Abstract: The cloud computing VoIP system includes a software as a service (SaaS) site and one or more additional sites. The SaaS site includes a gateway and a call manager and is coupled for communication across an IP network and the PSTN. The call manager includes a media monitoring module, a bypass module, a PSTN control module and a GUI module. The SaaS site handles and controls voice and video calls over the IP network and the PSTN for endpoints at the one or more additional sites. In another embodiment, the cloud computing VoIP system is also coupled to a cellular network and the endpoints include an associated cellular phone. The cloud computing VoIP system also includes a method for falling back to a secondary network, namely the cellular network, when there is insufficient quality on a primary network.

Abstract: A system and method for managing an IP telephony network that uses a configuration map for organizing sites in a tree-like hierarchy. The system is capable of managing the addition, removal and movement of IP telephony elements within an IP telephony network and locating services and IP telephony elements within the IP telephony network in an efficient manner.

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: Systems and methods are provided wherein an incoming plain old telephone systems (PSTN) signal is input into a network conversion interface. The network conversion interface first determines the data type of the incoming signal. Next, the network conversion interface creates a routing sequence of the incoming signal based on the determined data type. Finally, the network conversion interface converts the incoming signal into an 802.X format and sends the signal to an appropriate IP device based on the determined signal type. Should the network conversion interface lose electrical power, a drop contactor routes the incoming signal directly to an analog device without creating a routing sequence or performing an 802.X conversion.

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 compact low cost beamforming microphone assembly for a desk telephone is described. The assembly includes a microphone carrier array having a top surface and having faces arrayed about an exterior surface, each to receive a microphone. Each microphone is mounted in a microphone boot and inserted into a microphone carrier. The carrier array fits into a housing having a cover allowing sound to reach the microphones.

Abstract: A compact low cost beamforming microphone assembly for a desk telephone is described. The assembly includes a microphone carrier array having a top surface and having faces arrayed about an exterior surface, each to receive a microphone. Each microphone is mounted in a microphone boot and inserted into a microphone carrier. The carrier array fits into a housing having a cover allowing sound to reach the microphones.

Abstract: A distributed VoIP system includes a network and a first switch at a first site coupled to the network. The first switch is configured to provide telephony services to a first communication device. The system also includes a second switch at a second site coupled to the network. The second switch is configured to provide telephony services to a second communication device. The system also includes a spare switch coupled to the network. The spare switch is configured to provide telephony services to the first communication device if the first communication device is unable to register with the first switch, and the spare switch is configured to provide telephony services to the second communication device if the second communication device is unable to register with the second switch.

Abstract: An apparatus and a method for monitoring availability of a contact are described. The apparatus includes an interface module for receiving a selection from a user that indicates the contact to be monitored. The apparatus also includes a presence change alert manager that is coupled to the interface module and is coupled to a network. The presence changer alert manager is configured to determine at least one relevant type of availability for the contact and to determine if availability information obtained from the network corresponds to a change in a relevant type of availability for the contact. The apparatus further includes a notification manager coupled to the presence change alert manager. If the presence change alert manager determines that the obtained availability information corresponds to a change of a relevant type of availability for the contact, the notification manager generates a notification alerting the user of the change.

Abstract: A distributed VoIP system includes a first switch coupled to a network and configured to provide voicemail features to a first group of one or more users. The system also includes a voicemail server coupled to the network. The voicemail server is configured to provide voicemail features to a second group of one or more users. The first switch is configured to store voicemail data associated with only the first group, and the voicemail server is configured to store voicemail data associated with both the first and second groups.

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 bandwidth management and codec negotiation, according to one embodiment of the present invention comprises: a configuration storage module having supported codecs storage, codec lists and preferred site settings storage, and a call manager having an extension module, a trunk module, a location service engine, a codec manager, a bandwidth manager, and a media manager. The codec manager and the bandwidth manager used for negotiating a codec for a call between two endpoints. The present invention also includes a number of methods including a method for negotiating a codec for a call, a method for managing bandwidth for a call, a method for adding a description of a new codec supported by an endpoint, a method for adding an identifier of a supported codec to a codec list and a method for editing code site codec settings.

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: A system and method for managing an IP telephony network that uses a configuration map for organizing sites in a tree-like hierarchy. The system is capable of managing the addition, removal and movement of IP telephony elements within an IP telephony network and locating services and IP telephony elements within the IP telephony network in an efficient manner.

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: Session-based data, such as call detail accounting information, is tracked and distributed to a plurality of servers in a distributed telephony environment. One type of session is a telephone call between participants. Session-based data includes information about a session provided by the switches along the media path. Session-based data further includes information related to the session obtained from the application layer (e.g., user input metadata). A network proxy maintains a participant list to which the session data is distributed. The network proxy sends updates to the participants and determines the network status to implement fault tolerance of data distribution.

Abstract: The number of holes that are opened in a firewall for internet telephony is limited to a first hole used for call control and a second hole used for audio traffic. Fixed destination ports for telephony traffic and call control traffics are created at a destination. Media streams are received at the telephony fixed destination port. The source of each media stream is commanded to provide a unique identifier for each media stream arriving at the destination from each source. Each media stream is identified by a unique identifier provided by the source. The unique identifier for each media stream is communicated to the destination by each source over call control. All telephony traffic is received only at the fixed destination port or telephony and all call control is received only at the fixed destination port for call control.

Abstract: A simultaneous hunt group system and method are provided. A call status module notices an incoming call to a simultaneous hunt group and offers the call to a plurality of endpoints. One of the plurality of endpoints answers the call. After the call is answered, a resource allocation module allocates resources for a communications channel, and a call transfer module transfers the caller to the agent that answered the call.

Abstract: A device forms packets for a stream of information being transmitted over a packet switched network (such as the Internet) in the normal manner. The device responds to a predetermined event by deliberately forming packets to provide a different quality of service, as compared to the service quality provided by packets that were formed prior to the event. The predetermined event can be related to deterioration or improvement in network performance, and the device can respond by correspondingly decreasing or increasing the service quality of the packets being formed. Specifically, the predetermined event can indicate an increase in processing requirements beyond a threshold, and the device reduces service quality by including a larger amount of information (also called “payload”) in each packet formed after the event, as compared to the amount of information included prior to the event.