An intelligent interface responds to text messages, which in a particular embodiment are addressed to a landline phone number. An IP network interface receives requests including a source phone number, message content and the destination phone number. The intelligent interface includes access to a serialized intent file created by a vectorizer. An intent classifier determines one intent from the serialized intent file based on the received message content. A response is generated according to the one intent. A reply generator generates a reply including the source phone number as a destination, the landline phone number as a source and the response.

Methods for handling traffic between a first network including a virtual communication platform and a second network, such as a circuit-switched network, include inserting customer identifiers into data packets. For traffic sent to the second network, the packets with the identifiers are transmitted to a trunking service that uses the identifier to determine one or more applicable service attributes for handling the data contained within the packets. For data traffic received from the second network, the virtual communication platform uses the identifiers to determine a destination, such as an endpoint device, for the traffic within the first network.

An intelligent interface responds to text messages, which in a particular embodiment are addressed to a landline phone number. An IP network interface receives requests including a source phone number, message content and the destination phone number. The intelligent interface includes access to a serialized intent file created by a vectorizer. An intent classifier determines one intent from the serialized intent file based on the received message content. A response is generated according to the one intent. A reply generator generates a reply including the source phone number as a destination, the landline phone number as a source and the response.

A system integrating a web browser and telephone is provided. A user enters, and the web browser receives, an input specifying a request to receive an audio output from the telephone. The web browser generates an audio stream which is communicated to the WebRTC gateway. The WebRTC gateway converts the audio stream from a first format into a second format. The WebRTC gateway communicates the converted audio stream to the SIP gateway, which forwards the audio stream to the telephone switch along with information identifying the telephone number to which the audio stream is directed. The telephone switch communicates the audio stream to the telephone where the audio stream is converted to be played using the speaker of the telephone.

There is described a gateway system comprising: a local-call device for receiving and sending phone call signals from and to a telephone device located in a same telecommunication zone; a VoIP device, distinct from the local-call device, for receiving and sending VoIP call signals from and to a remote and distinct VoIP device located in another distinct telecommunication zone; and a voice transmission apparatus comprises a set of two acoustically distinct chambers. Those comprise a first chamber comprising a first microphone and a second speaker; and a second chamber comprising a second microphone and a first speaker. The first microphone and the first speaker are in communication with the local-call device only and the second microphone and the second speaker are in communication with the VoIP device of the gateway system only.

An audio bridge unit is provided for connecting a conventional analog page/party station in an analog page/party system to IP-based page/party stations that employ multicast Voice over Internet Protocol (VoIP) technology to support page/party communications between analog page/party stations and the VoIP page/party stations. The audio bridge unit converts page and/or party audio line(s) to multicast channel(s). The audio bridge unit analog interfaces for its respective page and/or party lines, and each analog interface is assigned to an independent multicast channel on which it will transmit and receive full duplex audio between an analog page/party station and one or more IP-based page/party stations. The audio bridge unit provides the ability to configure the multicast group, port and Time to Live (TTL) setting for each of the analog channels associated with the page and/or party lines.

An embodiment of the present disclosure provides an intercom system including a cloud server, and a relay gateway. The cloud server is configured to receive a trigger signal from a visitor door station, and to send a notification when the cloud server receives the trigger signal from the visitor door station. In one embodiment, the relay gateway is configured to receive the notification from the cloud server and to send the notification to one of a plurality of local intercom stations via one of plurality of hard wires, the hard wires being connected to the local intercom stations, respectively. In one embodiment, the cloud server is configured to send the notification to a remote intercom device that corresponds to the one of the plurality of the local intercom stations.

Methods for handling traffic between a first network including a virtual communication platform and a second network, such as a circuit-switched network, include inserting customer identifiers into data packets. For traffic sent to the second network, the packets with the identifiers are transmitted to a trunking service that uses the identifier to determine one or more applicable service attributes for handling the data contained within the packets. For data traffic received from the second network, the virtual communication platform uses the identifiers to determine a destination, such as an endpoint device, for the traffic within the first network.

The present invention pertains to a method and system for preventing unauthorized access via signal interception and hacking to a user's secure mobile device. One embodiment of the system further comprises an encryption server in communication with the secure mobile device, a clear server in communication with a clear mobile device, and a termination gateway in connection with secure and clear POTS phones on the PSTN. The termination gateway communicates with the clear and encryption servers by IP tunneling. The system enables universal access between secure and non-secure packet-switched phone lines, operating via the Internet, and clear and secure circuit-switched phone lines operating on the PSTN.

A switch device for use in a gateway equipment and a gateway equipment comprising the switch device are suggested. The suggested switch comprises a Foreign Exchange Station (FXS) port configured to be connected to a telephone set, a Foreign Exchange Office (FXO) port connected to communicate analog voice signals with an external network, and a Voice over Internet Protocol (VOIP) unit configured to communicate digital data signals with the external network and enable a VOIP call using the digital data signals. The switch device comprises a switch configured to, when the gateway equipment is power down, connect the FXS port to the FXO port. The switch device also comprises a power unit configured to generate a work voltage from the analog voice signals at the FXO port to provide a power supply to the switch.