Abstract:
A VoIP gateway apparatus is used to provide an IP centrex service to accommodated devices. The VoIP gateway apparatus (1) assigns, to a call originated from an accommodated device (PBX2), a telephone number for which the number of simultaneous call connections has not reached a simultaneous connection tolerance number, and sends the call to an IP centrex network (5) with the assigned telephone number used as a calling number. The VoIP gateway apparatus (1) includes a number-of-calls management part (105) and a SIP processing part (104). The number-of-calls management part (105) manages the number of the current simultaneous call connections for each of the telephone numbers as assigned to the apparatus. When a call is received from an accommodated device, a SIP processing part (104) selects, among the telephone numbers as assigned to the apparatus, a telephone number, for which the number of the current simultaneous call connections as managed by the number-of-calls management part (105) is smaller than the predetermined simultaneous connection tolerance number, and then transmits, to the IP centrex network (5), a SIP packet of a call control message to which the selected telephone number has been added as the calling number.
Abstract:
A voice communication system, which is connected to a LAN to which communication terminals are connected and to a public network to which telephones are connected, is provided with a communication server between the LAN and public network having different protocols from each other. The communication server enables a voice communication between a telephone on the public network and a communication terminal connected to the LAN by performing processing similar to that for a voice communication between two communication terminals connected to the LAN. The communication server determines whether an address of the other party inputted by a user is a communication terminal address or a telephone number, and transmits a voice communication request to a communication terminal of the other party when the address is a communication terminal address. When the address is a telephone number, the user acquires the communication terminal address of the communication server, and transmits a voice communication request to the communication server. Thereafter, the voice communication processing is performed through the communication server.
Abstract:
According to one embodiment of the invention, a method for conducting a conference call between two or more participants is provided. The method includes receiving an indication of a request for text from a participant. The method also includes converting, in response to the indication, any speech of the other participants of the conference call into text. The method also includes sending the text to a device associated with the participant who requested test. The device is operable to display the text.
Abstract:
A packet switch apparatus including a plurality of connecting points, each being coupled to one of a plurality of data processing units, and apparatus for receiving a first packet including identification information of a first data processing unit via a first connecting point. The invention further includes a storage for storing correspondence between the first data processing unit and the first connecting point based on the identification information. Apparatus, responsive to reception of a second packet from a second data processing unit to be transmitted to the first data processing unit, transmits the second packet to the first data processing unit via the first connecting point based on the correspondence stored in the storage.
Abstract:
The present invention provides a presence system capable of monitoring state information derived from a plurality of sources over any number of disparate networks. The sources of state information are devices, which are frequently used by a user throughout a normal day and configured to provide state information to the presence system. The sources monitor normal user interactions and automatically provide corresponding state information to the presence system without requiring the user to enter or otherwise provide information bearing on their status or availability. Based on a profile provided by the user, the presence system evaluates the state information from one or more sources to create presence information to deliver to subscribers. The state information bears on the presence or availability of the user and may take many forms. The presence information may range from complex analysis of state information from many devices to simply the states of selected devices.
Abstract:
Disclosed is an apparatus and a method for placing calls between legacy terminals and IP terminals in an IP-PBX system. Features that were once only used in traditional PBX telephone systems, like group ring services and station group services are now made available for IP terminals in an IP-PBX system. Methods of providing station group service and ring group service to both legacy terminals and IP terminals in an IP-PBX system is disclosed.
Abstract:
Methods and apparatus for automated provisioning of voice over internet protocol (VoIP) gateways, proxy servers, session border controllers, softswitches and/or softswitch/proxy servers are disclosed. A disclosed method comprises performing a query of a database containing records specifying a communication path between a first communication network and a second communications network via a communication device, translating a result of the database query into a configuration parameter for the communication device, and configuring the communications device with the configuration parameter.
Abstract:
A system and method for improving the intelligibility of a moderator during a multi-party communication session includes receiving a plurality of participant voice streams from a plurality of respective conference participants. An incoming moderator voice stream may be received from a moderator. The plurality of participant voice streams and the moderator voice stream are transmitted such that the intelligibility of the moderator voice stream is improved relative to at least one of the participant voice streams.
Abstract:
A system and method for providing packet-switched telephony service. The system provides call control, signaling, and/or delivery of voice, video, and other media in substantially real time. One embodiment of the system includes a call client application on a user device, and a call server located at a packet-switched telephony service provider. The call server is preferably operable to communicate with the call client in a non-native protocol and with the gateway in a native protocol.
Abstract:
Data for identifying users is detected by a Server, which is preferably connected to the Internet, in order to connect two users is provided. The server establishes a first signaling connection to user A and a second signaling connection, to user B by taking into account said data, whereupon the server combines the two signaling connections to a continuous signaling connection, allowing a continuous signaling connection to be established in which fees are charged by server S rather than by user A, whereby alternative options become available for charging for a traffic channel connection allocated to the continuous signaling connection.