Abstract: A telephony system is presented that enables one user to monitor another user's extension and perform functions on calls destined for the other user's extension. The system includes a server with a telephony management software (TMS) unit, a telephony application programming interface (TAPI) unit, and an application unit. The application unit includes a control module, association module, presentation module, park module, monitoring device interface module, and storage interface module. A server sends information to a monitoring device regarding the status of a monitored extension and calls offering on that extension. A server also enables a user to navigate through multiple calls offering and park a selected call.

Abstract: A method for resolving conflicts between conference bridges in a distributed conferencing system includes determining that a conflict exists between a first conference bridge and a second conference bridge. The conflict may arise from the first conference bridge and the second conference bridge independently providing a meeting resource for a first meeting. The method also includes determining at the first conference bridge and at the second conference bridge that the meeting resource is to be provided by the first conference bridge. In determining that the meeting resource is to be provided by the first conference bridge, there is a probability of determining that the meeting resource is to be provided by the second conference bridge. The method also includes, providing the meeting resource for the first meeting at the first conference bridge and stopping providing the meeting resource for the first meeting at the second conference bridge.

Abstract: A graphical user interface for a telephone system includes programmable buttons which may be customized for individual users or groups of users. The programmable buttons enable user specific functionality to be provided, for example, enabling a button to be defined to record a call at a particular extension with one touch. The programmable buttons are collected onto toolbars and displayed on the user interface.

Abstract: An Internet Protocol (IP) phone transmits on a network, an IP packet comprising a dynamically-assigned IP address. A computer associated with the network uses the dynamically-assigned IP address in the packet to identify a specific site to which the IP phone belongs and/or a specific local area network (LAN) to which the IP phone is connected. Then, data to be used to provide telephony service to the phone is recorded in a memory. In some embodiments, the computer associates the IP phone with a dialing plan unique to the specific site, wherein the dialing plan includes an area code to be used by the IP phone.

Abstract: A system and method for falling back to a secondary network when there is insufficient quality on a primary network is disclosed. In one embodiment, a call manager has a media monitoring module, a bypass module, a PSTN control module and a GUI module. Once a call has been established between endpoints using a primary IP network, the media monitoring module monitors for degradation in call quality. If degradation in call quality is detected, the GUI module presents an interface to the user that allows selection of the secondary network or the primary IP network for continuation of the call. If the user selected the secondary network, the bypass module is activated to transfer the data communication to a secondary network such as a PSTN while the call control continues to be over the primary network.

Abstract: In one example embodiment, a method can include determining resource time data for an update contact data record corresponding to a state of a given resource in a distributed contact center system. A repository can be queried for conflicting data records associated with the given resource based on the resource time data. Any inconsistency for the resource time data can be resolved based on results to the querying and the update contact data record.

Abstract: A conference call system with a plurality of endpoints, a participant control server, a conference bridge and a switch is described. A first endpoint transmits conference data to the participant control server. The participant control server transmits the conference data to the other endpoints. In response to requests from the endpoint, the participant control server transmits recorded conference data for video, audio and web sessions that are individually controllable to pause, fast-forward and rewind the conference data. The participant control server establishes secondary connections for the endpoints including a ghost hold and a sidebar.

Abstract: A system and method for using a PSTN as a fallback for the IP media when the IP network cannot pass the media acceptably and while continuing to use the IP network for call control and other features. In one embodiment, a call controller has a media monitoring module, a bypass module and a PSTN control module. Once a call has been established between endpoints using a primary IP network, the media monitoring module monitors for degradation in call quality. If degradation in call quality is detected, the bypass module transfers the data communication to a secondary network such as a PSTN while the call control continues to be over the primary network. The present invention also includes a variety of methods including a method for falling back to a secondary network and a method for dynamically switching between a primary and a secondary network to ensure call quality.

Abstract: A computer readable medium has machine readable instructions that can include a fingerprint detector to evaluate input data relative to mobility fingerprint data that defines at least one predetermined zone that is a proper subset of overlap in coverage areas between first and second radio technologies to determine whether to trigger a handover between the first and second radio technologies for a mobile unit based on an evaluation of the mobility fingerprint data relative to the input data.

Abstract: A method for providing a status of a device to a plurality of devices in a distributed system includes receiving a status data packet from the device, the status data packet including the status of the device. The method also includes saving the status of the device to a memory and obtaining a list of the plurality of devices in the distributed system. The list is associated with a sequence for circulating the status data packet between the plurality of devices. The method further includes sending the status data packet including the status of the device to a next device in the sequence.

Abstract: A telephone comprising one or more speaker apertures, a microphone indicator disposed on a user-facing surface of the telephone, and one or more microphone apertures disposed proximate to the microphone indicator on a surface of the telephone other than the user-facing surface, where at least one of the microphone apertures is operable to allow audio to pass from the user to a microphone disposed in the telephone, and where the microphone indicator is arranged to direct the user to speak in a direction toward at least one of the microphone apertures.

Abstract: This invention overcomes the problem of delay associated with establishing connections with individual phones by providing a method for sending a virtual real time voice message processed through a VOIP system to a group of phones concurrently. The method includes assembling a portion of the voice message. The voice message includes a voice portion and an address portion. The voice portion of the voice message is buffered in a digital buffer. The address portion is used to determine the address of each phone in the group. After the address of each phone in the group is determined, an attempt is made to establish a connection with each phone. The method further includes waiting for a period of time. The period of time is determined based at least in part on a time duration required to establish a connection with a phone. After waiting the period of time, the voice portion of the voice message is sent to at least one phone in the group of phones.

Abstract: A system for placing a telephone call on hold and presenting video to the user that has been placed on hold comprises video on hold arbiter that either is a part of the server or the switch of an IP telephony system. The video on hold arbiter comprises a hold monitoring module, a switch interface module and a media source interface module. The video on hold arbiter monitors calls between parties for the selection of a hold signal indicating that a first caller is placing a second caller on hold. The video on hold arbiter then determines the capabilities of the first caller and the second caller, a source video signal and switches the call to a video-on-hold status. In the video-on-hold status, the video on hold arbiter provides a video signal to the caller on hold from the determined source video and matching the capabilities of the party on hold. Video on hold arbiter and monitors for an off hold signal.

Abstract: A computer system (a method) for exchanging information such as a Universal Resource Locator (URL) or a Universal Resource Identifier (URI) over a communication channel in general and over a telephone communications channel in particular in a unified communications system is disclosed. In one embodiment, the system comprises a phone URL exchange unit configured to receive a phone URL exchange request from a first endpoint, to determine the URL to be sent and URL encoding method based on the request, to encode the URL into audio data or as a part of the communications control signals using the URL encoding method and to send the encoded data to a second endpoint over the telephone communications channel. The phone URL exchange unit is further configured to receive audio data or the communications control signals embedded with a URL and to decode the received data to reveal the URL.

Abstract: A system and method for determining a speaker's position and a generating a display showing the position of the speaker. In one embodiment, the system comprises a first speakerphone system and a second speakerphone system communicatively coupled to send and receive data. The speakerphone system comprises a display, an input device, a microphone array, a speaker, and a position processing module. The position processing module is coupled to receive acoustic signals from the microphone array. The position processing module uses these acoustic signals to determine a position of the speaker. The position information is then sent to other speakerphone system for presentation on the display. In one embodiment, the position processing module comprises an auto-detection module, a position analysis module, a tracking module and an identity matching module for the detection of sound, the determination of position and transmission of position information over the network.

Abstract: A telephony system that accepts inputs, such as account codes, from a variety of devices and interfaces. The system includes a server with a telephony management software (TMS) unit, a telephony application programming interface (TAPI) unit, and an application unit. The application unit includes a control module, a call status module, an endpoint interface module, a storage interface module, a call transfer module, a TAPI interface module, a configuration module, and an extension library module. A server is capable of prompting several devices for an input and then accepting an input from any of the devices. Thus, a server enables a user to initiate a phone call on one device and then input information using a different device.

Abstract: A system for dynamically anchoring a call initiated by a mobile unit can include call anchor logic programmed with anchoring criteria to determine if a new outbound call being initiated from the mobile unit should be anchored at an anchor point located outside of a service provider network for the mobile unit based on application of the anchoring criteria to input data. The call anchor logic provides an anchor control signal to cause the outbound call to be one of (i) anchored at the anchor point and (ii) connected to a called party without anchoring at the anchor point. Memory that stores the input data, the memory being accessible by the call anchor logic.

Abstract: A microphone assembly includes a microphone array carrier having a top surface and a plurality of side faces adjacent to the top surface. The top surface and the plurality of side faces each have a substantially planar portion. The substantially planar portion of each of the side faces extends substantially perpendicular to the substantially planar portion of the top surface. Each side face has a side opening extending through the side face and the top surface has a top opening extending through the top surface. The microphone assembly also includes a plurality of microphones corresponding to the side faces. Each microphone is mounted with a microphone boot in a side opening and is isolated from the microphone array carrier by the microphone boot. The microphone assembly also includes a housing extending around the microphone array carrier and sound isolating material disposed between the microphone array carrier and the housing.

Abstract: A voice over internet protocol telephone system is described in which services for handling calls invoking account codes are distributed to voicemail servers in the system. If the voicemail server associated with a particular user does not have services for handling account code calls, then the request is forwarded to another voicemail server in accordance with a predefined procedure.