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    Convenience Features in a Method and System for Providing Enhanced Caller Identification
    6.
    发明申请
    Convenience Features in a Method and System for Providing Enhanced Caller Identification 有权
    提供增强的来电显示的方法和系统中的方便功能

    公开(公告)号：US20080031432A1

    公开(公告)日：2008-02-07

    申请号：US11869534

    申请日：2007-10-09

    申请人： Teresa Latter ,  John Moss ,  Thomas McBlain ,  James Maciejewski ,  Mary Hardzinski

    发明人： Teresa Latter ,  John Moss ,  Thomas McBlain ,  James Maciejewski ,  Mary Hardzinski

    IPC分类号： H04M1/64

    CPC分类号： H04M3/42 ,  H04M3/42042 ,  H04M3/436 ,  H04M2207/12 ,  H04M2242/22

    摘要： A computer readable storage medium storing computer readable program code, the code configured to provide caller identification information to a called party when standard Caller ID cannot be provided. If standard caller ID information cannot be provided for a call, the call is interrupted by the present service and the calling party is prompted for audible caller identification information. In one embodiment, when the audible caller identification is provided, the call is presented to the called party with a distinctive ring at the called telephone station. In another embodiment, the calling party may enter a personal identification number (PIN) to bypass the requirement for audible caller identification information. In still another embodiment, when the PIN is entered, the call is presented with a distinctive ring at the called communication station. In still another embodiment, the called party may send the incoming call to a voice mail system.

    摘要翻译： 一种存储计算机可读程序代码的计算机可读存储介质，所述代码被配置成当不能提供标准来电显示时，向被叫方提供呼叫者识别信息。 如果不能为呼叫提供标准呼叫者ID信息，则呼叫被当前服务中断，并且提示主叫方听到呼叫者识别信息。 在一个实施例中，当提供可听见的呼叫者识别时，在被叫电话台将呼叫呈现给具有独特响铃的被叫方。 在另一个实施例中，呼叫方可以输入个人识别号码（PIN）以绕过对可听呼叫者识别信息的要求。 在另一个实施例中，当PIN被输入时，呼叫在被叫通信站处被呈现为具有特殊的环。 在另一个实施例中，被叫方可以将呼入呼叫发送到语音信箱系统。

    Method and system for text-to-speech conversion of caller information
    7.
    发明申请
    Method and system for text-to-speech conversion of caller information 审中-公开
    呼叫者信息的文本到语音转换的方法和系统

    公开(公告)号：US20060083364A1

    公开(公告)日：2006-04-20

    申请号：US11245479

    申请日：2005-10-05

    申请人： Robert Bossemeyer ,  Mary Hardzinski ,  Thomas McBlain

    发明人： Robert Bossemeyer ,  Mary Hardzinski ,  Thomas McBlain

    IPC分类号： H04M11/00 ,  H04M1/64

    CPC分类号： H04M3/42042 ,  G10L13/00 ,  H04M1/575 ,  H04M3/42059 ,  H04M3/428 ,  H04M3/4288 ,  H04M2201/39 ,  H04M2201/60 ,  H04M2207/12 ,  H04M2242/22

    摘要： A system (10) and method (40) for providing audible caller information to telephone service subscribers is presented. The system (10) includes a pre-processor (28) that converts textual caller information from a first data format to a second data format suitable for text-to-speech synthesis (TTS). The pre-processor (28) can be incorporated in a service node/intelligent peripheral (SN/IP) (24) in an advanced intelligent network (AIN). In this configuration, the pre-processor (28) can be used to convert caller-ID with name database entries into a predetermined TTS format. The TTS formatted information can be provided to a TTS synthesized (30) included in the SN/IP (24). In conjunction with a switching service point (SSP) (20), the pre-processor (28) can provide a talking call-waiting feature to subscribers with significantly improved intelligibility.

    摘要翻译： 提出了一种用于向电话服务订户提供可听呼叫者信息的系统（10）和方法（40）。 系统（10）包括将文本呼叫者信息从第一数据格式转换成适合于文本到语音合成（TTS）的第二数据格式的预处理器（28）。 预处理器（28）可以并入高级智能网（AIN）中的服务节点/智能外设（SN / IP）（24）中。 在该配置中，预处理器（28）可用于将具有名称数据库条目的呼叫者ID转换成预定的TTS格式。 可以将TTS格式的信息提供给包含在SN / IP（24）中的合成的TTS（30）。 结合切换服务点（SSP）（20），预处理器（28）可以向订户提供具有显着改善的可懂度的谈话呼叫等待功能。

    Distributed text-to-speech synthesis between a telephone network and a telephone subscriber unit
    8.
    发明申请
    Distributed text-to-speech synthesis between a telephone network and a telephone subscriber unit 审中-公开

    公开(公告)号：US20050157861A1

    公开(公告)日：2005-07-21

    申请号：US11052625

    申请日：2005-02-07

    申请人： Robert Bossemeyer ,  Mendel Hamrick ,  Kelly Multach ,  Elizabeth Hart ,  Kevin Kaschke ,  Mary Hardzinski ,  Thomas McBlain

    发明人： Robert Bossemeyer ,  Mendel Hamrick ,  Kelly Multach ,  Elizabeth Hart ,  Kevin Kaschke ,  Mary Hardzinski ,  Thomas McBlain

    IPC分类号： H04M1/57 ,  H04M3/428 ,  H04M1/56 ,  G10L13/00 ,  G10L13/08 ,  H04B1/38 ,  H04M1/00 ,  H04M15/06

    CPC分类号： H04M3/428 ,  H04M1/575 ,  H04M3/42042 ,  H04M3/42102 ,  H04M2201/60 ,  H04M2242/22

    摘要： A telecommunications system (600) distributes text-to-speech synthesis (900) between a telephone network (603) and a telephone subscriber unit (602). The telephone network (603) receives (1102) a telephone call from a first telephone subscriber unit (601) over a first communication channel (604) intended for a second telephone subscriber unit (602), determines (1103) that the second telephone subscriber unit (602) subscribes to a speech-based caller identification service provided by the telephone network (603), converts (1107) text information (707), representing the caller identification of the first telephone subscriber unit (601) into symbols (708), encodes (1108) the symbols (708) to form a data stream (709), opens (1109) a second communication channel (605) between the telephone network (603) and the second telephone subscriber unit (602), and sends (1110) the data stream (709) to the second telephone subscriber unit (602) over the second communication channel (605). The second telephone subscriber unit (602) detects (1202) that the telephone network (603) opened the second communication channel (605), receives (1203) the data stream (807) from the telephone network (603), decodes (1206) the data stream (807) to form decoded symbols (808), converts (1207) the decoded symbols (808) to form a speech waveform (809), and generates (1211) speech from the speech waveform (809) to permit a second party associated with the second telephone subscriber unit (602) to listen to the speech to identify an identity of a first party associated with the first telephone subscriber unit (601) prior to accepting the telephone call from the first telephone subscriber unit (601). The symbols (708) may be generated at various points (909, 910) within the distributed text-to-speech synthesizer (900) depending on the requirements and limitations of the telecommunication system (600).