Apparatus and methods concerning routing of calls in an IPBX are disclosed. As an example, one apparatus includes an IPBX server configured to route VoIP calls for a plurality of end-users having respective VoIP-enabled devices communicatively coupled to the IPBX via a network. A user-activity detection circuit is communicatively coupled to the IPBX server. The user-activity detection circuit is configured to receive activity data indicating activity of end-users in a virtual office environment and receive event messages indicating statuses of pending calls in the IPBX server. The user-activity detection circuit determines user-activity statuses of the end-users based on received activity data and event messages. A computing server is communicatively coupled to the user-activity detection circuit and is configured to provide a set of services for one or more of the plurality of users as a function of the determined user-activity statuses of the end-user.

When a first user intends a call with a second user via a packet switched communication service (e.g. VoIP), but the second user terminal or second user is currently unavailable to answer a call through the packet-switched service, then a call intent notification is sent to the second user terminal via an out-of-band communication channel, the call-intent notification notifying the second user about the intended call. Includes in the notification is a gateway PSTN number enabling the second user terminal to establish a hybrid PSTN call with the first user terminal via a connection between the second user terminal and a gateway over a PSTN network, and via a connection between the gateway and the first user terminal over the packet-switched network using the packet-switched communication service.

An unitary electronic speaker device and method for rendering digital audio data are provided. A list comprising an ordered sequence of audio item indicators indicating a plurality of audio items stored on at least one other device. is received in an embodiment. In one embodiment, the unitary electronic speaker device does not have any storage space other than the memory, and uses the hard disk embodied in at least one other device to supply the digital audio data. The audio data is received and rendered at the unitary electronic speaker device in according to the sequence defined by the list. Any number of the unitary electronic speaker devices may be connected in a network with the at least one other device. In one embodiment a plurality of unitary electronic speaker devices are connected through a LAN using Ethernet.

A method and a system for establishing a communication session between a first communication device in a main communication network and a second communication device in a local communication network comprising a wireless network, the second communication device being located in the vicinity of a mobile device, the first communication device contacting an external phone number, wherein the external phone number is assigned to a unique identifier of the mobile device. The method comprises (i) the mobile device sending the unique identifier detectable on the wireless network; (ii) providing the unique identifier of the mobile device) and a local communication identifier to a service provider adapter; (iii) determining the location of the mobile device, associating the mobile device to the second communication device in dependence of its location.

Apparatus and methods concerning routing of calls in an IPBX are disclosed. As an example, one apparatus includes an IPBX server configured to route VoIP calls for a plurality of end-users having respective VoIP-enabled devices communicatively coupled to the IPBX via a network. A user-activity detection circuit is communicatively coupled to the IPBX server. The user-activity detection circuit is configured to receive activity data indicating activity of end-users in a virtual office environment and receive event messages indicating statuses of pending calls in the IPBX server. The user-activity detection circuit determines user-activity statuses of the end-users based on received activity data and event messages. A computing server is communicatively coupled to the user-activity detection circuit and is configured to provide a set of services for one or more of the plurality of users as a function of the determined user-activity statuses of the end-user.

When a first user intends a call with a second user via a packet switched communication service (e.g. VoIP), but the second user terminal or second user is currently unavailable to answer a call through the packet-switched service, then a call intent notification is sent to the second user terminal via an out-of-band communication channel, the call-intent notification notifying the second user about the intended call. Includes in the notification is a gateway PSTN number enabling the second user terminal to establish a hybrid PSTN call with the first user terminal via a connection between the second user terminal and a gateway over a PSTN network, and via a connection between the gateway and the first user terminal over the packet-switched network using the packet-switched communication service.

Apparatus and methods concerning call routing are disclosed. As an example, one apparatus includes a circuitry configured to route calls, as data communications (e.g., between a cellular network or PSTN and a plurality of user communication devices) on a broadband network. In response to receiving a first call notification from one of a plurality of mobile devices, a processing circuit of the apparatus caches the call notification in a first database. A first call notification indicates a source number and a destination number. In response to a first incoming call from the source number included in the cached call notification, the processing circuit redirects the call to the destination number indicated in the call notification via the broadband network.

The present disclosure is directed to a method and system for routing communications. A plurality of communication devices are coupled, via a shared common (interface) communication device, to a communications network. Each such communication device has a device identifier. A user-programmable database associates each of a plurality of user identifiers with a device identifier. A call-routing switch arrangement is coupled to the communication network through a portal communication device having a portal identifier. A user designates a destination user identifier when making a communication. The call-routing switches the call, directing the call to a destination device identifier responsive to the user-programmable database association between the destination user identifier and a device identifier. According to certain aspects, communications are directed to users and are routed to associated equipment, a user identifier serving as a virtual extension number. Users remotely program the database to direct their calls accordingly.

Novel tools and techniques for implementing Transparent PSTN Failover. In some embodiments, a VoIP switch might receive a call request for connection between a calling party at a calling telephone number (TN) and a called party at a called TN. Based on a determination that connection via a session border controller (SBC) between the VoIP switch and an Internet Protocol (IP) gateway has been lost, the VoIP switch might route the call request to a PSTN redirect TN through the PSTN. One of the Class 5 switch of the PSTN or the IP gateway might replace the redirect TN with the called TN, based on a determination that the redirect TN represents a PSTN failover. The IP gateway may then route the call request to an IP PBX for terminating (establishing) the call connection between calling party and called party, without interception by an operator, receptionist, or auto attendant.

The embodiments described herein recite a telephone communication system used for handling information such as messages, typically voice mail messages, and, more particularly, is directed to a system that provides distributed session initiation protocol (SIP) silos. Distributed SIP silos (DSS) is a Communications Application Platform (CAP) feature that maintains the site's call capacity even when a signaling server fails. DSS uses multiple non-redundant signaling servers to provide SIP signaling for the same set of media ports. Because there are multiple signaling servers providing signaling for the same set of ports, the failure of one signaling server only terminates the calls it was actively processing and once those calls have been cleaned up, all the available (non-suspended) ports in the configuration are available to the remaining signaling servers.