МАШИННОЕ ОБУЧЕНИЕ

... 1. Способ, содержащий этапы, на которых: принимают набор сеансов общения между членами стороны первого типа и членами стороны второго типа, причем каждый из этих сеансов общения включает в себя сообщение члена стороны первого типа и сообщение члена стороны второго типа, которое является ответом на упомянутое сообщение члена стороны первого типа; группируют сообщения членов стороны первого типа в первый набор кластеров; группируют ответные сообщения членов стороны второго типа во второй набор кластеров, основываясь на группировании сообщений членов стороны первого типа; и с помощью машины генерируют набор классификаторов стороны второго типа для одного или более кластеров во втором наборе кластеров. 2. Способ по п.1, в котором сообщения содержат произнесения. 3. Способ по п.1, в котором сообщения содержат текстовые послания. 4. Способ по п.1, в котором сообщения членов стороны первого типа содержат сообщения являющихся людьми агентов по обслуживанию клиентов в центре обработки звонков. 5 ...

ТЕЛЕФОННАЯ СТАНЦИЯ ЧАСТНОГО ПОЛЬЗОВАНИЯ С СИСТЕМОЙ УПРАВЛЕНИЯ УСТРОЙСТВАМИ

... 1. Способ, включающий следующие шаги:принимают сообщение о регистрации от оконечной точки, поддерживающей голосовую связь по протоколу Интернет (VolP), причем сообщение о регистрации указывает на событие телефонии с участием оконечной точки, при этом событие телефонии представляет собой регистрацию оконечной точки на телефонной станции частного пользования, работающей по протоколу Интернет (IP PBX), отправку оконечной точкой вызова через IP-PBX или прием оконечной точкой вызова через IPPBX;на основе сообщения о регистрации обновляют состояние телефонной связи в центре состояний с указанием события телефонии с участием оконечной точки; ипередают уведомительное сообщение одному или более первым устройствам, зарегистрировавшимся для приема уведомительных сообщений о событиях телефонии от центра состояний, с целью информирования одного или более первых устройств о возникновении события телефонии с участием оконечной точки и обеспечения выполнения по меньшей мере одним первым устройством ответного ...

VERFAHREN ZUM ZUGRIFF AUF ECHTZEITDATEN IN EINEM AUTOMATISCHEN ANRUFVERTEILSYSTEM

Rufumleitungstechnik für ankommende Kommunikationen mit Ortsanzeige von angerufenem Teilnehmer

Verfahren und Gerät zum Adressieren von Anwendern in einem Netz

VERFAHREN UND VORRICHTUNG ZUR AUTOMATISCHEN ABLAUFSTEUERUNG VON TELEFONISCHEN ANRUFEN

Spontane Sprach- und Datenbenachrichtigung

Verfahren und Telekommunikationsnetz zur Steuerung von Massenanrufen

Gerät und Verfahren zur Verbesserung der Anrufleitweglenkung nach und innerhalb von Anrufzentralen

Route list regulation method for switching data network

The method uses route lists (LA,...LE) stored in memory in individual nodes (A,B,C,D,E) in a data network (N) to allow switching of connections. The nodes are coupled together by means of transmission connections (1,...10). The network is controlled by a central network computer (ZR). A determination of routes to be included in the lists is carried out repeatedly until two or more halting criteria are reached. The network computer (ZR) has a controller (PR) which carries out the iterative determination. The controller (PR) can control a data carrier (DT) on which data is stored, and interacts with the nodes using connections (D1-D5). The data includes information to enable the determination to be carried out.

Verfahren zum Errichten und Abrufen eines Telefonbenutzerprofils

A method of invoking a telephone user profile comprised of storing in a database 31 a user telephone service profile containing a directory number of a user, in association with a telephone set line, transmitting the user telephone service profile to a central computer 40, transmitting the service profile to a group of telephone office switches CO2 - CO5 from the computer, storing the service profile at each of the telephone office switches in the group, and providing service to the user at each of the telephone office switches in the group in accordance with the service profile. ...

Rückrufsystem und Verfahren für Internet telefon

Managing utilisation of interconnections between communications networks by routing local connections via interconnection to second network and back to first network

Data is transmitted between a calling and a called (9) telecommunications unit, firstly via the first communication network (3), which carries local communications, and then transferred to the second communication network (6). To achieve the desired utilisation of the interconnections, the telecommunications connections for local communications are routed from the first network via an interconnection (5) to the second network, and from this network back to the first network, or further to a third communications network. The first communications network then switches the connection to the called telecommunications unit (9). An Independent claim is included for a communications system for managing the utilisation of interconnections.

Dial plan transparency for fragmented networks

Browser-based communications enhanced with enterprise communication features

Voice communication apparatus

A voice communication apparatus is interposed between a telephone set and a public switched telephone network (PSTN) or a broadband network which can provide a frequency bandwidth broader than a frequency bandwidth of the public switched telephone network. The apparatus selects one of the PSTN and the broadband network, from which a voice signal is input. The apparatus takes account of the selected network, frequency characteristics of the selected network and of the telephone set, and appropriately adjusts frequency characteristics of a voice signal transmitted between the telephone set and the selected network by means of transmitter and receiver band-pass filter units. As a result, both the telephone set and the counterpart communication terminal can receive a voice signal of better transmission quality.

Interworking device

Telephone call screening

A destination telecommunication switch receives a request to establish a call to a first party served by the switch. The request includes call type data that identifies the party originating the call request as one of a plurality of call categories. A first record associated with the first party is read from a database. The first record includes at least one set of data which defines a call category and a corresponding call processing technique to be utilized for the call category. The switch processes the incoming call in accordance with the predetermined call category handling if the incoming call is identified by call category data in the database.

Private branch exchange back-up system

The back-up device for an IP-PBX of the present invention is connected to each network. The private branch exchange is connected to an individual network to manage and hold the entire terminal device. The back-up device holds each terminal data of a group of terminal devices in a connected network. As a result, when connection management by the private branch exchange develops a failure, the back-up device secures connection management in the network to which the backup device is connected. When the network or the private branch exchange itself develops a failure to disable control of an accommodated IP terminal, this arrangement enables the IP terminal to resume operation by temporarily shifting control of the IP terminal to the back-up device side, thereby minimizing stop time.

VOIP hardware and software

A method of routing a call comprising the steps of: receiving an outbound call transmission; analysing the call transmission to determine a type of call endpoint for the call; and routing the call according to the call endpoint type. Also disclosed is a system for routing calls comprising: a first interface for communicating with a private or internal telephone system; a second interface for communicating with a Public telephone system; a third interface for communicating with the internet; a processor; and memory storing computer readable instructions that cause the processor to: receive an outgoing call from the private or internal telephone system through the first interface; analyse the call to determine a type of call endpoint for the call; and route the call to the second or third interface depending on the determined call endpoint type. The determined call endpoint type is: a VOIP call, a mobile application client, or a PSTN telephone number or any other endpoint type.

A method of connection

Method for call setup time improvement for 3g-234m

AVOIDANCE OF A FAILURE OF A SWITCHING EQUIPMENT CONTROL (MEDIA GATEWAY CONTROLLER) OR SWITCHING EQUIPMENT (MEDIA GATEWAY)

PROCEDURE FOR PASSING ON SIGNALING DATA IN A NET TRANSITION UNIT AND IN A CONTROL UNIT

IMPROVED PLANT AND FUNCTIONALITY FOR A CALL CENTER IN THE TELEPHONE NATURE

EQUIPMENT AND PROCEDURE FOR THE IMPROVEMENT THE CALL ALTERNATE ROUTING AFTER AND IN CALL CENTERS

CALL FORWARDING IN A TELECOMMUNICATIONS NET

PROCEDURE FOR THE PURPOSEFUL MONITORING OF TELEPHONE CONNECTIONS IN A SUPERORDINATE CONTROL CENTER

PROCEDURE AND EQUIPMENT FOR ADDRESSING USERS IN A NET

OUPLE NETWORK FOR COMMUNICATION DEVICES

AUTOMATED PERIPHERAL COMMUNICATIONS SYSTEM

MONITOR FOR THE MONITORING OF A TELECOMMUNICATIONS NETWORK CONTAINING ONE OR MORE PARTICIPANT SWITCHING PLANTS

PROCEDURE AND EQUIPMENT FOR CALLING SEVERAL TELEPHONING DEVICES WITH AN ARRIVING DISCUSSION

Mass calling in public switching network

Induced answering method and system for circuit switching-based telephony

Method and apparatus for directing a wireless communication to a wireline unit

System and method for announcing messages to phone users in telecommunicationsnetwork

Arrangement for queuing a call to the best split

Method and communication network for cross coding between codecs

Query response device

The invention concerns a query response device comprising: an input adapted to receive user queries; a memory (106) adapted to store one or more routing rules; one or more live agent engines (116) configured to support interactions with one or more live agents; one or more virtual assistant engines (120) configured to support interactions with one or more virtual assistants instantiated by an artificial intelligence module (103); and a routing module (104) coupled to said live agent engines and to said virtual assistant engines, the routing module comprising a processing device configured: to select, based on content of at least a first user message from a first user relating to a first user query and on said one or more routing rules, a first of said live agent engines or a first of said virtual assistant engines; and to route one or more further user messages relating to the first user query to the selected engine ...

Proximity detection

A method of detecting a proximity to a target device. The method comprises scanning a wireless network frequency band in a personal mobile device to discover at least one wireless network signal, triggering, in response to a discovery of the at least one wireless network signal, a magnetic field scan of a magnetometer operated by the personal mobile device, detecting, in the personal mobile device, according to the magnetic field scan, a presence or an absence of a magnetic field signature associated with a predefined proximity to a target device.

METHOD AN APPARATUS FOR AUTOMATIC TELEPHONE SCHEDULING SYSTEM

A METHOD OF SCHEDULING CONNECTIONS

CONTEXT BASED DATA MEDIATION

A communication environment includes of one or more subscriber terminals capable of receiving and transmitting data over a communication network via a communication management system The communication management system receives mobile communication device context information and mobile communication device identification information from the mobile communication device The communication management system then identifies data availability profiles reflective of a prior determination of mobile communication device availability to receive data communications according to context The communication management system then implements data filtering rules corresponding to a current data availability profile ...

CORDLESS TELEPHONE SYSTEM

A cordless telephone system includes a connector connected to a subscriber telephone line and at least one radio telephone set through a radio channel to establish or hold a communication channel between the radio telephone set and a subscriber. The connector includes a control unit incorporating a timer. The control unit sets the system in a self hold state in which the radio channel is kept connected or a complete hold state in which the radio channel is disconnected on the basis of a hold signal from the radio telephone set, the presence/absence of a radio reception signal within a predetermined period of time preset in the timer, and the type of reception signal.

METHOD FOR TRANSMITTING INFORMATION IN WIRELESS COMMUNICATION SYSTEM AND TERMINAL SUPPORTING THE METHOD

SYSTEMS AND METHODS OF VOICE CALL SETUP INVOLVING MOBILE DEVICES

A mobile device can send a notification to place a call using a PBX (e.g., using SIP signaling) or receive a notification of call availability for the mobile device at the PBX. An authentication token is provided over a data channel for use in authenticating the mobile device when calling the PBX on a voice channel. However, when the mobile device calls the PBX (e.g., on a DNIS number, where ANI information is made available to an application server), if ANI information is found to correspond to the mobile device, the PBX can send tone(s) indicating that the mobile device can respond with an abbreviated validation tone, rather than the entirety of the authentication token. If the ANI information is not obtained or not matched, then a different tone(s) is sent, indicating to provide the authentication token. If the authentication token provided does not match or if the mobile device does not respond, the voice channel is not set up.

CONGESTION TUNING OF TELECOMMUNICATIONS NETWORK

A system for optimizing the traffic carrying capacity of a telecommunications network having a plurality of elements and a plurality of routes connecting those elements to one another. The congestion in the network is controlled by identifying and limiting defective devices and routes with exaggerated levels of congestion. Disturbance alarm levels are also adjusted to allow increased traffic through the network and the traffic within the network is reconfigured in real time in response to the occurence of network events to optimize traffic capacity.

INTELLIGENT BASED SELECTIVE CALL FORWARDING

Telecommunications Follow Me Services

MULTI-SYSTEM NETWORK ADDRESSING

MULTI-SYSTEM NETWORK ADDRESSING Storage locations, such as voice mailboxes are addressed on a plurality of widely distributed host systems, connect via a network, by defining logical domains each of which may include storage locations on more than one host system. The domains are defined by address tables which indicate which host systems provide ranges of address locations. Transfer of data between domains is simplified by the use of translation tables. One type of translation table is an alias table which has entries containing a range in the originating domain, a new domain and a starting point for a range in the new domain, thus defining a one-to-one correspondence between the ranges in the two domains. Another translation table is a gateway table which defines a gateway to a new domain when an input address has characteristics matching a prefix, number of digits, or both, of an entry in the gateway table. Upon finding the closest possible match in the gateway table, the portion of ...

TELEPHONE LINE INTERFACE CIRCUIT WITH VOLTAGE CONTROL

A telephone line interface circuit comprises two unity gain d.c. amplifiers (132, 134) having outputs d.c. coupled via feed resistors to the line wires. Currents in the feed resistors are monitored via a digital control circuit (152), which can be programmed to provide desired operating conditions. The control circuit controls, via D-A converters (158, 160), currents passed by current sources (142, 148) through resistors (140, 146) at inputs of the d.c. amplifiers, thereby determining voltages at the amplifier inputs and hence on the line wires. These d.c. voltages are smoothed by capacitors (136, 138) used for coupling an a.c. signal to the d.c. amplifiers. The voltage control facilitates adaptive adjustment of the line interface circuit to the line in dependence upon the monitored currents.

SIGNAL PROCESSING SYSTEM FOR ACHIEVING PER-CALL CONTROL

A telecommunications network (10) includes an Originating Switching System (12 1) and a Terminating Switching System (12 2) linked by a plurality of tr unks(20 1-20 4). For an incoming call it receives, the O riginating Switching System establishes a Circuit Selection Capability Indicator (CSCI) that determines what enhanceme nt, if any, the Originating Switching System will provide to the call. The Originating S witching System also utilizes the CSCI to select an appropriate trunk that is compati ble with type of call enhancement, if any, to route the call to the Terminating Switching System which also enhances the call in accordance with the CSCI. The signal process ing system per-call-control (SPCC) concept allows different types of calls which are routed on the same trunk to get different call quality enhancements (such as True V oiceR and Echo Control), depending on the call type. SPCC allows a call quality enhanc ement to be applied to an individual call on a per call basis, and these ...

METHOD OF USING CARRIER INFORMATION FOR ENHANCED CALL DATA PROCESSING BY A TELECOMMUNICATIONS PROVIDER

Disclosed is a capability for differentiated in-network or off-network call treatment (e.g., different call features or services) on a per-call basis us ing the value of the carrier identification code (CIC) passed from an access provider to a second carrier in a call set-up message associated with an incoming call. In additi on, the caller selection information (CSI) parameter may be used in conjunction with theCIC to provide differentiated call treatment. The CI C and CSI parameters together provide information about the call and the caller which can be used for, e.g ., directed marketing activities aimed at non-presubscribed callers using the second car rier's network on a per-call basis.

Arrangement for Queuing a Call to the Best Split

METHOD AND APPARATUS FOR ON-DEMAND TELECONFERENCING

An on-demand teleconferencing system and method for setting up an on-demand conference call in a telecommunications system having the Advanced Information Network (AIN) architecture with system signaling the number 7 (SS7) and a Public Switched Telephone Network (PSTN). A subscriber is assigned an on-demand conference call number. When that number is dialed into the PSTN, it is identified by the PSTN that the dialed number requires handling by the SS7. The SS7 links the dialed number to a conference allocation and control system (CACS) which is connected to a plurality of scalable bridge servers. The CACS selects bridge servers available to handle the conference call and based upon a selection criteria such as a peak load sets up the on-demand conference call in one of the selected bridge servers.

ANONYMOUS VOICE COMMUNICATION USING ON-LINE CONTROLS

A system and method for providing anonymous voice communications using the telephone network under the direction of an on-line data network. A call broker receives call setup information from on-line session participants, su ch as chat room participants, after providing callback telephone numbers and, as appropriate, billing information. In one illustrative embodiment, a user (th e call initiator) present in a chat room session establishes a data connection to a telephone company site using, e.g., an Internet web browser to establish a Call Broker session. The Call Broker then provides session information (such as a telephone company URL) and a Participant Authorization Code (PAC), which information is passed to a selected chat participant in the current chat roo m. When the chosen participant uses the received URL to access the telephone company site, and enters the received participant code and a call-back number, the telepho ne company completes voice links to the initiator and the selected ...

SUBRATE VOICE SWITCHING OVER SWITCHING AND TELECOMMUNICATIONNETWORKS

Apparatus for switching full rate (e.g., 64 kilobit per second) signals and composite signals comprising a plurality of subrate signals (e.g., 32 kilobit per seco nd subrate signal). The apparatus includes a unit for compressing selected ones of a plurality o f full-rate signals into a smaller plurality of full-rate single and composite signals, and apparatus for decompressing composite signals into a plurality of full-rate signals. The main elements of the switching fabric of the switch continue to switch full-rate signals. Advantageously, during periods of heavy load, the switching network fabric of a switching system can carry more calls, and more traffic can be carried between switches equipped for subrate switching.

MULTI-PROTOCOL TELECOMMUNICATIONS ROUTING OPTIMIZATION

A telecommunications switching system (10) employing multi-protocol routing optimization which utilizes predetermined and measured parameters in accordance with a set of user priorities in determining the selection of a telecommunications path to be utilized for transmitting a data file to a remote destination. The switching system (10) has a first memory (30) for storing the data file, a second memory (22) for storing predetermined parameters, a third memory for storing a set of user priorities (32), means (24) for measuring the value of variable parameters associated with each of the telecommunications paths, and a processor means (26) are operatively associated with the second and third memories (22 and 23) and the variable parameters for determining which of the plurality of telecommunications path s should be utilized.

IMPROVED CALL CENTER APPARATUS AND FUNCTIONALITY IN TELEPHONY

A client-server telephone call router system as part of a customer premises system has a client-server router adapted to execute on a telephony switch, such as a public branch exchange (PBX) or other telephony switch, or on a processor connected by CTI link to a telephony switch. The telephony switch or processor executing the router is connected to a local area network (LAN) th at also interconnects computer workstations proximate to telephones connected t o the telephony switch. Client user interface applications run on the computer workstations, allowing clients to edit routing rules for the router, which h as a list of routing rules keyed to users and workstations of the customer premises system. The editing rules are kept by the router in portions dedicated to individual users. With this system a user can edit at a workstation on the LAN his/her own routing rules, and transmit the edits to the client-server router where the rules will be followed to route calls for that user and protocol ...

METHOD FOR ALLOCATING PHYSICAL MEANS STACK OF USERS, DEVICE, MONITORING APPARATUS AND NETWORK OPERATING IN APPLICATION OF THE METHOD.

PROCEEDED OF ORDER, REGULATION AND MANAGEMENT OF THE TRAFFIC NETWORK ON A PHONE NETWORK COMMUTATES INTELLIGENT

Procédé de régulation et de gestion de trafic de réseau en un point de commande de service à accès interposé dans un environnement ouvert avancé de réseau intelligent. L'invention est mise en oeuvre en un point de commande de services à accès interposé dans le réseau, et se trouve déclenchée lorsqu'une interrogation destinée à un point de commande de service d'un tiers, est générée à un point de commutation de service. Le point de commande de services à accès interposé reçoit cette interrogation et la traite pour déterminer si elle doit être envoyée au point de commande de services du tiers. Si l'interrogation ne viole pas l'un quelconque des taux ayant été négociés entre la messagerie d'échange locale et le tiers, cette interrogation est transmise du point de commande de services à accès interposé, vers le point de commande de services du tiers. Si l'interrogation amène l'un ou plusieurs des taux négociés à être violés, l'interrogation est traitée en défaut d'une manière prédéterminée, ...

METHOD AND SYSTEM FOR ADDRESS TRANSLATION AND ALIASING TO EFFICIENTLY UTILIZE UFMI ADDRESS SPACE

An address server comprising a UFMI server application adapted to allocate an inter-working UFMI to an iDEN subscriber unit and map the inter-working UFMI to an external subscriber address. The inter-working UFMI is associated with an iDEN network, and the external subscriber address is associated with an external network. The address server includes an iHLR, a data storage storing a pool of reusable iDEN UFMIs and mapped pairs of iDEN UFMIs to external subscribers. The mappings are stored on a per user, per fleet or per account basis. The data storage also stores a pool of global iDEN UFMIs. Use of the inter-working UFMIs facilitates dispatch communications between the iDEN subscriber unit and the external subscriber.

CALL FORWARDING IN A TELECOMMUNICATIONS NETWORK

Le problème que rencontrent les réseaux de télécommunications en matière de restriction de transfert d'appel, selon les techniques antérieures, est que l'accès à des numéros par transfert d'appel avec garantie d'une réponse plutôt qu'un nouveau transfert éventuel n'est pas possible. L'idée, selon l'invention, est que le transfert d'appel est autorisé malgré la restriction de transfert d'appel dans le cas où le numéro à utiliser pour ledit transfert satisfait à des conditions fixées au préalable. Le numéro est ainsi comparé avec lesdites conditions et, en fonction des résultats obtenus de cette comparaison, l'appel est, ou n'est pas, transféré.

METHOD AND SYSTEM FOR CHANGING A DIRECTION OF ESTABLISHING A TELECOMMUNICATION CONNECTION

The invention relates to a method for changing a direction of establishing a telecommunication connection in a telecommunication network, in which telecommunication connections are transmitted between the subscribers in different countries. In this method besides the telecommunication network it is used also a telecommunication connection (6), by which the connections in the desired direction are established and transmitted to the telecommunication network (7).

ROUTING SYSTEM AND METHOD FOR A TELECONFERENCING SYSTEM

In a networked teleconferencing system, a receiver (30) with a number of receiving stations (50, 52, 54) receives calls from a number of different calling stations (10, 12, 14). To get appropriate bandwidth, two calls are connected at the same time from the calling station to one of the receiving stations. The receiver (30) includes an access switch (36) that uses information from the calling station to pair calls together so that two calls from one calling station are routed to the same receiving station, regardless of the order in which the calls are received by the access switch (36).

CALL DISTRIBUTION IN DISTRIBUTED CALL CENTERS

The invention relates to a method for acceptance and processing of calls within a telephone network (2) that comprises lines for a speech and for a data channel for connecting a caller (13) with groups of subscribers (1) that are connected to the network. A group of subscribers (1) is associated with a network code via which all members of the group of subscribers (1) can be called by a caller. A control (4) distributes the calls (6) directed to a group of subscribers (1) to the individual members of the group according to predetermined criteria. The method is further characterized in that data sets are exchanged via the data channel that are associated with a connection, and a control unit (4) that is integrated in the network carries out the control function and has access to the complete data sets. A group of subscribers (1) is associated with a control unit (4) that automatically takes a decision regarding the connection of a caller (13) with a specific member of the group.

Soft switch using distributed firewalls for load sharing voice-over-IP traffic in an IP network

A switch capable of handling voice-over-IP (VoIP) traffic between calling devices and called devices. The switch comprises: 1) call application nodes for executing call process server applications, wherein a first call process server application and a similar second call process server application form a first load sharing group server application; and 2) network address translation nodes for executing firewall server applications. A first firewall server application executed on a first network address translation node is associated with a similar second firewall server application executed on a second network address translation nodes separate from the first network address translation node. The first and second firewall server applications form a second load sharing group server application. The second load sharing group server application receives VoIP traffic and selects one of the first and second firewall server applications to verify that the VoIP traffic is authorized to access ...

Methods and systems for enabling a reply call to voice mail message

Methods and systems for enabling a subscriber to reply to a voice mail message that has been made available for the subscriber at a voice mail system (VMS). In response to an indication, the VMS transmits a release message with respect to the subscriber's communication to the VMS, which communication then is terminated to an intelligent network element (INE). The INE originates a call to the party to whom the subscriber desires to reply. The subscriber's communication and the INE's call are bridged to form the reply call from the subscriber to the party. The VMS is not included in the call path. After the reply call is concluded, the bridge is dismantled, the IP is released, and the subscriber's communication is returned to interaction with the VMS without the subscriber having to make a separate call into the VMS.

Voice message system using synthetic speech

A voice message service system stores subscriber-specific voice phrases (e.g., names) using digitally encoded phonemes while common system voice phrases are stored using digitally encoded voice samples. System voice announcements are formed by combining selected common system voice phrases with selected subscriber-specific voice phrases generated from phonemes. When a voice mail message is sent from one message system to another message system, the sender's identification is sent using digitally encoded phonemes in the message header. The destination message system uses the received digitally encoded phonemes to generate the sender's identification as a synthetic voice signal.

Automated telecommunication peripheral system

Method and apparatus for providing customized telecommunication services through a peripheral node which includes a node interface for connection through incoming and outgoing telephone trunks to a carrier switch of a public switched network, an audio response unit for recording, playing, and analyzing audio signals, and a node controller for controlling operation of the node interface and the audio response unit.

Order wire accommodation system for interoffice communications

An order wire accommodation system comprises four interface boxes each of which has the same order wire interface section as the others. The order wire interface section has a circuit which digitally adds order wire signals of the transmission lines from two ports and the order wire signals which interface with the other interface boxes or a telephone through the other two ports. Each output from the order wire interface section is a sum of all the order wire signals inputted except for those of the order wire interface section itself. Because of a provision of two ports which interface externally, the interface boxes can be cascaded connected to one another, by which order wire communications among all the transmission lines accommodated in the interface boxes become possible. Further, the order wire accommodation system can be minimized and be simplified.

Portable handy phone system

A portable handy phone system comprises a first radio base station connected to a telephone network for executing radio communication with portable handy phones existing in a first radio communication zone, and a second radio base station connected to the telephone network for executing radio communication with portable handy phones existing in a second radio communication zone different from the first radio communication zone. The first radio base station comprises a first radio interface section for executing radio communication with portable handy phones in the first radio communication zone, a first system-to-system interface section for communicating with the second radio base station, and a first control section. The first control section responds to an incoming call from the telephone network designated to a portable handy phone existing in the first radio communication zone, causes the designated portable handy phone to receive the incoming call through the first radio interface ...

Method and apparatus for on-demand teleconferencing

An on-demand teleconferencing system and method for setting up an on-demand conference call in a telecommunications system having the Advanced Information Network (AIN) architecture with system signaling the number 7 (SS7) and a Public Switched Telephone Network (PSTN). A subscriber is assigned an on-demand conference call number. When that number is dialed into the PSTN, it is identified by the PSTN that the dialed number requires handling by the SS7. The SS7 links the dialed number to a conference allocation and control system (CACS) which is connected to a plurality of scalable bridge servers. The CACS selects bridge servers available to handle the conference call and based upon a selection criteria such as a peak load sets up the on-demand conference call in one of the selected bridge servers.

VOIP ROUTE SELECTION USING CALL METRICS

A method includes determining quality-of-service (QoS) metrics for each of a plurality of data streams routed by a node according to a stream-routing ratio that specifies a proportion of total data streams to be routed over each of a plurality of network paths. The method also includes determining a quality score for each of the plurality of network paths based on the QoS metrics. The method further includes modifying the stream-routing ratio based on the quality scores for the plurality of network paths to produce a modified ratio for routing new data streams, the modified ratio being selected to leverage particular ones of the plurality of network paths having high quality scores. The method still further includes applying the modified ratio to the node.

Connected communication terminal, connecting communication terminal, session management server and trigger server

The communication system according to the present invention is made up of: a connected communication terminal (1) to which communication is requested; a connecting communication terminal (3) that requests a communication with the connected communication terminal (1); a session management server (4) that manages sessions between the communication terminals; a trigger server (5) that transmits, to the connected communication server (1), the request made by the session management server (4); and routers (6 and 7) that relay between the connecting communication terminal (3) and the connected communication terminal (1), and a network (2). In the system, the connected communication terminal (1) holds a permission list (L) that is a list of communication terminals to which connection is permitted, and the session management server (4) judges, with reference to the permission list (L), whether or not the communication between the connected communication terminal (1) and the connecting communication ...

Communications systems

This disclosure concerns systems and methods for network communications. In one example, the method includes: providing subscriber terminals of each network with a respective publicly-known telephone number for allowing communications to be addressed to each subscriber terminal; associating selected subscriber terminals of each network together in a group such that those subscriber terminals can address communications to one another using a respective code allocated to each of the selected subscriber terminals; and, maintaining a database of the codes and corresponding publicly-known telephone numbers of all of the networks and providing to one of the networks a publicly-known telephone number corresponding to a code of a subscriber terminal of another one of the networks to allow communication from a subscriber terminal of one of the networks to be addressed to a subscriber terminal of the other of the networks using the code of the latter subscriber terminal.

Соnnесtеd соmmuniсаtiоn tеrminаl, соnnесting соmmuniсаtiоn tеrminаl, sеssiоn mаnаgеmеnt sеrvеr аnd triggеr sеrvеr

Тhе соmmuniсаtiоn sуstеm ассоrding tо thе prеsеnt invеntiоn is mаdе up оf: а соnnесtеd соmmuniсаtiоn tеrminаl (1) tо whiсh соmmuniсаtiоn is rеquеstеd; а соnnесting соmmuniсаtiоn tеrminаl (3) thаt rеquеsts а соmmuniсаtiоn with thе соnnесtеd соmmuniсаtiоn tеrminаl (1); а sеssiоn mаnаgеmеnt sеrvеr (4) thаt mаnаgеs sеssiоns bеtwееn thе соmmuniсаtiоn tеrminаls; а triggеr sеrvеr (5) thаt trаnsmits, tо thе соnnесtеd соmmuniсаtiоn sеrvеr (1), thе rеquеst mаdе bу thе sеssiоn mаnаgеmеnt sеrvеr (4); аnd rоutеrs (6 аnd 7) thаt rеlау bеtwееn thе соnnесting соmmuniсаtiоn tеrminаl (3) аnd thе соnnесtеd соmmuniсаtiоn tеrminаl (1), аnd а nеtwоrk (2). In thе sуstеm, thе соnnесtеd соmmuniсаtiоn tеrminаl (1) hоlds а pеrmissiоn list (L) thаt is а list оf соmmuniсаtiоn tеrminаls tо whiсh соnnесtiоn is pеrmittеd, аnd thе sеssiоn mаnаgеmеnt sеrvеr (4) judgеs, with rеfеrеnсе tо thе pеrmissiоn list (L), whеthеr оr nоt thе соmmuniсаtiоn bеtwееn thе соnnесtеd соmmuniсаtiоn tеrminаl (1) аnd thе соnnесting соmmuniсаtiоn ...

A network-based system enabling image communications

METHOD FOR PREPARING ANNOUNCEMENTS AND APPROPRIATE TELECOMMUNICATIONS NETWORK

The invention relates to a method and a telecommunications network, characterized in that one or several announcement texts (ANT) are contained in a service control point (SCP). A message (IDP) initiated by a call from a user (A) and containing information (IPCA) about the supportability of announcements by an announcing unit (IP) is received and evaluated by the service control point (SCP). If an announcement is supported by an announcing unit (IP, A-IP) a message (PAP) containing the announcement text (ANT) is transmitted by the service control point (SCP). The announcing unit (IP, A-IP) converts an announcement text (ANT) that has been received into an announcement (ANN), which is then transmitted on a speech channel to the user (A) making the call.

METHODS AND APPARATUS FOR IMPLEMENTING A NETWORK CALL CENTER

An apparatus and method for providing telephony call center services from a central location (102), such as the premises of a telephone service provider, with agents of the call center (104) located remotely from the central location. The apparatus comprises equipment and control allowing a person acting as an agent at the call center site remote from the central location, to dial a pre-assigned number in a telephony switch (152) at the call center site (104) such that the calling agent becomes connected to the switch until the agent disconnects. This connectivity allows the call center services to connect incoming calls to connected agents immediately and reliably, without delay engendered by having to dial up the agent.

Telephone agent management information system

Network based outbound call management

Data transmission system and method

INTERNATIONAL TELEPHONE INCOMING CALL REGULATOR

PURPOSE: To provide an international telephone incoming call regulator which is capable of reducing the connection of a prankster call from abroad with an international telephone operator and relieving the work burden on the operator. CONSTITUTION: When an international telephone incoming call is received from an originating terminal existing abroad, the control part 11 of a switchboard equipment 7 judges whether an incoming call is from a prankster call countermeasure object area or not from the originating call country number of the incoming call by referring to a data base part 12. In the case of the prankster call countermeasure object area, an annoucement to judge whether the call is the mischievous call or not is performed for an originating caller from a voice response part 8. When the response to this annoucement from the originating caller is received, a voice recognition part 9 judge whether this response is right or not. When the response is right, the call is connected with ...

SUBSCRIBER'S MOVEMENT TO BE AUTOMATICALLY EXECUTED BETWEEN ELECTRONIC MESSAGE SERVICE SYSTEMS IN NETWORK

PURPOSE: To automate and simplify a management procedure following movement by identifying at least one subscriber by a 1st system, automatically communicating requested information in response to the reception of 2nd information to be identified and starting the offer of the information to the subscriber by a 2nd system. CONSTITUTION: An input for identifying at least one subscriber 25 to be moved and a system 12 in which the subscriber 25 moves is received from a system controller 29. A device in the source system 12 automatically contacts with destination system information, i.e., subscriber's profile information 210, requested to be provided to the subscriber 25 to be moved. The contacted information 210 is the contents of a subscriber's mail box 200 and the device of the system 12 including messages such as an address book to be stored in the information 210 in addition to the information 210 automatically informs other systems 13, 14 in a network of subscriber's movement. Consequently ...

СПОСОБ УМЕНЬШЕНИЯ ВРЕМЕНИ УСТАНОВЛЕНИЯ СОЕДИНЕНИЯ ДЛЯ 3G-324M

Изобретение относится к технике связи. В частности, к установлению соединения между двумя системами, совместимыми со стандартом 3G-324M. Предложены способ и устройство для обеспечения быстрого и гибкого установления соединения. Множество предварительно конфигурированных каналов устанавливают в канале связи. Предварительно назначают предварительно конфигурируемый канал первому способу согласования сеанса связи. Из оставшихся каналов предварительно назначают предварительно конфигурируемый канал второму способу согласования сеанса связи, при этом скорость первого способа согласования отличается от скорости второго способа согласования. Назначают предварительно конфигурированный канал, оставшийся после назначения каналов первому и второму способу согласования, фиксированному рабочему режиму среды передачи, посредством чего первый способ согласования сеанса связи используется для гибкого установления сеанса связи. Технический результат заключается в уменьшении времени установления соединения ...

УСТРОЙСТВО МЕЖСЕТЕВОГО ВЗАИМОДЕЙСТВИЯ

Изобретение относится к системам связи. Технический результат заключается в усовершенствовании процедуры межсетевого взаимодействия. Устройство межсетевого взаимодействия содержит блок интерфейса IuCS (interconnection point circuit switched, интерфейс точки межсоединения с коммутацией каналов) для сигнала протокола IuUP (interconnection User Plane, протокол плоскости пользователя интерфейса точки межсоединения); блок интерфейса Mb для сигнала протокола RFC3267; и средство преобразования протоколов для выполнения преобразования протоколов между сигналом протокола IuUP и сигналом протокола RFC3267, причем средство преобразования протоколов соединено с блоком интерфейса IuCS и блоком интерфейса Mb. 2 н. и 7 з.п. ф-лы, 5 ил.

УСТРОЙСТВО МЕЖСЕТЕВОГО ВЗАИМОДЕЙСТВИЯ

... 1. Устройство межсетевого взаимодействия, содержащее: ! блок интерфейса IuCS для сигнала протокола IuUP; ! блок интерфейса Mb для сигнала протокола RFC3267; и ! средство преобразования протоколов для выполнения преобразования протоколов между сигналом протокола IuUP и сигналом протокола RFC3267, причем средство преобразования протоколов соединено с блоком интерфейса IuCS и блоком интерфейса Mb. ! 2. Устройство по п.1, в котором блок интерфейса IuCS принимает сигнал протокола IuUP, блок интерфейса Mb передает сигнал протокола RFC3267, и средство преобразования протоколов преобразует сигнал протокола IuUP в сигнал протокола RFC3267. ! 3. Устройство по п.2, в котором средство преобразования протоколов содержит: ! анализатор сигнала, который извлекает значение ИККД (индикатор комбинаций подпотока канала радиодоступа) и значение ККК (классификатор качества кадра) из сигнала протокола IuUP и устанавливает значение ТК (индекс типа кадра), бит Q (бит индикатора качества кадра) и значение ЗРК (запрос ...

Koppelnetzwerk für Kommunikationseinrichtungen

The invention concerns a communication device interface which is used for optional connection of input lines (E1, ..., E128) assembled to form a plurality of input line groups to at least one output line group (A1, ..., A16), each of the input line groups being connected to a separate interface element (SE) of at least one interface element arrangement (e.g. KA21) with a funnel structure. Each interface element arrangement further comprises a number of outputs corresponding to the number of output lines belonging to the respective output line group. Each of these outputs is connected to one of the output lines belonging to the respective output line group. According to the invention, the respective interface element arrangement (KA21) consists simply of the interface elements (SE) connected to the input line groups. Each of the interface elements has outputs which are individually associated with the output lines belonging to the respective output line group. Only one of the mutually corresponding ...

Verfahren und Gerät zum Rufen mehrerer Fernsprechendgeräte bei einem ankommenden Gespräch

Verfahren und Gerät zum Adressieren von Anwendern in einem Netz

Automatische Dienstklassen-Leitweglenkung

Method and arrangement for multimedia communication

Screening of incoming telephone calls prior to call completion to the destination party

Arrangement system and method relating to telecommunications access and control

Switching method and system with portable directory numbers

A method of processing a call within a cluster of PABXs is comprised of receiving a dialed directory number designating a called line within the cluster, accessing a database and obtaining an identity of a terminating PABX within the cluster on which a line corresponding to the dialed directory number is terminated, seizing a trunk, and generating and transmitting a message to the terminating PABX which identifies the terminating PABX and the dialed directory number, at the terminating PABX, recognizing the terminating PABX identification and as a result translating the dialed directory number to an electrical line location, and completing the call on the terminating PABX to the line corresponding to the dialed directory number.

Wireless internet telephone and system

A wireless Internet telephone (10) enables placement of long-distance telephone calls via an Internet connection, thereby avoiding long-distance telephone charges associated with proprietary long-distance networks. The wireless Internet telephone (10) includes a wireless communication subsystem (20) that enables the wireless Internet telephone (10) to access an Internet telephony provider using a set of Internet telephony parameters (200).

Soft switch using distributed firewalls for load sharing voice-over-ip traffic in an ip network

A switch, 105, capable of handling voice-over-IP (VoIP) traffic between calling devices and called devices. The switch comprises: 1) call application nodes, 115A-115C, for executing call process server applications, wherein a first call process sever application, CP1(P)-CP3(P), and a similar second call process server application, CP1(B)-CP3(B), form a first load sharing group server application, LSG1; and 2) network address translation nodes, 110A-110E, for executing firewall server applications. A first firewall server application, FWP1(P)-FWP5(P), on a first network address translation node, NATP1(P)-NATP5(P), is associated with a similar second firewall server application, FWP1(B)-FWP5(B), on a second network address translation node, NATP1(B)-NATP5(B), separate from the first network address translation node. The first and second firewall server applications form a second load sharing group server application, LSG3. The second load sharing group server application receives VoIP traffic ...

Telephone communication system

System and method for efficient call management for directory assistance services

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.

Systems and methods of establishing user groups in an internet protocol environment

A IP telephony service allows customers to form user groups. A user group is established by collecting a plurality of identifiers that are associated with members of a group, and associating the member identifiers with a group identifier. The member identifiers could be telephone numbers of telephony devices for each of the members, or device IDs of IP telephony devices for members of the group. The group identifier could be any type of identifier, and in some instances, the group identifier could be a telephone number.

Systems and methods for terminating communication requests

A IP telephony service allows customers to form user groups. Each user group can include multiple telephony devices that are associated with one or more users. One or more group identifiers would be associated with each user group. When an incoming communication is directed to a user group, a group identifier is used to retrieve a list of the members of the group, or a list of devices that correspond to the members of the user group. The communication is then sent to one or more members of the group, or to one or more of the devices that correspond to members of the user group. Handling preferences may determine how the incoming communication is delivered. In some instances, the incoming communication could be a telephone call. In other instances, the incoming communication could be a SMS message or an instant message.

Method And System For Providing Location Aware Tracking And Services Via An IP Multimedia Residential Gateway

In an IP multimedia gateway, locations corresponding to communicatively coupled communication devices are determined and transmitted to a service provider device (SPD) that provides services to the communication devices. Communication devices receive location based service profiles from the (SPD) based on the determined locations. Services content from the (SPD) is communicated with the communication devices and/or with the (SPD), based on the profiles. The determined locations are received from communication devices which may be local or remote. A device location map is generated to be displayed by communication devices. The gateway may be integrated in an STB/DTV that displays the determined locations. Content is communicated to communication devices based on their location. The (SPD) may provide content and/or control information for communication device services. Locations are associated with client devices for detecting cloned client devices where determined locations are different than associated locations and/or for restricting services based on location.

System and apparatus for voice/video communication and method thereof

A voice/video communication system is disclosed. The system includes a caller having a first PSTN subsystem and a first data subsystem, and a callee, wherein the first data subsystem encodes the caller's public IP address/port into a first key sequence and triggers the first PSTN system to send first audio coding representing the first key sequence to the callee.

System And Method For Identifying Audio Command Prompts For Use In A Voice Response Environment

A system and method for identifying audio command prompts for use in a voice response environment is provided. A signature is generated for audio samples each having preceding audio, reference phrase audio, and trailing audio segments. The trailing segment is removed and each of the preceding and reference phrase segments are divided into buffers. The buffers are transformed into discrete fourier transform buffers. One of the discrete fourier transform buffers from the reference phrase segment that is dissimilar to each of the discrete fourier transform buffers from the preceding segment is selected as the signature. Audio command prompts are processed to generate a discrete fourier transform. Each discrete fourier transform for the audio command prompts is compared with each of the signatures and a correlation value is determined. One such audio command prompt matches one such signature when the correlation value for that audio command prompt satisfies a threshold.

Intelligent formatting of voip telephone numbers

A system and method are disclosed for intelligent formatting of VoIP telephone numbers. The intelligent VoIP formatting system includes a presentation and user interface layer, an E.164 formatting engine, a location routine, and a database of telephone number data which are used to deconstruct input telephone numbers and reconstruct them as E.164 compliant telephone numbers.

Distributed Hardware/Software System for Managing Agent Status in a Communication Center

An agent presence application for monitoring target agent resources and rendering agent states to subscribing applications is provided. The presence application has at least one first portion for collecting data regarding states of activity of the target agent resources, and at least one second portion for integrating the data and rendering the agent states to the subscribing applications.

Mobile Automatic Location Identification (ALI) for first responders

Automatic Location Indication (ALI) data is digitally transferred from a Public Safety Answering Point (PSAP) to a first responder via a web site accessed by the first responder. The web site lists each live E911 call within their jurisdiction, and appends the caller's ALI and/or other location data relative to each call. Preferably the first responders (e.g., police car, paramedic, ambulance, fire truck, etc.) can view not only the emergency caller's information for those E911 calls that they are responding to, but also information about other emergency callers that other first responders are responding to. This facilitates the actual response to individual emergency calls as well as the overall management of multiple responders to multiple emergency calls.

Retrieval and presentation of network service results for mobile device using a multimodal browser

A method of obtaining information using a mobile device can include receiving a request including speech data from the mobile device, and querying a network service using query information extracted from the speech data, whereby search results are received from the network service. The search results can be formatted for presentation on a display of the mobile device. The search results further can be sent, along with a voice grammar generated from the search results, to the mobile device. The mobile device then can render the search results.

Method and Apparatus for Telephonically Accessing and Navigating the Internet

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.

Determination of bypass zones from network configuration settings

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.

Method And Apparatus For Providing An Identifier For A Caller Id Function In A Telecommunication System

Method, system, and computer readable medium for providing an identifier for a caller identifier (ID) function when processing a communication request between a calling party and a called party in a communication network is described. In some examples, an initial identifier for the caller ID function associated with the calling party is received. A determination is made if the calling party has a virtual identifier configured in the communication network. A determination is made if the called party has a predefined identifier associated with the calling party configured in the communication network. The initial identifier is replaced with a terminating identifier based on configuration of the virtual identifier and the predefined identifier in the communication network.

Method and system to automatically park a voice call for data transfer

A system and method to automatically park a voice call on a mobile electronic device so that the device can receive data, comprising a switch for establishing a voice call at the at least one caller device, a database for storing at least one call processing rule, and an enterprise data server for receiving a data message, comparing contents of the data message with at least one rule in the database, and in the event that the contents match the rule then parking the call, switching the mobile electronic device to a data mode of operation, downloading the data message to the device and un-parking the call.

Method for allocating resource in lte system

The present invention relates to a fractional RB (Resource Block) allocation method which enables resource allocation in a unit smaller than RB to improve the capacity of VoIP (Voice over Internet Protocol) in an LTE (Long Term Evolution) system. That is, the present invention introduces a resource arrangement method which allows different users to share the same resource, without even using an SDMA, and a method for informing the fractional RB allocation and how to use the same. A method for introducing multiplexation into an RB and changing the RE arrangement is proposed to support the downlink fractional RB allocation. A method for changing the RE arrangement is also proposed to support the uplink fractional RB allocation. According to the present invention, to provide a consistent service with a small amount of traffic generated as in the VoIP, resources of a smaller amount than a single RB can be used for a terminal with a good channel state. In the present invention, the occupancy of resources for the terminals of an excellent channel state is reduced so that the capacity of the VoIP can be improved.

Method and apparatus of providing messaging service and callback feature to mobile stations

Disclosed are an apparatus and method of performing automated administrative operations on a mobile device. One example method may include determining via a hosted server that an action needs to be performed by a mobile device under management. The method may further include generating a message via the hosted server, the message includes the action, and transmitting the message to the mobile device under management, and receiving a result message from the mobile device under management responsive to the transmitted message, the result message indicating that the action has been satisfied.

Systems and Methods for Remote Deletion of Contact Information

An exemplary system for providing remote deletion of contact information includes a first device associated with a first user. The first device is configured to generate and send a delete request message to a second device. In some embodiments, the delete request message includes a request to delete contact information for the first user from the second device. The second device can be configured to receive the delete request message and delete the first user's contact information. A method for providing remote deletion of contact information includes a delete request message being generated at a first device, associated with a first user. The delete request message can include a request to delete contact information for the first user. The first device can send the delete request message to a second device. The delete request message can instruct the second device to delete the first user's contact information.

Providing Called Number Characteristics to Click-to-Dial Customers

A system and method to provide content and call attributes for a destination phone number using a click-to-dial connection includes accepting a query, retrieving links to a document, and searching through cached data using the query to identify relevant or destination number information. If no match is found, the method continues with accessing the document identified by the link for identifying relevant number information and creating a click-to-dial icon to link to the relevant numbers included in the relevant phone number information identified by the query. Next, a popup box for content relevant to the click-to-dial icon and a click feature on the popup box are created to retrieve call destination attributes for viewing by a user. Further, computer instructions create at least one communication connection between two communication devices after viewing call destination attributes.

Methods, systems and computer-readable media for conducting communications

The present disclosure relates to methods, systems, and computer-readable media for conducting communications. One method includes determining a context associated with a communication between a first participant and a second participant while the communication is occurring, determining a failure of the communication, and resuming the communication based, at least in part, on the context.

Telephone relay apparatus, telephone relay method, and program

A telephone relay apparatus 200 receives a connection request from a first telephone terminal 100 to a second telephone terminal 300. The connection request includes phone numbers of the first telephone terminal 100 and the second telephone terminal 300. The telephone relay apparatus 200 stores phone numbers of a plurality of telephone terminals and codec information that indicates types of audio codecs of the telephone terminals so that the phone numbers and the code: information correspond to each other. When connecting to the first telephone terminal 100, the telephone relay apparatus 200 reads out codec information that corresponds to the phone number of the second telephone terminal, and then transmits the read-out codec information to the first telephone terminal 100. The first telephone terminal 100 performs a call process using an audio codec that corresponds to the codec information transmitted from the telephone relay apparatus 200.

Method and system for performing remote network management on a voice home gateway terminal

This disclosure provides a method for performing remote network management on a voice home gateway terminal, and the method includes that: a remote network management server transmits voice signaling to the voice home gateway terminal; the voice home gateway terminal determines whether the voice signaling contains a non-voice triggering identifier, processes a voice service according to the voice signaling when no non-voice triggering identifier is contained, requests non-voice configuration information from the remote network management server when a non-voice triggering identifier is contained, and processes a non-voice service according to the received non-voice configuration information. This disclosure further provides a system for performing remote network management on a voice home gateway terminal. Application of the method and system enables the remote network management server to manage simultaneously a voice service and other non-voice services. The remote management performed on the voice home gateway terminal having a voice function is reliable, and this disclosure realizes the convergence for managing the integrated services of the voice home gateway terminal.

Method and system for provisioning packetized voice communication services

An approach for enabling a customer to provision resources of a packetized voice service provider based on one or more customer conditions is described. A provision management platform receives a request for provisioning a domain name service as part of packetized voice service. In response to the request, the provision management platform prompts via a graphical user interface to obtain information for populating a service record of the domain name service. The provision management platform then generates the service record for storage corresponding to a customer account for the packetized voice service.

Contact Number Encapsulation System

A contact number encapsulation system is disclosed that is operable to consolidate a plurality of contact numbers associated with a given user into a single contact number. The system includes a call processing application that uses information contained in a message header to determine a list of contact numbers to call. The message header contains a call destination indication that is used by the call processing application to retrieve the list of contact numbers. The list of contact numbers is user defined and may be updated or modified by an account holder at any given time. Once the list of contact numbers is obtained, a simultaneous call module is used to simultaneously call each number. The account holder only needs to provide the primary contact number to individuals desiring to reach them instead of various numbers that are associated with locations the account holder may be located.

System and method for automatically establishing a concurrent data connection with respect to the voice dial features of a communications device

A system and method for automatically establishing a concurrent data connection between a communications device and a remote data server upon initiating a voice call. A data connection application can be configured in association with a communications device in order to concurrently connect the communications device and a remote data server on initiating a voice call. A directory server can be employed to match an outgoing contact number with respect to the communications device with a list of data-active contact numbers stored in the directory server. If a match is found, the location data can be transmitted to the communications device. The data connection application can be further communicated with the remote data server in order to visually transmit a wide range of potential business application associated with the remote data server to the communications device.

Communications terminal, a system and a method for internet/network telephony

A communication terminal for Internet telephony is provided that handles and control communication of data in accordance with a standardized network protocol and exchanges data with a connecting unit connected to the Internet where the resulting data exchanged between the terminal and a connecting unit consist of packets in a standardized protocol data packet format embedded in a wireless format. This provides a communications terminal which uses a network or the Internet for the transfer of digitized speech, etc., thereby achieving great economic savings. Also, the flexibility is increased with respect to wireless communication with the network or the Internet without any need for specialized equipment and functionality.

Automated billing and distribution platform for application providers

Systems and methods according to certain embodiments of the invention include integrating a network-enabled application with a platform having a plurality of users and a plurality of communication channels with a respective plurality of wireless network carriers. The systems and methods include providing a developer library for use by developers in creating their network-enabled applications.

Systems and methods for initiating communications with contacts based on a communication specification

Disclosed herein are methods and systems for initiating communications with contacts based on a communication specification. According to an aspect, a method may include receiving a communication specification. The communication specification may include identification of one or more contacts, a communication persistence parameter, a geographic location, an event, a date, and a communication channel. The method may also include determining one or more contacts based on the communication specification. Further, the method may include initiating multiple communications with the contacts via at least one communication channel. A communication channel may be a telephone call, an email, a text message, an instant message, and the like.

Apparatus, systems and methods for managing incoming and outgoing communication

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.

Video call method and system thereof

A video call method is disclosed herein and includes the following steps. The step of installing a high definition video call router (HDVCR) program in at least two mobile devices; the step of activating the HDVCR program in one of the mobile devices to start a video call; the step of connecting the at least two mobile devices; the step of enabling at least two routers by the HDVCR program to search for each other over Internet; the step of transmitting a plurality of video streams between the at least two routers; and the step of sending a plurality of high definition video images respectively to at least two displays from the at least two routers.

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

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.

Method for re-configuring a communications device

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 ).

One number, intelligent call processing system

Systems and methods for retrieving household data based on an origination identifier. In an embodiment, an origination identifier of a communication is captured. The origination identifier is indexed into a master table comprising a plurality of records. Each of the records comprises an association between an origination identifier and a universal database linkage key, and each universal database linkage key comprises an index into one or more databases. A universal database linkage key associated with the captured origination identifier is retrieved and indexed into one or more databases. Household data associated with the captured origination identifier is retrieved from the one or more databases and communicated to at least one recipient.

Peer-to-peer, internet protocol telephone system with auto-attendant

Various embodiments of the invention provide a Peer-to-Peer (P2P, Internet Protocol (IP) telephone system. The telephone system includes a plurality of terminals coupled together via an IP network. The terminals cooperate with one another to provide telephony features without a dedicated central controller such as a PBX and/or a KSU controller. The terminals may store system-wide configuration data and files referenced by the system-wide configuration data. The terminals may further determine whether the system-wide configuration data references a file that is not stored in the terminal, and requesting the file from another terminal of the telephone system in response to determining the terminal does not have a copy of the file.

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

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).

Spinning off chat threads

A solution for providing an instant messaging (IM) space, such as for a chat session, is provided, which includes provisions for spinning off of chat threads. An embodiment of the invention provides a method of providing an IM space, the method comprising: providing a first graphical user interface (GUI) space for an IM conversation; obtaining a selected portion of the IM conversation represented on the first GUI; and creating a second GUI space for a second IM conversation based on the portion selected.

System and method for selecting network egress

A system and method for selecting communications routing. Network performance information is gathered from a communications network using performance information packet data packets. A network connection offering the best quality of service is selected between the communications network and an outside network based on the network performance information. The network connection is established between the communications network and the outside network for routing communications.

Method for providing clean endpoint addresses

A method for providing clean endpoints including managing a collection of endpoints; testing communication of an endpoint of the collection of endpoints; and generating a cleanliness score for the endpoint according to results of the test. The method functions to create a metric that can be used to evaluate undesired communication activity and functionality of an endpoint. The method may be implemented by any suitable platform providing endpoint addresses to clients, which may include service providers, internet-telephony platforms, or any suitable entity. The method is preferably applied to telephony endpoints used in internet-telephony applications, but may be used in any suitable application.

Facilitating a text message conversation using tele-links

One embodiment of the present invention provides a system for facilitating a text message conversation using tele-links. During operation, the system receives a text message from a service at a mobile device, wherein the text message includes a tele-link, wherein the tele-link is comprised of a phone number. Next, the system receives a selection of the tele-link from a user at the mobile device. In response to the selection, the system then automatically initiates a call from the mobile device to a number indicated by the tele-link, wherein the act of calling the number results in the service sending a second text message to the mobile device. Finally, the system receives the second text message from the service at the mobile device.

System and method for managing a computing cluster

A method and system for managing a computing cluster including hosting a plurality of machines in a networked computing cluster, wherein the plurality of machines include service instances running on hosts, where the services have configured machine state; and wherein machine state includes configuration data and software of the machine; integrating the plurality of machines of the cluster with at least one configuration controller component; and the at least one configuration controller component, altering the machine state of at least one service instance.

Routing voip calls through multiple security zones

Call setup signaling is performed across at least a first security zone, a second security zone, and a third security zone to set up a call. At least one gate is then established between the first security zone and the third security zone to enable traffic flow for the call between the first security zone and the third security zone.

Method and Apparatus for Automatically Building Conversational Systems

A system and method provides a natural language interface to world-wide web content. Either in advance or dynamically, webpage content is parsed using a parsing algorithm. A person using a telephone interface can provide speech information, which is converted to text and used to automatically fill in input fields on a webpage form. The form is then submitted to a database search and a response is generated. Information contained on the responsive webpage is extracted and converted to speech via a text-to-speech engine and communicated to the person.

Method for selection and routing of an outbound voice call to an appropriate network for completion

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

Tandem Access Controller Within The Public Switched Telephone Network

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.

Pbx mobility system with multiple call legs

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.

Using pstn reachability to verify voip call routing information

A system for verifying VoIP call routing information. In particular implementations, a method includes verifying one or more Voice-over-Internet-Protocol (VoIP) call agents for respective destination telephone numbers based on demonstrated knowledge of previous public switched telephone network (PSTN) calls to the respective destination telephone numbers; receiving a call initiation message identifying a destination telephone number; and conditionally initiating a call over a VoIP network to a target VoIP call agent, or over a circuit switched network, based on whether the target VoIP call agent has been verified for the destination telephone number identified in the call initiation message.

System and Method for Delivering Advertising to Packet-Based Telephones

A system and method of delivering targeted advertising to packet-based network telecommunications devices may include accessing a database configured to store telephone services information of telecommunications devices. The telecommunications services information may indicate whether a telecommunications device has packet-based network access. Telecommunications devices that have telephone number information including NPA-NXX of a certain NPA-NXX code and packet-based network access may be identified. Advertising content may be communicated to the identified telecommunications devices for display on the telecommunications device.

System and method for providing call handling in law enforcement actions

A method for providing law enforcement call preemption of VoIP subscribers may include receiving a call preemption request from a law enforcement agency (LEA) designating at least one subscriber telephone number and at least one LEA telephone number, wherein the at least one subscriber telephone number is associated with a voice over Internet protocol (VoIP) calling plan subscriber. An incoming call policy may be generated at a voice application server and configured to allow incoming calls to the at least one subscriber telephone number from only the at least one LEA telephone number. The incoming call policy may be stored in a memory associated with the voice application server. The incoming call policy may be activated. An outgoing call transfer policy may be generated at the voice application server and configured to automatically transfer all outgoing calls from the at least one subscriber telephone number to a designated one of the at least one LEA telephone numbers. The outgoing call transfer policy may be stored in the memory associated with the voice application server. The outgoing call transfer policy may be activated.

Telephone number-based advertising

Methods, systems, and apparatus, including computer program products, in which a first telephone number is associated with a second telephone number, and a web-based advertisement is served for an advertiser that includes the first telephone number. A call placed to the first telephone number is received, and the call is routed to the second telephone number based on the association of the second telephone number with the first telephone number. A determination as to whether a call-through event in response to receiving the call placed to the first telephone number has occurred, and an advertiser associated with the second telephone number is charged a cost-per-call amount if the call-through event is determined to have occurred.

System, method, and computer-readable medium for mobile-to-mobile calls within femtocell network

A system, method, and computer readable medium for terminating calls in a network system is provided. The communication system features an IP-based femtocell system for provisioning communication services to a user equipment. The femtocell system is provisioned with a list of electronic serial numbers of user equipments that are authorized to access the femtocell system. When a call is received by the femtocell system, the femtocell system determines a directory number of a user equipment to which the call is directed. If the destination user equipment is authorized to access the femtocell system and is currently within the service area of the femtocell system, the call setup may be completed by the femtocell system without any call setup signaling being transmitted to a core telecommunication network.

Processing files from a mobile device

The presently described subject matter allows the user to interactively browse a collection of electronic files, such as a digital photo collection, remotely using a mobile telephone. Files from the collection of electronic files can be directed from the mobile telephone to be sent to a remote display device, where the file can be displayed. Also, a user can record voice and text annotations from the mobile phone to preserve with an electronic file, and which is transported along with the file when it is displayed on a remote display device.

System and method for managing a computing cluster

A method and system for managing a computing cluster including hosting a plurality of machines in a networked computing cluster, wherein the plurality of machines include service instances running on hosts, where the services have configured machine state; and wherein machine state includes configuration data and software of the machine; integrating the plurality of machines of the cluster with at least one configuration controller component; and the at least one configuration controller component, altering the machine state of at least one service instance.

Ship Based Cellular and Satellite Communication

A ship based cellular and satellite transceiver receives, demodulates and processes data, voice and video signals from a satellite transmitter and provides the processed signals in cascade to a OFDM and to a CDMA modulator and transmitter for transmission to a ship based mobile unit. The OFDM signal is transmitted in a Wi-Fi network and the CDMA signal is transmitted in frequency bands used by cellular phones and cellular connected devices. The ship based transceiver receives from a mobile unit, e.g. from a cellular phone or a computer an OFDM or a CDMA signal generated and transmitted by the mobile unit and receives, demodulates and processes the cellular phone transmitted signal and provides these processed signals for cascade processing, modulation and transmission to a satellite.

SS7 ANSI-41 to SIP Based Call Signaling Conversion Gateway for Wireless VoIP E911

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.

Personal IP Toll-Free Number

In an internet telephony system, toll-free service and the calls that are accepted and paid for can be specified by the IP telephony system user to be limited according to the calling party's identity, time of day, day of week or other criteria. Data that is used to screen or filter incoming calls is programmed into a call manager by the IP telephone system user's broadband residential gateway. Using such a system the user can carefully screen calls that are paid for and more closely control telecommunications service costs. More generally, the user has the ability to store predefined call handling criteria, which may be executed by the broadband residential gateway and/or the call manager.

Systems, devices and methods for providing access to a distributed network

Systems and apparatus, and methods relating thereto, can be implemented to include base station having a transceiver adapted to establish a radio-frequency link with a mobile telephone; a first interface, separate from said transceiver, that is adapted for communication over the public Internet; and a controller. The controller can be adapted to obtain, from a server accessed via the public Internet, gateway address information for a remote gateway that provides an interface between the public Internet and a network of a telephone service provider and route data from the mobile telephone, over the public Internet, to the remote gateway.

Dual Tone Multi-Frequency Signal Transmission Method and Device

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.

System and method of providing recommendations

A method of providing recommendations is disclosed and includes receiving a selection of an offering via an online site. The method also includes outputting a detail page related to the offering via the online site. The detail page includes a first selectable indicator corresponding to a social network of a visitor and a second selectable indicator corresponding to unfiltered content. The method also includes outputting at least one recommendation via the detail page. Each recommendation is associated with a member of the social network when input received via the online site indicates a selection of the first selectable indicator.

System and methods to connect people in a marketplace environment

Systems and methods to provide telephonic connections. In one embodiment, a method includes: providing a listing on a marketplace, the listing including an assigned reference to be used to initiate a real time communication connection; and billing for a real time communication lead generated from presenting the listing in response to a real time communication initiated via the reference. In one embodiment, a method includes: conducting online auction of telephonic leads; determining a way to present listings based at least partially on received price bids on the telephonic leads (e.g., the order of placement, the selection for placement, etc); and providing the listings with information to initiate a telephonic connection in a way trackable to measure a number of telephonic calls generated from presenting the listings.

Efficient and cost-effective distributed call admission control

A distributed call control system is provided that can allot bandwidth amongst several call controllers. The distributed call control system includes one or more access elements that interface with a cloud that execute two or more instances of call processing servers that administer call control. The cloud members negotiate and determine bandwidth allocation amongst the members and the access elements. If an access element requires more bandwidth, the access element assesses its own needs and requests more bandwidth from the cloud. The negotiation and requests for bandwidth are accomplished with a set of dynamic and static bandwidth data that regiment the control of the bandwidth.

System and method for processing telephony sessions

In one embodiment, the method of processing telephony sessions includes: communicating with an application server using an application layer protocol; processing telephony instructions with a call router; and creating call router resources accessible through a call router Application Programming Interface (API). In another embodiment, the system for processing telephony sessions includes: a call router, a URI for an application server, a telephony instruction executed by the call router, and a call router API resource.

Computer telephony

Computer-telephony events are initiated by a user operating a computer terminal to control, through a computer-telephony controller, operation of a communications terminal. The computer-telephony controller receives from a first computer terminal a request for a first computer-telephony event; in which the request for the first computer-telephony event comprises a label value. The computer-telephony controller refers to a mapping between the label value and an identifier of a communications terminal; and initiates the requested first computer-telephony event controlling operation of the communications terminal. The computer-telephony controller then receives, from second computer terminal 2 B, a request for the label value. The computer-telephony controller authenticates the label value request and provides to second computer terminal the requested label value. The computer-telephony controller then receives from second computer terminal a request for a second computer-telephony event, in which the request for the second computer- telephony event comprises the label value. The computer-telephony controller refers to the mapping; and initiates the requested second computer telephony event controlling operation of the communications terminal.

Easy call for content

A method and apparatus for quickly and easily requesting content over a mobile device via a telephone network without requiring user entry of a number of pieces of information or having to forward through to certain request screens and/or browse for content. The method includes verifying a mobile device based on an incoming telephone call to a particular destination telephone number; retrieving by a content server the specific content associated with the particular destination telephone number in which contact has been initiated by the mobile device; and providing the specific content to the mobile device requesting the content. Specific content includes but is not limited to broadcast data, satellite transmitted data, and cable network data provided during a telephone call, so that a mobile device can receive everything from television shows to news broadcasts and educational shows, etc. so long it is within range of a telephone network.

System and method for initiating online social interactions based on conference call participation

Disclosed herein are systems, computer-implemented methods, and tangible computer-readable storage media for automatically initiating online social interactions based on participation in a conference call. Audio of the conference call is analyzed for data, such as timestamps indicating when people spoke, the volume of speech, words per minute, and emotional context. This analysis determines relationships between participants, and initiates online social interactions compatible with those relationships. This initialization can take the form of a prompt, or can be performed without interruption to the participants.

Mechanism for dynamic signaling of encoder capabilities

The present disclosure provides systems and methods for dynamically signaling encoder capabilities of vocoders of corresponding communication nodes. In one embodiment, during a call between a first communication node and a second communication node, a control node (e.g., base station controller or mobile switching center) for the first communication node sends capability information for a voice encoder of a vocoder of the first communication node to a control node for the second communication node. As a result, the second communication node is enabled to select and request a preferred encoder mode for the voice encoder of the vocoder of the first communication node based on the capabilities of the voice encoder of the vocoder of the first communication node.

Integrated private branch exchange and device control system

In one embodiment, a state center operates as a repository and exchange point for telephony state information that would not typically be accessible from an Internet Protocol private branch exchange (IP PBX), and control state information that would not typically be accessible from a programmable multimedia controller. In such manner, the state center enables telephony responsive audio/video (A/V) and home automation device control, and A/V and home automation device responsive telephony control.

Phone number recognition

Method and system for recognising a numeric or alphanumeric sequence of characters in a document, the sequence conforming to predetermined rules and representing user identifiers for identifying users in a communication system include identifying a country of origin of the document, recalling rules relating to the format of the sequence associated with the determined country of origin, searching the document to identify any sequence in the document satisfying the format and returning any such sequence.

Electronic tool and methods for meetings

An electronic meeting tool for communicating arbitrary media content from users at a meeting includes a node configuration operating a display node of a communications network that is coupled to a display. The node configuration receives user selected arbitrary media content and controls display of the user selected arbitrary media content on the display. At least one peripheral device communicates the user selected arbitrary media content via the communications network. The peripheral device is a connection unit including a connector that couples to a port of a processing device having a second display, a memory and an operating system; and a transmitter communicating with the communications network. A program is provided to run on the operating system of the processing device and obtains user selected arbitrary media content, while leaving a zero footprint on termination.

System and Apparatus for Managing Calls

A system and apparatus for managing calls to voice mail is disclosed. A system that incorporates teachings of the present disclosure may include, for example, a voicemail system having a controller element that receives a call redirected away from a targeted communication device without providing the targeted communication device notice of the call responsive to a voicemail redirection request supplied by signaling information associated with the call. Additional embodiments are disclosed.

Method and apparatus of providing messaging service and callback feature to mobile stations

Disclosed are an apparatus and method of performing automated administrative operations on a mobile device. One example method may include determining via a hosted server that an action needs to be performed by a mobile device under management. The method may further include generating a message via the hosted server, the message includes the action, and transmitting the message to the mobile device under management, and receiving a result message from the mobile device under management responsive to the transmitted message, the result message indicating that the action has been satisfied.

Voice instant messaging

Systems and techniques for transferring electronic data include enabling instant messaging communication between a sender an at least one recipient through an instant messaging host. In addition, voice communication is enabled between the sender and the recipient through the instant messaging host.

METHODS AND APPARATUS TO PROTECT AND AUDIT COMMUNICATION LINE STATUS

Methods and apparatus to protect and audit communication line status are disclosed. An example method includes receiving a set of available communication lines from a media gateway, associating a plurality of destination numbers with respective ones of the available communication lines, and dialing the plurality of destination numbers from the media gateway to identify the communication line status information. 1. A method to conditionally block a destination number , comprising:dialing the destination number via a media gateway;identifying a designated operation mode of the destination number as one of fax-only or voice-only;blocking the destination number when negotiation tones are detected and the destination number is designated as voice only; andblocking the destination number when negotiation tones are absent and the destination number is designated as fax-only.2. (canceled)3. A method as defined in claim 1 , wherein dialing the destination number emulates an outbound modem connected to a set of communication lines communicatively connected to the media gateway.4. A method as defined in claim 1 , wherein identifying the designated operation mode comprises identifying at least one of a fax machine or a modem associated with the destination number.5. A method as defined in claim 1 , further comprising determining which of a plurality of communication lines is free of communication services.6. A method as defined in claim 5 , wherein the communication services comprise at least one of telephone traffic claim 5 , facsimile traffic claim 5 , or modem traffic.7. (canceled)8. A method as defined in claim 1 , wherein the destination number is retrieved from a file comprising a plurality of destination numbers.9. A method as defined in claim 8 , further comprising parsing the plurality of destination numbers from the file.10. A method as defined in claim 1 , further comprising receiving a log file from the media gateway to identify network devices associated with a ...

System and Method for Handling Toll Free Number Telephone Calls

A system and method for processing toll free telephone calls by a telecommunications service provider may include receiving, by the telecommunications service provider, a telephone call request for a telephone call being placed to a toll free telephone number from a telephone within a service area of the telecommunications service provider. The telecommunications service provider may determine that the toll free telephone number is one operated by the telecommunications service provider. The telephone call may be routed via a network of the telecommunications service provider to a telephone associated with the toll free telephone number to connect the telephone call. 1. A system for processing toll free telephone calls by a telecommunications service provider , said system comprising:a local switch operated by the telecommunications service provider, and configured to receive a telephone call request by a customer in a local service area serviced by the telecommunications service provider;a signaling transfer point (STP) in communication with said local switch, and configured to, in response to receiving information associated with the telephone call request from said local switch, determine that the telephone call is being placed to a toll free telephone number; in response to receiving the information associated with the telephone call request, determine whether the telephone call is being placed to a toll free telephone number that is operated by the telecommunications service provider; and', 'if so, then communicate routing instructions to said STP; and, 'an internal service control point (SCP) in communication with said STP and configured to receive the routing instructions and telephone call request from the SCP via the STP; and', 'connect the telephone call to a toll free telephone that is operated by the telecommunications service provider., 'a soft switch in communication with said local switch, and configured to2. The system according to claim 1 , wherein ...

Methods and devices for interworking of wireless wide area networks and wireless local area networks or wireless personal area networks

Embodiments describe registration in a wireless communication system. A method includes wirelessly transmitting over a WWAN a first registration message from a mobile device, wirelessly transmitting through the WWAN a second registration message to a WLAN access point and receiving at the mobile device access through the WLAN access point. According to another embodiment is a method for constructing a self-configuring ad-hoc network. The method can include receiving a GPS coordinate from a WWAN channel node at a management system and creating an initial topography based at least in part on the GPS coordinate to achieve a network connectivity with diverse routes between a plurality of nodes.

INTELLIGENT END USER DEVICES FOR CLEARINGHOUSE SERVICES IN AN INTERNET TELEPHONY SYSTEM

Clearinghouse services architectures that support the use of end user devices, such as personal computers, Internet Protocol (IP) phones, cable multimedia terminal adapters, and residential gateways, in an Internet telephony system. The innovative architectures include a proxy-based system model, a direct communication model, and a hybrid proxy/direct communication model. A user can operate an “intelligent” end user device. i.e., a device running a client program with knowledge of the architecture particulars, to access a clearinghouse service on an IP network. This enables the user to communicate a telephony call over the IP network and via the combination of a terminating gateway identified by the clearinghouse service and the Public Switched Telephone Network. 13-. (canceled)4. A computer-implemented method for providing clearinghouse services to a client device in an Internet Protocol (IP) telephony system , comprising the steps of:transmitting an authorization request from a proxy server to a clearinghouse service, the clearinghouse service being accessible by the proxy server;transmitting an authorization response from the service point to the proxy, the authorization response comprising the identity of at least one terminating gateway available to complete the IP telephony communication session, and an electronic authorization token generated by the clearinghouse service for each identified terminating gateway, each terminating gateway operating independently such that the proxy server may gain access to a terminating gateway with the electronic authorization token; andselecting a terminating gateway with the proxy server to complete the IP telephony communication session.5. The method of claim 4 , further comprising transmitting an IP telephony communication session set-up request for an IP telephony communication session to the proxy server.6. The method of claim 5 , wherein transmitting an IP telephony communication session set-up request for an IP ...

System and Method for Originating a Call via a Circuit-Switched Network from a User Equipment Device

In one aspect, an application server (AS) node comprises a processor configured to receive a first SIP Invite request from the UE device, the first SIP Invite request having call information which includes a URI of a called party and data indicating that the first SIP Invite request is for a bearer over a circuit-switched (CS) network. An E.164 number is associated with the call information in the first SIP Invite request and a SIP response including the E.164 number is sent to the UE device in response to receiving the first SIP Invite request. When a call setup message including the E.164 number received from the CS network, the received E.164 number is verified if it is valid and if so, the call information is identified based on the E.164 number and is sent to the called party via a second SIP Invite request. 1. A method for an application server (AS) node , the method comprising:receiving a first SIP Invite request from a mobile communication device, the first SIP Invite request having call information which includes a uniform resource indicator (URI) of a called party and data indicating that the first SIP Invite request is for a bearer over a circuit-switched (CS) network;associating an E.164 number with the call information in the first SIP Invite request;sending a SIP response to the mobile communication device in response to receiving the first SIP Invite request, the SIP response including the E.164 number;after sending the SIP response, receiving a call setup message from the CS network, the call setup message having the E.164 number;verifying that the received E.164 number remains valid;identifying the call information based on the E.164 number; andsending, to the called party, a second SIP Invite request that includes the call information.2. The method of claim 1 , further comprising:identifying the URI of the called party in a TARGET field of the first SIP Invite request.313. The method of claim claim 1 , wherein a URI field of the first SIP Invite ...

COMMUNICATIONS SWITCHING ARCHITECTURE

Presently described is a system and method for switching multimedia data communications, including but not limited to Voice over IP (VoIP) telephony, cable TV, digital audio and video. The system utilizes a single, integrated device to provide all PacketCable-compliant functionality, including enhanced user privacy, compliance with CALEA, E911 and other mandated services not available in conventional distributed PacketCable systems. High speed and efficient, low cost operation are provided by means of an optimized data unit encapsulation scheme for internal switching and routing. A proprietary fiber optic backplane and removable optical connectors are used to enable lightspeed internal communications hot-swapping of components. Furthermore, the present system is extensible to all forms of digital data switching and is secure, resistant to Denial of Service attacks, and fault-resilient. 116.-. (canceled)17. A packet-based telephony architecture comprising:a first addressable communication terminal configured for real-time packet-based communication;a second addressable communication terminal configured for real-time packet-based communication, the first and second packet telephone terminals communicatively coupled with a packet network;a switch in the packet network, wherein said packet-based telephony architecture is configured such that when a real-time communication session is established between said addressable communication terminals, each said addressable communication terminal is not provided the address of the other said addressable communication terminal, and the communication from each said addressable communication terminal during said session is instead directed to an address of said switch.18. The architecture of in which the first packet telephone terminal comprises a first Multimedia Terminal Adapter (MTA) claim 17 , and in which the second packet telephone terminal comprises a second MTA.19. The architecture of in which the switch provides its own ...

System and Method for Dynamic Telephony Resource Allocation Between Premise and Hosted Facilities

A population of networked Application Gateway Centers or voice centers provides telephony resources. The telephony application for a call number is typically created by a user in XML (Extended Markup Language) with predefined telephony XML tags and deployed on a website. A voice center provides facility for retrieving the associated XML application from its website and processing the call accordingly. The individual voice centers are either operated at a hosted facility or at a customer's premise. Provisioning Management Servers help to allocate telephony resources among the voice centers. This is accomplished by suitably updating a voice center directory. In this way, the original capacity at a premise, predetermined by the hardware installed, can be adjusted up or down. If the premise is under capacity, it can be supplemented by that from a hosted facility. If the premise has surplus capacity, it can be reallocated for use by others outside the premise. 1. In a networked computer telephony system , a method of processing telephone calls for a subscriber , comprising:providing a first set of application gateway centers of a hosted facility and a predetermined second set of application gateway centers at a premise of the subscriber for processing telephone calls;providing a directory for routing a telephone call to an application gateway center among either the first set or the second set, the directory designating an application gateway center as a function of the telephone number, predefined routing rules and operating capacities of individual application gateway centers;providing a user interface for a user to update user-specified routing rules in the directory; androuting the telephone call to an application gateway center according to the directory.2. The method as in claim 1 , wherein:the user is a subscriber, and routing rules configurable by the subscriber include telephone calls to be preferentially processed by an application gateway center on premise ...

PERSONAL DESKTOP ROUTER

A client-server telephone call router system as part of a customer premises system has a client-server router adapted to execute on a telephony switch, such as a public branch exchange (PBX) or other telephony switch, or on a processor connected by CTI link to a telephony switch. The telephony switch or processor executing the router is connected to a local area network (LAN) that also interconnects computer workstations proximate to telephones connected to the telephony switch. Client user interface applications run on the computer workstations, allowing clients to edit routing rules for the router, which has a list of routing rules keyed to users and workstations of the customer premises system. The editing rules are kept by the router in portions dedicated to individual users. With this system a user can edit at a workstation on the LAN his\her own routing rules, and transmit the edits to the client-server router where the rules will be followed to route calls for that user and protocol. 1(a) receiving a telephone call at a telephony switch connected to telephones at individual telephones at the customer premises;(b) providing identifying data for the telephone call to a client-server router;(c) determining a routing for the call at one of the computer workstations by use of a client interface, and transmitting the routing determination to the client-server router via the LAN connection ; and(d) routing the telephone call by the client-server router according to the determination transmitted to the client-server router via the LAN connection from the client interface application.. A method for routing telephone calls at customer premises having a telephony switch, individual telephones connected to the telephony switch, and computer workstations including video display units (VDUs) connected on a local area network (LAN) also coupled to the telephony switch, wherein individual ones of the computer workstations are located proximate individual ones of the ...

Systems and methods of introducing beneficial communication latency in contact handling systems

Methods and apparatus for improving a contact with a customer in a contact handling system by introducing a slower expected communication rate to the contact are disclosed. The method can include communication via a first media having a first communication rate. The method can include ceasing communication via the first media. The method can include beginning communication via a second media having a second communication rate slower than the first communication rate. The method can further include ceasing communication via the second media and beginning communication via a third media having a third communication rate faster than the second communication rate.

Systems and methods for ip and voip device location determination

A method and system for precise position determination of general Internet Protocol (IP) network-connected devices. A method enables use of remote intelligence located at strategic network points to distribute relevant assistance data to IP devices with embedded receivers. Assistance is tailored to provide physical timing, frequency and real time signal status data using general broadband communication protocols. Relevant assistance data enables several complementary forms of signal processing gain critical to acquire and measure weakened or distorted in-building Global Navigation Satellite Services (GNSS) signals and to ultimately extract corresponding pseudo-range time components. A method to assemble sets of GNSS measurements that are observed over long periods of time while using standard satellite navigation methods, and once compiled, convert using standard methods each pseudo-range into usable path distances used to calculate a precise geographic position to a known degree of accuracy.

A telephone outlet for implementing a local area network over telephone lines and a local area network using such outlets

An outlet for coupling at least one data unit to digital data carried over wiring that simultaneously carry a packet-based serial digital data signal and a power signal over the same conductors. The outlet includes: a wiring connector for connecting to the wiring; a transceiver coupled to the wiring connector for transmitting and receiving packet-based serial digital data over the wiring; a LAN connector coupled to the transceiver for bi-directional packet-based data communication with at least one data unit; a bridge or a router coupled between the transceiver and the LAN connector for passing data bi-directionally between the at least one data unit and the wiring; and a single enclosure housing the above-mentioned components. The enclosure is mountable into a standard wall outlet receptacle or wall outlet opening, and the transceiver and the bridge or router are coupled to the wiring connector to be powered from the power signal.

Active speaker identification

Procedures for identifying clients in an audio event are described. In an example, a media server may order clients providing audio based on the input level. An identifier may be associated with the client for identifying the client providing input within the event. The ordered clients may be included in a list which may be inserted into a packet header carrying the audio content.

VOICE COMMUNICATION APPARATUS FOR INTERMITTENTLY DISCARDING PACKETS

In a voice communication device for use on an IP network, a transmitter encodes voice information and assembles RTP packets with the encoded voice information inserted in a payload of the RTP packets to transmit a stream of the assembled RTP packets. RPT packets may intermittently be discarded at a rate of discarding from a stream of the RTP packets to be transmitted. The remaining packets will be transmitted to a destination. 1. A voice communication apparatus for use on an IP (Internet Protocol) network , comprising a transmitter encoding voice information and assembling RTP (Real-time Transport Protocol) packets with the encoded voice information inserted in a payload of the RTP packets to transmit a stream of the assembled RTP packets , wherein said transmitter includes:a packet generator generating the RTP packets to transmit the stream of the generated RTP packets; anda packet discarder intermittently discarding an RTP packet from the stream of the generated RTP packets to be transmitted, and sending the remaining RTP packets.2. The apparatus in accordance with claim 1 , wherein said packet discarder further includes a rate controller controlling a rate of intermittently discarding an RTP packet.3. The apparatus in accordance with claim 2 , wherein said rate controller sets the rate of intermittently discarding an RTP packet to a predetermined value.4. The apparatus in accordance with claim 2 , wherein said rate controller variably sets the rate of intermittently discarding an RTP packet.5. The apparatus in accordance with claim 1 , wherein said transmitter further includes an encoder encoding the voice information claim 1 ,said encoder including a power calculator calculating power of a frame of the voice information to be encoded,said packet discarder comparing the calculated power of the RTP packets to discard an RTP packet having a lowest power.6. The apparatus in accordance with claim 1 , wherein said transmitter further includes a correlation detector ...

Voice over ip method for developing interactive voice response system

A personal interactive voice response service node includes a memory that stores executable instructions, and a processor that executes the executable instructions. The personal interactive voice response service node accepts personalized instructions to define interactive voice response service node prompts on behalf of an individual assigned a communications address serviced by the personal interactive voice response service node. The personal interactive voice response service node accepts personalized instructions to define actions to take in response to selections of corresponding interactive voice response service node prompts. When executed by the processor, the executable instructions cause the personal interactive voice response service node to identify a selection of an interactive voice response service node prompt and execute an action associated with the selection identified.

Dynamic Application Integration Associated with Hosted VoIP PBX Using Client-Side Integration Proxy

A system for dynamically integrating and synchronizing a plurality of software applications of an end user as part of a telephonic communication between the end user and a third party managed and handled by a hosted VoIP PBX includes a client side integration proxy having (i) an API for communicating and exchanging data with the software applications and (ii) memory cache dedicated to storing information about the telephonic communication handled and managed by a hosted VoIP PBX, wherein the client side integration proxy assigns a record in the memory cache for storing information about the telephonic communication and enables the software applications periodically to access and dynamically update, modify, or add to the record as the information about the telephonic communication is updated or changed by other software applications and by the end user. 2. The system of claim 1 , wherein the client side integration proxy detects initiation of the telephonic communication based on data obtained from the hosted VoIP PBX through the API of the hosted VoIP PBX.3. The system of claim 2 , wherein the information about the telephonic communication initially stored in the assigned portion of the memory cache includes the data obtained from the hosted VoIP PBX.4. The system of claim 1 , wherein the information about the telephonic communication includes one or more of the following: a phone number of the third party claim 1 , a name of the third party claim 1 , an IP address of the device being used by the third party for the telephonic communication claim 1 , a phone number or extension of the end user claim 1 , a name of the end user claim 1 , an IP address of the client device of the end user claim 1 , an indication whether a portion or all of the telephonic communication is encrypted claim 1 , an indication whether a portion or all of the telephonic communication is using a secured communication line claim 1 , an indication whether the telephonic communication is inbound ...

COMMUNICATION NETWORK SYSTEM, CALLING TERMINAL AND VOICE CALL ESTABLISHING METHOD THEREOF

A communication network system, a calling terminal and a voice call establishing method thereof are provided. The communication network system comprises a called terminal, the calling terminal and a session initiation protocol (SIP) server. The calling terminal generates and transmits an invite message including IPv4 connection information and IPv6 connection information of the calling terminal. The SIP server is communicatively connected to the calling terminal and the called terminal. The SIP server receives the invite message from the calling terminal and forwards the invite message to the called terminal. The called terminal establishes a voice call with the calling terminal according to one of the IPv4 connection information and the IPv6 connection information of the calling terminal. 1. A communication network system , comprising:a called terminal;a calling terminal, being configured to generate and transmit an invite message including IPv4 (Internet Protocol version 4) connection information and IPv6 (Internet Protocol version 6) connection information of the calling terminal; andan SIP (Session Initiation Protocol) server communicatively connected to the called terminal and the calling terminal, being configured to receive the invite message from the calling terminal and forward the invite message to the called terminal;wherein the called terminal establishes a voice call with the calling terminal according to one of the IPv4 connection information and the IPv6 connection information of the calling terminal.2. The communication network system as claimed in claim 1 , wherein:the called terminal generates and transmits a ringing message and an OK message to the SIP server according to the IPv4 connection information and the IPv6 connection information of the calling terminal respectively after receiving the invite message, the ringing message or the OK message including IPv4 connection information of the called terminal;the SIP server forwards the ringing ...

Method, Computer-Readable Medium, and Apparatus for Providing Different Services to Different Users of an Aggregate Endpoint in an Internet Protocol Multimedia Subsystem (IMS) Network

Different services are provided to different users or groups of users of an aggregate endpoint in an internet protocol multimedia subsystem (IMS) network. The different users or groups of users are differentiated based on service profiles designated by different PUIDs for the different users or groups of users of each of the users or groups of users. Different services for transmission and/or receipt of packets for the different users or groups of users of the aggregate endpoint are provided, depending upon the differentiation between the different users or groups of users. 1. A method comprising:differentiating, by a call session server within an internet protocol multimedia subsystem network, between different users of an aggregate endpoint based on an identifier for each of the users, wherein the identifier is associated with a priority value assigned to each device associated with each of the users and indicates a service profile including contact information, device address, and device capabilities for each device associated with each of the users;obtaining, by the call session server, the indicated service profile from a home subscriber server, the service profile further including filter criteria designating conditions under which particular application servers should be involved in transmission and/or receipt of packets for each of the users;determining, by the call session server, different services to provide for the different users of the aggregate endpoint depending upon the differentiating between the different users based on the identifier and the indicated service profile; anddetermining, by the call session server, routing for transmission and/or receipt of packets for the different users depending upon the differentiating between the different users based on the identifier, the indicated service profile, and the associated priority value, including evaluating the conditions designated by the filter criteria included in the indicated service profile ...

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

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).

Voicemail Proxy Server

Methods and systems directed to retrieving voicemails from a voice mailbox server. In particular embodiments, a mailbox server may receive a notification from the voice mailbox server and connect to the voice mailbox server providing the credentials of a client device, and pull new voicemails for storage at the mailbox server. In particular embodiments, mailbox server periodically polls voice mailbox server for new voicemail messages. In particular embodiments, a client device polls or receives notification of new voicemails, and transmits a downloaded message to the mailbox server.

Intelligent handoffs for enhancing or avoiding dropped and interrupted communication sessions

Systems and methods for intelligently enhancing communication sessions or avoiding dropped and interrupted communication sessions between or among communication devices. Characteristics of a communication link are monitored and are compared to intelligent communication session data to predictively determine whether a handoff may be needed or whether multi-streaming should be invoked or terminated. Intelligent communication session data may be monitored, recorded and/or stored by a handoff service, a bridging service and/or a multi-streaming service, as well as applications executed on communication devices or other network devices. Intelligent communication session data may be any data relating to a communication session, including detected signal strengths, available networks, protocol and buffer statistics and analysis, environmental and/or geographical factors, the performance of access points and other network components, past interactions between or among communication devices, and access points and other network components, context of conversations during voice calls.

Retrieval and presentation of network service results for mobile device using a multimodal browser

A method of obtaining information using a mobile device can include receiving a request including speech data from the mobile device, and querying a network service using query information extracted from the speech data, whereby search results are received from the network service. The search results can be formatted for presentation on a display of the mobile device. The search results further can be sent, along with a voice grammar generated from the search results, to the mobile device. The mobile device then can render the search results.

Voip analog telephone system with connection to home monitoring system

A multi-port VoIP telecommunications system that allows the user to gain access to telephone connectivity through the Internet by connecting directly to the Internet or by connecting to the Internet through the existing Internet connection of a computer or cell phone device. The present system includes an Ethernet port, a Wi-Fi receiver to facilitate the transmission and receipt of Internet protocol signals wirelessly, a USB plug connectable to the ATA, connectivity to a home monitoring network and connectivity to Bluetooth devices.

Solutions for Voice Over Internet Protocol (VoIP) 911 Location Services

An E-9-1-1 voice-over-IP (VoIP) solution is provided wherein a 911 call from a mobile VoIP device is routed directly to the correct Public Safety Answer Point (PSAP) via dedicated trunks, together with correct location information and call-back number. VoIP gateways are implemented locally, at least one per LATA, and accept VoIP packetized data inbound, and convert it to standard wireline voice calls. Calls are routed to an IP address at the VoIP gateway, which then egresses the call to a voice port at a selective router. Mid-call updating of location of a moving VoIP terminal is provided to a PSAP. The location of the VoIP is validated using HTTP based protocol by pushing location information to a VoIP location server, and comparing it against a geographic location database to confirm that a contained street address is valid. 1. A method of providing mid-call updating of a location of a VoIP terminal , comprising:allowing a VoIP call to be established with said VoIP terminal;requesting updated location information relevant to movement of said VoIP terminal since said call was established; andtransmitting updated location information relevant to said movement of said VoIP terminal since said call was established.2. The method of providing mid-call updating of a location of a VoIP terminal according to claim 1 , wherein:said established call is an established emergency call to a Public Safety Access Point (PSAP).3. The method of providing mid-call updating of a location of a VoIP terminal according to claim 2 , wherein:said updated location information is requested by said PSAP.4. The method of providing mid-call updating of a location of a VoIP terminal according to claim 1 , wherein:said updated location information is transmitted from said VoIP terminal.5. The method of providing mid-call updating of a location of a VoIP terminal according to claim 1 , wherein:said updated location information is determined by said VoIP terminal.6. The method of providing mid-call ...

Method and system for transferring control of a conference bridge

A conference calling system enables transfer of control of a conference bridge. The transferor terminal has an option to transfer control of the conference bridge to a transferee terminal. The transferee terminal has an option to accept or reject control of the conference bridge. When the transferee terminal accepts control prior to the transferor terminal severing control, the control of the conference bridge is transferred to the transferee terminal. However, when the transferee terminal rejects or does not accept within a predetermined time the control of the conference bridge, the transferor terminal is provided an option to transfer control of the conference bridge to alternate terminals, one terminal at a time, until an alternate terminal accepts control or the transferor terminal severs control of the conference bridge. As long as a terminal has control of the conference bridge, the conference bridge is not severed.

Distributed hardware/software system for managing agent status in a communication center

An agent presence application for monitoring target agent resources and rendering agent states to subscribing applications is provided. The presence application has at least one first portion for collecting data regarding states of activity of the target agent resources, and at least one second portion for integrating the data and rendering the agent states to the to subscribing applications.

Enrichment, management of multimedia content and setting up of a communication according to enriched multimedia content

An enrichment of multimedia content, management of multimedia content and setting up of a communication according to enriched multimedia content. A multimedia content enrichment method comprising an association of at least one identifier of a first user with multimedia content during management of said content by a communication terminal of said first user, said association allowing a second user having access to said multimedia content to request setting up of a communication between a communication terminal of said second user and a communication terminal of said first user by using one of said at least one identifier(s) of said first user.

Communication Session Processing

Measures for use in processing communication sessions in a telecommunications network are provided. Each communication session has a signalling path spanning a plurality of devices including one or more intermediate network devices and at least two endpoint devices, the signalling path comprising a plurality of signalling segments, each segment being between two devices in the plurality of devices. A first signalling message, comprising a first identifier associated with the communication session, is received via a first signalling segment for a communication session. At least part of the first identifier is transformed using a deterministic encryption algorithm to generate a second identifier. A second signalling message, comprising the second identifier, is transmitted via a second signalling segment for the communication session to associate the second identifier with the communication session. 1. A method of processing communication sessions in a telecommunications network , each communication session having a signalling path spanning a plurality of devices including one or more intermediate network devices and at least two endpoint devices , the signalling path comprising a plurality of signalling segments , each segment being between two devices in the plurality of devices , the method comprising:receiving, via a first signalling segment for a communication session, a first signalling message comprising a first identifier associated with the communication session;transforming at least part of the first identifier using a deterministic encryption algorithm to generate a second identifier; andtransmitting, via a second signalling segment for the communication session, a second signalling message comprising the second identifier to associate the second identifier with the communication session.2. The method according to claim 1 , wherein the first signalling message comprises a first communication session setup signalling message for the communication session and ...

METHODS, SYSTEMS, AND COMPUTER PROGRAM PRODUCTS FOR PROVIDING INTER-CARRIER IP-BASED CONNECTIONS USING A COMMON TELEPHONE NUMBER MAPPING ARCHITECTURE

A system includes a network entry point configured to receive and process internet protocol (IP) based connection requests originating from outside the communications network, and a telephone number mapping (ENUM) database configured to store a uniform resource identifier (URI) of the network entry point and to respond to a request for routing information for a subscriber terminal within the communications network received from a remote entity outside the communications network by providing a naming authority pointer resource (NAPTR) record including the URI of the network entry point to the remote entity in response to the request for routing information. 1. A system comprising:a network entry point to receive and process internet protocol based connection requests originating from outside a communications network;a tier 2 telephone number mapping database that is accessible from outside the communications network and that is to store a uniform resource identifier of the network entry point and to respond to a request for routing information for a subscriber terminal within the communications network received from a remote entity outside the communications network by outputting a naming authority pointer resource record including the uniform resource identifier of the network entry point to the remote entity in response to the request for routing information wherein the uniform resource identifier of the network entry point is different than a uniform resource identifier of the subscriber terminal; anda private telephone number mapping database to store internal routing information for subscriber terminals in the communications network and to respond to requests for routing information from entities within the communications network.2. The system of claim 1 , further comprising:an internet protocol multimedia subsystem core to manage voice over internet protocol communications sessions;wherein the network entry point is to forward the incoming call setup request to ...

System and method for providing automatic determination of a call type in telephony services over a data network

A system and method for providing automatic determination of a call type in a voice over Internet Protocol data network is provided. A 10-digit telephone number for making a telephone call is entered and a look up of the telephone number is performed in a database comprising at least known long distance numbers. If no match exists, the telephone number is transmitted over the data network to a local Call Management Server. If the local call is not successful, the number is automatically transmitted with a pre-pended “I” to a Long Distance Call Management Server. Advantageously, a VoIP system is provided for automatically determining that a number is a long distance call, upon which it will automatically pre-pend a leading “1” to the number and re-transmit the resultant 11 digit number to the appropriate Long Distance Call Management Server.

Systems and methods for enabling access to emergency services

Systems and methods for providing a user with access to emergency services are described. A cable modem/multimedia terminal adapter employing voice over internet protocol (VoIP) can be configured to provide access to emergency services during periods of unavailability of a connection to a network due to software upgrades of the cable modem/multimedia terminal adapter or a VoIP handset, a power outage or a malfunction of the cable modem/multi-media terminal adapter or the VoIP handset. Access to emergency services can be provided through the use of a dedicated button on the cable modem/multimedia terminal adapter and the use of dedicated hardware in the cable modem/multimedia terminal adapter that is configured to initiate operation when a connection to the network is unavailable.