number.6. The method of claim 1 , wherein the telephony action comprises dialing a numb7. The method of claim 1 , wherein the telephony action comprises starting a Text-To-Speech conversion.8. The method of claim 1 , wherein the telephony action comprises playing an audio file.9. A telephony system comprising:a memory that stores instructions; and ...

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05-03-2020 дата публикации

TELEPHONE SIGNAL PROCESSING

Номер: US20200076953A1
Автор: Baldwin Thomas, Tao Yufei
Принадлежит:

A method of processing a telephone signal comprising voice signals and data signals, the method comprising detecting the presence of an artefact in the telephone signal indicative of the presence of a data signal fragment associated with an earlier attenuation of a data signal and processing the telephone signal by further attenuating the telephone signal in the region of the artefact in order to remove the data signal fragment from the telephone signal. 1. A method of processing a telephone signal comprising voice signals and data signals , the method comprising:detecting the presence of an artefact in the telephone signal indicative of the presence of a data signal fragment associated with an earlier attenuation of a data signal; andprocessing the telephone signal by further attenuating the telephone signal in the region of the artefact in order to remove the data signal fragment from the telephone signal.2. A method according to claim 1 , wherein the data signal comprises at least one of:a) an acoustic signal,b) acoustic signal according to an acoustic data transmission protocol, andc) a DTMF tone.3. A method according to or claim 1 , wherein attenuating the telephone signal in the region of the artefact comprises at least one of:a) omitting or dropping or deleting a portion of the telephone signal,b) replacing a portion of the telephone signal, and/orc) modifying a portion of the telephone signal.4. A method according to any preceding claim claim 1 , further comprising further attenuating the telephone signal only when data signal fragments are expected to be present.5. A method according to any preceding claim claim 1 , wherein processing of the telephone signal occurs in the time domain.6. A method according to any preceding claim claim 1 , wherein the artefact comprises a spike in the telephone signal claim 1 , defined by the ratio of the maximum or peak amplitude of the telephone signal to the noise floor exceeding a threshold.7. A method according to claim ...

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22-03-2018 дата публикации

METHOD FOR HOME GATEWAY TO REALIZE IVR SERVICE AND HOME GATEWAY

Номер: US20180084114A1
Принадлежит:

A method for a home gateway to realize an Interactive Voice Response (IVR) service and a home gateway are provided. The method includes that: when a voice user belonging to a home gateway serves as a called user, the home gateway provides an IVR prompt to a calling user, so that the calling user makes an IVR service request according to the IVR prompt; and the home gateway receives the IVR service request, and performs an IVR service operation corresponding to the IVR service request. The present disclosure satisfies multi-level service requirements of a home user. 1. A method for a home gateway to realize an Interactive Voice Response (IVR) service , comprising:when a voice user belonging to a home gateway serves as a called user, providing, by the home gateway, an IVR prompt to a calling user, so that the calling user makes an IVR service request according to the IVR prompt; andreceiving, by the home gateway, the IVR service request, and performing an IVR service operation corresponding to the IVR service request.2. The method as claimed in claim 1 , wherein providing claim 1 , by the home gateway claim 1 , the IVR prompt to the calling user comprises:feeding back, by the home gateway, the IVR prompt to the calling user while playing a predefined voice file to the calling user, wherein the IVR prompt is that: different dial key tones correspond to different IVR service requests; the IVR service requests comprise: making a home phone ring, receiving a fax, or receiving a message.3. The method as claimed in claim 2 , wherein when the IVR service request is making the home phone ring claim 2 , performing claim 2 , by the home gateway claim 2 , the IVR service operation corresponding to the IVR service request comprises:making, by the home gateway, the home phone ring, and if the voice user does not answer, sending information of the missed call to a default mailbox/default short message number preset by the voice user through a mail/short message, so as to prompt the ...

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23-03-2017 дата публикации

ADJUSTABLE DUAL-TONE MULTI-FREQUENCY PHONE SYSTEM

Номер: US20170085713A1
Автор: Casasola Jose
Принадлежит:

Provided herein are methodologies, systems, apparatus, and non-transitory computer-readable media for providing an adjustable dual-tone multi-frequency (DTMF) phone system. The DTMF phone system includes a DTMF adjustment module that interfaces with a base DTMF phone system and retrieves call parameter values from at least one properties file. The call parameter value is indicative of an audio file to be played by the DTMF phone system, or a DTMF call flow option file. The DTMF adjustment module provides the call parameter values to the base DTMF phone system. The base DTMF phone system includes a number of variables, and a DTMF call flow is generated by assigning at least one of the call parameter values to one or more of the variables within the base DTMF phone system. 1. An adjustable dual-tone multi-frequency (DTMF) phone system , comprising: provide a DTMF adjustment module located at the one or more servers, the DTMF adjustment module separate from a base DTMF phone system;', 'retrieve at least one call parameter value using the DTMF adjustment module from a properties file located at the one or more servers, the at least one call parameter value indicative of at least one of: an audio file to be played by the adjustable DTMF phone system, or a DTMF call flow option file;', 'provide the at least one call parameter value to the base DTMF phone system, via the DTMF adjustment module, the base DTMF phone system including a plurality of variables; and', 'generate a DTMF call flow by assigning the at least one call parameter value to at least one of the plurality of variables within the base DTMF phone system., 'one or more servers programmed to2. The system of claim 1 , wherein the audio file to be played includes at least one of: an exit announcement claim 1 , an error announcement claim 1 , a clarifying question claim 1 , a confirmation announcement claim 1 , or a prompt for additional caller input.3. The system of claim 1 , wherein the DTMF call flow option ...

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12-03-2020 дата публикации

COMMAND AND CONTROL OF DEVICES AND APPLICATIONS BY VOICE USING A COMMUNICATION BASE SYSTEM

Номер: US20200082826A1
Принадлежит:

A first communication path for receiving a communication is established. The communication includes speech, which is processed. A speech pattern is identified as including a voice-command. A portion of the speech pattern is determined as including the voice-command. That portion of the speech pattern is separated from the speech pattern and compared with a second speech pattern. If the two speech patterns match or resemble each other, the portion of the speech pattern is accepted as the voice-command. An operation corresponding to the voice-command is determined and performed. The operation may perform an operation on a remote device, forward the voice-command to a remote device, or notify a user. The operation may create a second communication path that may allow a headset to join in a communication between another headset and a communication device, several headsets to communicate with each other, or a headset to communicate with several communication devices. 1. A method for controlling a device using voice-commands , the method comprising:receiving a communication at a base station;identifying, with the base station, a speech pattern as including a voice-command by the detection of an address word, wherein the address word is associated with the base station and is unassociated with the voice-command;determining a second device operation corresponding to the voice-command;communicating the second device operation from the base station to the second device; andexecuting the second device operation at the second device.2. The method of claim 1 , further comprising separating the voice command portion of the speech pattern from the address word portion of the speech pattern.3. The method of wherein the address word is a phrase in a non-English language as predetermined by a user.4. The method of wherein the address word is a user-specified noun.5. The method of claim 1 , wherein the second device operation corresponding to the voice command of the user differs from ...

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02-04-2015 дата публикации

PROXY MEDIA SERVICE FOR DIGITAL TELEPHONY

Номер: US20150092571A1
Принадлежит:

A Session Initiation Protocol (SIP) service system includes a SIP-enabled soft switch at a telephony service provider, executing code from a coupled machine-readable medium, routing SIP transactions to remote destinations, a media server coupled to the SIP-enabled soft switch storing media including ring tones and music-on-hold for use in progressing transactions, and an interface to a wide-area-network (WAN) for transmitting transactions and media. 1. A Session Initiation Protocol (SIP) service system , comprising:a SIP-enabled soft switch at a telephony service provider, executing code from a coupled machine-readable medium, routing SIP transactions to remote destinations;a media server coupled to the SIP-enabled soft switch, the media server having local access to stored ring tones and music-on-hold for use in progressing transactions; andan interface to a wide-area-network (WAN) for transmitting transactions and media;wherein the SIP-enabled soft switch determines for each transaction from stored data whether media services are to be provided or not provided for that destination, and in the event media services are not to be provided, alters packet data to indicate media services to be provided by a server local to the destination.2. The service system of wherein the WAN is the Internet network.3. The service system of wherein the destination further comprises a plurality of SIP-enabled communication appliances coupled on a local-area-network (LAN) and a telephony border controller (TBC) including a SIP-PBX coupled to a local media server also supported on the LAN claim 1 , and wherein transactions for the destination are routed to the SIP-PBX claim 1 , and media services are provided from the local media server according to SIP state of transaction packets as monitored by the SIP-PBX.4. The service system of wherein the TBC is configurable claim 3 , enabling users of individual ones of the SIP-enabled communication appliances to designate specific media for use ...

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12-03-2020 дата публикации

USER-BASED DIFFERENTIATED ROUTING SYSTEM AND METHOD

Номер: US20200084140A1
Принадлежит: LEVEL 3 COMMUNICATIONS, LLC

A differentiated routing system includes an electronic service in communication with an ingress gateway that receives a communication service, such as a call from a terminal, over a trunk. The service receives a request from the ingress gateway for establishing a communication service for the calling party terminal. The service obtains information associated with the calling party terminal, which may be based on the trunk and its relationship to a customer, in which the information is to be used for applying at least one of a routing decision and policy decision to the communication service, and appends a tag to the request based on the information. The service may then transmit the request appended with the information to a routing device, which may then use the information when providing the service. 1. A differentiated routing system comprising:a routing device in communication a computing system in a native network domain, the native network domain comprising a plurality of egress gateways that route communication services to a corresponding plurality of destination network domains, the routing device comprising a processor and a memory to store instructions that are executed by the processor to:receive a request from the computing system appended with a tag, wherein the tag includes at least one of a country designation portion, a quality of service portion, a class of service portion, and a service type portion;select a destination network domain of the plurality of destination network domains according to the at least one of a country designation portion, a quality of service portion, a class of service portion, and a service type portion included in the tag; andtransmit an instruction to an ingress gateway in the native network domain in communication with the computing system to establish a communication service with the selected destination network domain.2. The differentiated routing system of claim 1 , wherein the tag further includes geographical ...

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