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:
A method, an apparatus, and a computer program product thereof for enabling an Internet extension to ring a conventional extension are disclosed. The apparatus comprises an SIP proxy and an RTP relay. The SIP proxy receives a calling request from the Internet extension, substitutes an exchange number for a conventional extension number comprised in the calling request, and transmits the substituted calling request to a gateway, so that the substituted calling request can be transmitted to a voice automatic machine via a telephone exchange. The RTP relay transmits the conventional extension number in DTMF format to the gateway. Thus, this invention enables the Internet extension to ring the conventional extension with lower hardware costs.
Abstract:
A method and system of routing telephone calls from a calling party to a called party including placing a call to a called party, accessing context information associated with the called party, permitting the calling party to opt to continue with the call or terminate the call based on the accessed context information, wherein continuing with the call comprises connecting the call to the called party or selecting a redirection option with the call.
Abstract:
Techniques for providing a gateway transfer mechanism are disclosed. In one particular exemplary embodiment, the techniques may be realized as a method, comprising identifying, at a gateway, a call to be routed to a first location, determining whether to transfer the call to a second location, in the event the call is to be transferred to the second location, transferring the call, wherein transferring the call comprises ending a process to route the call to the first location, generating transfer information for the second location, and transferring the call to the second location using the generated transfer information.
Abstract:
Methods, systems, and computer program products for providing caller ID and call waiting and for switching or toggling between active and waiting calls using SIP are disclosed. According to one method, a first call is established between a first phone and a SIP termination. The first call is established using the first media connection between the SIP termination and a media gateway and a second media connection between the media gateway and the first phone. During the first call, signaling for establishing a second call to SIP termination is received. In response to the signaling, caller ID information for the second call is communicated to the SIP termination. A hook flash is received from the SIP termination. In response to the hook flash, the SIP termination is connected to the second phone using the first media connection and a third media connection between the media gateway and the second phone.
Abstract:
There are provided an IP telephone system and method for establishing a connection to the IP network 4 via a public line network and performing talking by transmitting/receiving IP packet data into which audio data has been converted. In a location server 3 arranged on a PSTN 5 or the IP network 4, a public line telephone number and an IP telephone number which are attached to the line connected to a telephone terminal 1a and a terminal identifier identifying the telephone terminal 1a are registered while correlating them to each other. When an IVR 8 is called, the public line telephone number, the IP telephone number of the telephone terminal 1a and/or the terminal identifier is acquired from the telephone terminal 1a and the registration content in the location server 3 is modified via a registrar system 7.
Abstract:
A transfer unit transfers a message between a network and an external terminal. An input unit inputs a user ID for identifying a user. A generating unit generates a registration message requesting a registration of address information of the user. A transmitting unit transmits the registration message to a server. A receiving unit receives a response message including registration information and connection information from the server. When the connection information indicates a permission of a connection of the external terminal to the network, a control unit controls the transfer unit to transfer the message between the network and the external terminal.
Abstract:
A wireless communications device is a hub for a conference call between a user of the device, a first remote party, and a second remote party. The wireless communications device communicates with the first remote party over a first connection and with the second remote party over a second concurrently active connection. A circuit in the wireless communications device transcodes voice data received over the first and second connections so that the voice data may be shared between the parties.
Abstract:
A cable infrastructure includes a plurality of cable modem circuits communicatively coupled to a cable network and associated with a corresponding one of a plurality of subscribers. Each of a plurality of telephones has both a POTS (Plain Old Telephone System) interface and a unique telephone number and associates with a corresponding one of the plurality of subscribers. Each of a plurality of interface circuits couples one of the plurality of cable modem circuits with a corresponding one of the plurality of telephones via the POTS interface. A head end system communicatively couples to the cable network, the public switched telephony network, and the Internet network and supports address mapping that enables communication exchanges between one of the plurality of telephones and an Internet telephony device. The address mapping also enables communication exchanges between telephones serviced by differing head ends via an Internet pathway that is independent of the public switched telephony network.
Abstract:
Providing a single number presentation to the party called by a dual-mode phone. The operation of the cellular side of a dual-mode phone is altered such that when the user attempts to place an outgoing call using the cellular phone, the call is redirected to a preprogrammed incoming phone number associated with the enterprise. When the enterprise PBX answers this call, the dual-mode phone transmits the desired number to the enterprise PBX. The enterprise PBX then places the call to the desired number, and in the process transmits the caller-id information assigned to the dual-mode phone.