Abstract: A method of transmitting DTMF signals over a compressed computer network comprises sending a packet containing a compressed DTMF signal followed by a confirmation packet. The confirmation packet preferably comprises an uncompressed DTMF signal. To detect the DTMF signal transmitted, a telephony Internet server decompresses packets as they are received. If a packet appears to contain a compressed DTMF signal, the telephony Internet server looks for the presence of the confirmation packet before processing the previously received packet as a DTMF signal. In another embodiment, an output of a speech compressor and a DTMF signal detector are transmitted on separate virtual channels of a real time protocol data stream.

Abstract: A telephony-over-LAN client is provided which disables a screen saver during active calls by generating a software interrupt indicative of either false keyboard interrupts or false mouse movements. The ToL client is configured to periodically activate such an interrupt during an active call immediately prior to the end of the timeout period of the screen saver. Alternatively, or in addition to the above-mentioned false keyboard interrupts or mouse movements, a false write to the hard drive might be provided.

Abstract: An improved prompting system for accessing specified functions. According to one embodiment, a particular user may identify himself to the system and thereafter preprogram one or more paths that lead directly to a desired function within the hierarchical menu prompting system. The next time the user needs to access the system the user can navigate directly to the desired function by pressing a predetermined key or sequence of keys on his or her Touch-Tone dial. Alternatively, the prompting system may present as an option a direct path to the desired function.

Abstract: A method and system for automated silent call monitoring in an automatic call distributor (ACD) environment includes configuring a set of call performance profiles which include voice data patterns which are descriptive of voice data transmissions during an ACD call and which are associated with substandard agent performance. Each voice data pattern has a corresponding threshold which represents the maximum number of detections tolerated in an ACD call prior to execution of a notification routine. A digital signal processor (DSP) monitors a first call between an agent terminal and a customer terminal for the voice data patterns and stores detection data in memory upon detecting of one of the voice data patterns. A central processor unit (CPU) compares the number of detections of the voice patterns within predetermined time intervals to the threshold numbers of detections represented in the thresholds to determine if any threshold has been exceeded.

Abstract: A system and a method for communicating dialed digits between a communication device and a call processor during call setup include transmitting at least the first digit of a telephone number from the communication device to the call processor within a first data packet. The call processor references a dialing plan to determine if the first data packet provides sufficient digits to complete the requested call. If not, the call processor determines the minimum quantity of digits required under the dialing plan to determine the destination of the call and transmits a protocol command to the communication device to transmit the minimum quantity of digits to the call processor within a subsequent data packet. The call processor continues to reference the dialing plan to determine whether additional digits are required and to request additional packets until sufficient packets have been received to establish the call.

Abstract: A method and system provide user management of electronic message transmissions between servers and client devices, particularly transmissions of files attached to electronic messages. The method and system operate to transfer selected attached files to and from a client device. When accessing electronic messages with attached files stored at a server by a user, the attached files that satisfy a prescribed requirement are automatically downloaded from the server to the client device utilized by the user. The prescribed requirement may include maximum file size and download time, and approved list of file formats and senders. Preferably, the prescribed requirement is entered into the client device by the user. The prescribed requirement can vary depending upon the personal preference of the user. When forwarding received electronic messages with attached files, only the attached files that have been modified are uploaded from the client device to the server.

Abstract: A system and a method for managing incoming calls within a call queue include a network bandwidth monitor for monitoring network bandwidth availability on a call center and a queue call rate monitor for determining a total of all call charge rates associated with the calls. A queue manager determines the order of advancement of the calls within the queue. If the network bandwidth monitor determines that network bandwidth availability is below a predetermined threshold, high network bandwidth calls are preferentially advanced to free up additional network bandwidth. Calls with high charge rates are preferentially advanced if the total call charge rate in the queue exceeds a predetermined maximum threshold. If the maximum total call charge rate threshold is exceeded and the network bandwidth availability level is below the minimum threshold, a relative priority for maintaining network bandwidth availability and reducing total call charge rates will determine the order of call advancement.

Abstract: An embodiment of the present invention provides a system and method for selecting voice compression and coding based on capabilities of all intermediary networks and links in addition to the capabilities of the end points. The present invention optimizes voice quality by determining a minimum number of new conversions, herein referred to as transcodings, necessary in voice connections, on a system wide basis. According to an embodiment of the present invention, a sender sends a signaling message to a receiver. The signaling message is sent to the receiver prior to sending an actual message. Each entity or device capable of converting voice coding, herein referred to as a “station”, between the sender and the receiver identifies their capabilities to the signaling message. The receiver, which is also a station, also identifies its capabilities to the signaling message.

Abstract: A method and apparatus for preventing network rerouting when the overhead associated with the reroute operation could result in system resources being overwhelmed. The system monitors link congestion and CPU utilization that will be caused by rerouting. A comparator determines whether the amount of congestion and CPU utilization is beyond an acceptable predetermined threshold. A reroute control unit prevents rerouting if the amount of link congestion and CPU utilization exceeds the predetermined threshold.

Abstract: A CPU speed control system for use in devices having microprocessors or microcontrollers which may also be operated using battery power. The system includes a programmable frequency synthesizer for providing the CPU and other system buses in the device with a variable clocking frequency based on the application or interrupt being executed by the device. The frequency synthesizer output may be dynamically varied according to a particular application or task that is to be executed by the device or based on total CPU usage. The operating system of the device and the CPU are both capable of controlling the frequency synthesizer output to ensure that the CPU is operating at the most power efficient level possible.

Abstract: A system and method for telephone trouble reporting. One aspect of the present invention is a context-sensitive graphical user interface (GUI). The GUI allows a user to select a trouble reporting option at various stages of a telephone call in a telephony-over-LAN system. Thus, a series of menus listing potential call difficulties are provided in the interface of the digital or IP phone or telephony-enabled computer to the user during the telephone call. Selecting one of the options allows the user to report the problem to the server or other central switch.

Abstract: An improved prompting system for accessing specified functions in an interactive voice response (IVR) system. The IVR system (101) presents a plurality of prompts to callers. The prompts are typically presented in a predetermined order. However, the IVR system (101) improves the presentation by adjusting the presentation order based on a caller pattern of prompt selection. By monitoring incoming calls to determine what prompts are most frequently selected, the IVR system (101) can tailor the presentation order to the needs of individual users. The IVR system (101) can monitor either the selections made by all users or those made by a particular individual user over a configurable time period to establish the pattern of prompt selection.

Abstract: A method and system provide user management of electronic message transmissions between servers and client devices, particularly transmissions of files attached to electronic messages. The method and system operate to transfer selected attached files to and from a client device. When accessing electronic messages with attached files stored at a server by a user, the attached files that satisfy a prescribed requirement are automatically downloaded from the server to the client device utilized by the user. The prescribed requirement may include maximum file size and download time, and approved list of file formats and senders. Preferably, the prescribed requirement is entered into the client device by the user. The prescribed requirement can vary depending upon the personal preference of the user. When forwarding received electronic messages with attached files, only the attached files that have been modified are uploaded from the client device to the server.

Abstract: A method of transmitting DTMF signals over a compressed computer network (24) comprises sending a packet containing a compressed DTMF signal followed by a confirmation packet. The confirmation packet preferably comprises an uncompressed DTMF signal which is transmitted on the same logical channel as the packets that represent voice signals. To detect the DTMF signal transmitted, a telephony Internet server (28) decompresses packets as they are received. If a packet appears to contain a compressed DTMF signal, the telephony Internet server looks for the presence of the confirmation packet before processing the previously received packet as a DTMF signal. In an alternative embodiment, the confirmation packet comprises a code that represents a DTMF signal. The packets that represent the voice signal and the code representing the DTMF signal are sent on separate virtual channels of the same logical channel.

Abstract: A conferencing system (100) improves audio quality by identifying and selectively allowing a dominant audio channel to bypass a lossy compression transformation included in a server 102. The system (100) places conference calls over a network (106) and includes the conference server (102), having an input interface (200), a decompressor (202), a summation circuit (212), a correlation unit (206) a compressor (214), and an output interface (220). The input interface (200) receives a plurality of audio channels, each being compressed using a lossy compression algorithm. The decompressor (202) decompresses each of the incoming channels to produce a plurality of decompressed audio channels, which are then summed by the summation circuit (212) to produce a summation stream. The compressor (214) compresses the summation stream to produce a first audio output. The correlation unit (206) identifies a dominant channel from the incoming compressed audio channels.

Abstract: A data routing system for dynamically optimizing bandwidth which includes a network control unit, a route comparator, a reroute control unit and a packet reordering unit. The network control unit reserves the initial route from the source node to the end node. The route comparator determines whether an alternate route is more optimal than the initial route and the reroute control unit establishes the alternate route and reroutes the data over the alternate route. The packet reordering unit maintains the sequence of the data packets on the alternate route.

Abstract: A communication system (70) for transmitting voice or data signals to a remote receiver detects a digital error message that is transmitted by a telephone service provider. A Caller_ID processor (80) detects and decodes the digital error messages that are transmitted on an analog telephone line. A central processing unit (180) detects and decodes the digital error messages that are transmitted on a signaling or D-channel of a digital telephone line. Once the received error messages are decoded, they are analyzed by the communication system for the appropriate action to take in order to complete a telephone call.

Abstract: A network node 102 capable of detecting how many parties have collided during attempts to transmit on a bus 108. If more than one party has been detected to collide, the node 102 selects a backoff window from a larger randomized group of possible backoff windows than if the node had collided with only one other party. In this way, delays due to multiple parties colliding are minimized.

Abstract: A method for communicating location-specific messages includes storing a library of such messages within a portable device having a capability of randoly accessing the messages. Each location-specific message has a content that is related to a particular geographical location within a facility. At each of the geographical locations, there is a device-readable identification of an identifier for each location-specific message relating to that geographical location. In the preferred embodiment, the identifier is transmitted to the portable device wirelessly. Also in the preferred embodiment, the message library is contained on a recording medium that facilitates automated random access, such as a compact disk, and the portable device is a personal digital assistant (PDA). In an application of the method, map information may be prerecorded and supplemented or complemented with time-dependent information downloaded from a source.

Abstract: A method and system of processing speech information includes segmenting the speech information based upon detection of logical speech boundaries, such as isolated words, prior to compressing and/or transmitting the speech information. In one embodiment, a continuous stream of voice data is analyzed to detect signal segments containing the characteristics of an isolated word, thereby forming frames of speech information. The frames are data compressed to form packets that are transmitted to a remote site. Preferably, the packets include error checking information. In a receive mode, incoming packets are error checked prior to packet decoding. If transmission errors are detected, repairable packets may be corrected. Non-correctable errors cause generation of notice data that are used to notify a listener of the location of lost speech information. Notice data are also generated if the duration between two arriving packets exceeds a preselected threshold.