Abstract: Methods for dynamically switching communications to text interactions are provided. In one example, a method includes the steps of receiving a phone call and determining a voice queue wait time exceeds a predetermined threshold. The phone call is routed to a speech-to-text queue based on the determination, and an audio signal associated with the phone call is converted to an instant message interaction.

Abstract: Methods for determining an active station based on sensor data are provided. In one example, a method includes the steps of receiving movement data and determining a pattern of movement associated with a contact center station based at least in part on the movement data. An active contact center station is updated based on the pattern of movement.

Abstract: Methods for enhancing call experiences through personal rules are provided. In one example, a method includes the steps of receiving a phone call and caller identification information associated with the phone call and routing the phone call to an interactive voice response system. A caller profile is developed based on a comparison of the caller identification information and one or more data stores. The caller profile is matched with a predetermined personal rule, and a dynamic message is generated in the interactive voice response system based on the matching predetermined personal rule.

Abstract: Methods for a secret supplemental username are provided. In one example, a method includes the steps of detecting a threat to an account with a primary username and password and disabling the primary username associated with the account in response to the threat. The method includes the additional steps of generating a supplemental username associated with the account and securely communicating the supplemental username.

Abstract: A method is presented for forming the excitation signal for a glottal pulse model based parametric speech synthesis system. In one embodiment, fundamental frequency values are used to form the excitation signal. The excitation is modeled using a voice source pulse selected from a database of a given speaker. The voice source signal is segmented into glottal segments, which are used in vector representation to identify the glottal pulse used for formation of the excitation signal. Use of a novel distance metric and preserving the original signals extracted from the speakers voice samples helps capture low frequency information of the excitation signal. In addition, segment edge artifacts are removed by applying a unique segment joining method to improve the quality of synthetic speech while creating a true representation of the voice quality of a speaker.

Abstract: A system and method are presented for predicting contact center behavior. In one embodiment, closed form simulation modeling may be used to simulate behavior from input distributions. Models may be created through staging and analysis of historical Automatic Call Distribution data. Service level, average speed of answer, abandon rate, and other data may be predicted to generate forecasts and analysis of contact center behavior. Examples of behavior may include staffing levels, workload, and the Key Performance Index of metrics such as service level percentage, average speed of answer, and abandonment rate percentage.

Abstract: A system and method are presented for communication routing. Communications may be routed into a queue based on criteria. A communication may be assigned to a distribution ring and a determination may be made as to whether there is an availability of resources to handle the communication. A communication may be re-routed if it cannot be handled to a new group. Re-routing may be based on criteria such as time delay, non-time based criteria such as skills, and environmental criteria, to name a few. Re-routing may continue until a communication is handled. The routing design may resemble concentric circles where the center represents the most desirable pool of handlers and each incremental ring represents the iterative expansion sets of agents.

Abstract: A system and method are presented for routing a communication through monitoring one or more of words and voice characteristics of a communicant during an interaction. A communication may be handled based on a score. A score may be calculated based on several factors that are analyzed during an interaction, such as, amplitude, word usage, call metrics, etc. Previous interactions of communicants may also be factored into a score. Such handling may include specialized routing of the communication to a designated handler, for example. A communication may be continually evaluated during its occurrence and said evaluation data stored for future use.

Abstract: A system and method are presented for context based knowledge retrieval. In one embodiment, such retrieval pertains to pattern recognition for data related to interactions between users, configuration and organization of systems data in an enterprise, and the quality of calculations of communication interactions. In one embodiment, a communication may be analyzed and a user interface created of potential sources of information related to the communication. This information may be from an internal source, such as a knowledge base, or an external source, such as an internet connected information source, for example. The user interface may comprise entities with associated links and actions, which may be configured based on a user's preference.

Abstract: A system and method are presented for the encoding of participants in a conference setting. In an embodiment, audio from conference participants in a voice-over-IP setting may be received and processed by the system. In an embodiment, audio may be received in a compressed form and de-compressed for processing. For each participant, return audio is generated, compressed (if applicable) and transmitted to the participant. The system may recognize when participants are using the same audio encoding format and are thus receiving audio that may be similar or identical. The audio may only be encoded once instead of for each participant. Thus, redundant encodings are recognized and eliminated resulting in less CPU usage.

Abstract: A system and method are presented for relationship management workflow processes. At least one embodiment may apply to process automation to health care. More specifically, the system and method may be applied to patient management of healthcare, such as the management of Diabetes or other medical conditions. Other embodiments may apply to process automation in other areas utilizing management workflow software.

Abstract: A system and method are presented for using spoken word verification to reduce false alarms by exploiting global and local contexts on a lexical level, a phoneme level, and on an acoustical level. The reduction of false alarms may occur through a process that determines whether a word has been detected or if it is a false alarm. Training examples are used to generate models of internal and external contexts which are compared to test word examples. The word may be accepted or rejected based on comparison results. Comparison may be performed either at the end of the process or at multiple steps of the process to determine whether the word is rejected.

Abstract: A system and method are presented for self-service callback modification. A user may request a callback. A unique confirmation identifier may be issued which may be used by the system to look up the callback record. A user may modify their callback request at any time during the callback process to receive their callback at a requested point in time, change the address to which the callback is addressed, alter the media through which the callback is placed, etc. Modification may occur in real time. Confirmation may be sent to the user after modification.

Abstract: Systems and methods for the matching of datasets, such as input audio segments, with known datasets in a database are disclosed. In an illustrative embodiment, the use of the presently disclosed systems and methods is described in conjunction with recognizing known network message recordings encountered during an outbound telephone call. The methodologies include creation of a ternary fingerprint bitmap to make the comparison process more efficient. Also disclosed are automated methodologies for creating the database of known datasets from a larger collection of datasets.

Abstract: Systems and methods for the matching of datasets, such as input audio segments, with known datasets in a database are disclosed. In an illustrative embodiment, the use of the presently disclosed systems and methods is described in conjunction with recognizing known network message recordings encountered during an outbound telephone call. The methodologies include creation of a ternary fingerprint bitmap to make the comparison process more efficient. Also disclosed are automated methodologies for creating the database of known datasets from a larger collection of datasets.

Abstract: A system and method are presented for acoustic echo cancellation. The echo canceller performs reduction of acoustic and hybrid echoes which may arise in a situation such as a long-distance conference call with multiple speakers in varying environments, for example. Echo cancellation, in at least one embodiment, may be based on similarity measurement, statistical determination of echo cancellation parameters from historical values, frequency domain operation, double talk detection, packet loss detection, signal detection, and noise subtraction.

Abstract: Systems and methods for the matching of datasets, such as input audio segments, with known datasets in a database are disclosed. In an illustrative embodiment, the use of the presently disclosed systems and methods is described in conjunction with recognizing known network message recordings encountered during an outbound telephone call. The methodologies include creation of a ternary fingerprint bitmap to make the comparison process more efficient. Also disclosed are automated methodologies for creating the database of known datasets from a larger collection of datasets.

Abstract: A system and method are presented for learning call analysis. Audio fingerprinting may be employed to identify audio recordings that answer communications. In one embodiment, the system may generate a fingerprint of a candidate audio stream and compare it against known fingerprints within a database. The system may also search for a speech-like signal to determine if the end point contains a known audio recording. If a known audio recording is not encountered, a fingerprint may be computed for the contact and the communication routed to a human for handling. An indication may be made as to if the call is indeed an audio recording. The associated information may be saved and used for future identification purposes.

Abstract: A system and method are presented for predicting speech recognition performance using accuracy scores in speech recognition systems within the speech analytics field. A keyword set is selected. Figure of Merit (FOM) is computed for the keyword set. Relevant features that describe the word individually and in relation to other words in the language are computed. A mapping from these features to FOM is learned. This mapping can be generalized via a suitable machine learning algorithm and be used to predict FOM for a new keyword. In at least embodiment, the predicted FOM may be used to adjust internals of speech recognition engine to achieve a consistent behavior for all inputs for various settings of confidence values.

Abstract: A system and method are presented for selectively biased linear discriminant analysis in automatic speech recognition systems. Linear Discriminant Analysis (LDA) may be used to improve the discrimination between the hidden Markov model (HMM) tied-states in the acoustic feature space. The between-class and within-class covariance matrices may be biased based on the observed recognition errors of the tied-states, such as shared HMM states of the context dependent tri-phone acoustic model. The recognition errors may be obtained from a trained maximum-likelihood acoustic model utilizing the tied-states which may then be used as classes in the analysis.