公开(公告)号：US06914899B2

公开(公告)日：2005-07-05

申请号：US09771991

申请日：2001-01-30

申请人： Joseph Siegrist ,  Amen Zwa ,  Ian Hunter

发明人： Joseph Siegrist ,  Amen Zwa ,  Ian Hunter

IPC分类号： H04L29/08 ,  H04M3/51 ,  H04M7/00 ,  H04L12/66

CPC分类号： H04M7/1295 ,  H04L67/02 ,  H04L67/1095 ,  H04L67/14 ,  H04L67/16 ,  H04L67/26 ,  H04L67/34 ,  H04M3/5191 ,  H04M7/00 ,  H04M7/0027 ,  H04M2201/42

摘要： An data packet network telephony system provides identification information of an end user terminal making an Internet telephone call to a call center terminal and provides a call center agent with the ability to cause an end user terminal making an Internet telephone call to display selected web pages and or to synchronize displays between the agent and end user terminals. In one embodiment, customer identification information is provided in-band. In another embodiment, customer identification information is accomplished out of band, in some cases using an identifier of the PSTN gateway as an index into a table to identify an end user terminal. In another embodiment, an agent may enter DTMF (dual tone multi-frequency tones) to push web pages to an end user terminal. In still other embodiments, a second communications link between the end user terminal and the agent's terminal is established, allowing for data sharing/synchronization.

摘要翻译： 数据分组网电话系统提供向呼叫中心终端进行因特网电话呼叫的终端用户终端的识别信息，并向呼叫中心代理提供使终端用户终端进行因特网电话呼叫来显示所选网页的能力， 或者同步代理和终端用户终端之间的显示。 在一个实施例中，带内提供了客户识别信息。 在另一个实施例中，客户识别信息是在频带内完成的，在某些情况下使用PSTN网关的标识符作为表中的索引来标识最终用户终端。 在另一个实施例中，代理可以进入DTMF（双音多频音调）以将网页推送到终端用户终端。 在其他实施例中，建立终端用户终端和代理终端之间的第二通信链路，允许数据共享/同步。

公开(公告)号：US07075921B2

公开(公告)日：2006-07-11

申请号：US09771992

申请日：2001-01-30

申请人： Joseph Siegrist ,  Amen Zwa

发明人： Joseph Siegrist ,  Amen Zwa

IPC分类号： H04L12/66 ,  H03G3/20

CPC分类号： H04L67/16 ,  H04L67/02 ,  H04L67/04 ,  H04L67/14 ,  H04L67/26 ,  H04L67/34 ,  H04L69/329 ,  H04M3/5183 ,  H04M7/0027 ,  H04M7/1295 ,  H04M2201/42

摘要： A person receiving a Voice over IP call, such as a call center agent, remotely controls media settings on a caller's multimedia terminal device functioning as a telephone terminal device. In some embodiments, the call center agent may remotely control media settings by entering a DTMF (dual tone multi-frequency) sequence from the telephone keypad. The DTMF sequence may be recognized by a gateway through which an Internet-to-PSTN call is made, stripped out of the audio stream and sent to the caller's multimedia terminal device in a separate command message through the Internet, or the DTMF sequence may be encoded into packets along with the voice information for detection and translation into a command message by software at the caller's terminal device. In pure VoIP applications, command packets may be sent directly. When a command message is received at the caller's terminal device, software on the end user computer makes the corresponding media setting changes without requiring any intervention on the part of the caller.

公开(公告)号：US06707811B2

公开(公告)日：2004-03-16

申请号：US09771993

申请日：2001-01-30

申请人： Jeffrey Douglas Greenberg ,  Joseph Siegrist ,  Amen Zwa

发明人： Jeffrey Douglas Greenberg ,  Joseph Siegrist ,  Amen Zwa

IPC分类号： H04L1266

CPC分类号： H04L61/157 ,  H04L29/1216 ,  H04M3/42178 ,  H04M3/51 ,  H04M7/003 ,  H04M7/1205

摘要： The universal Internet based telephony system is implemented as a process that is accessible via an Internet web page and executes on the web server that hosts the web page and/or a back-end Internet telephony server which is accessed and controlled by the web server. The customer accesses the universal Internet based telephony system via any existing Java Enabled Internet Browser software resident on the customer's personal computer, either as an adjunct process thereon, or as a dedicated Internet telephony process. When a customer accesses the Internet web page and clicks on the universal Internet based telephony system icon, the web server on which the web page resides executes the resident universal Internet based telephony system hyperlink script and transfers a newly opened browser session to the universal Internet based telephony system web site. The web server uploads an applet to the customer's personal computer to run on the client machine without disturbing the customer's existing web page access, by opening up a separate window on the customer's personal computer for the universal Internet based telephony system, to be seen and operated by the user. The universal Internet based telephony system web site then implements the Internet telephony communication connection between the customer's personal computer and the designated destination.

摘要翻译： 基于互联网的通用电话系统被实现为可以经由因特网网页访问并且在托管网页的网络服务器上执行的进程和/或由网络服务器访问和控制的后端因特网电话服务器。 客户通过驻留在客户的个人计算机上的任何现有的Java Enabled Internet Browser软件访问基于因特网的通用电话系统，作为其上的辅助处理，或者作为专用的因特网电话过程。 当客户访问互联网网页并点击通用因特网的电话系统图标时，网页驻留的网络服务器执行驻留的基于因特网的通用电话系统超链接脚本，并将新打开的浏览器会话传输到通用因特网 电话系统网站。 网络服务器将小程序上传到客户的个人计算机上，以在客户端计算机上运行，​​而不会影响客户现有的网页访问，通过在客户的个人计算机上打开用于通用因特网的电话系统的单独窗口，以便查看和操作 由用户 因此，通用因特网的电话系统网站实现了客户个人计算机与指定目的地之间的因特网电话通信连接。

公开(公告)号：US08498405B2

公开(公告)日：2013-07-30

申请号：US11742136

申请日：2007-04-30

申请人： Joseph Siegrist

发明人： Joseph Siegrist

IPC分类号： H04M5/00 ,  G06Q30/00

CPC分类号： H04M7/003 ,  H04L65/1076

摘要： A system and method for enhanced tracking of calls resulting from referrals to a website from third party servers includes the steps of identifying a tracking parameter, dynamically allocating a phone number to the tracking parameter, displaying the dynamically allocated number on a web page, and tracking calls received on the number. The dynamically allocated number may replace an existing number on the web page. The tracking parameter can be a keyword used in a search, an identifier of the referring website, or any other parameter, including combinations thereof. The tracking parameter may be included in the referring URL, the URL of the website, or in the body of a web page. In some embodiments, the call tracking capability is provided by a service provider other than an operator of the merchant website and JavaScript is used so that the process is easy to implement by the merchant website.

摘要翻译： 用于增强对从第三方服务器转到网站的呼叫进行跟踪的系统和方法包括以下步骤：识别跟踪参数，动态地将电话号码分配给跟踪参数，在网页上显示动态分配的号码，以及跟踪 收到的电话号码。 动态分配的号码可以替换网页上的现有号码。 跟踪参数可以是搜索中使用的关键字，引荐网站的标识符，或任何其他参数，包括其组合。 跟踪参数可以包括在引用URL，网站的URL或网页的正文中。 在一些实施例中，呼叫跟踪功能由除商户网站的运营商之外的服务提供商提供，并且使用JavaScript来使得该过程易于由商家网站实现。

公开(公告)号：US08284702B2

公开(公告)日：2012-10-09

申请号：US12878762

申请日：2010-09-09

申请人： Joseph Siegrist

发明人： Joseph Siegrist

IPC分类号： H04L12/16 ,  H04Q11/00 ,  H04M3/42

CPC分类号： H04Q3/0045 ,  H04L12/66 ,  H04M3/38 ,  H04M3/42059 ,  H04M3/42102 ,  H04M3/42195 ,  H04M3/436 ,  H04M7/006 ,  H04M2201/12 ,  H04M2201/14 ,  H04Q3/72

摘要： In one method and apparatus for blocking unwanted calls in a callback system, parameters associated with a callback request (e.g., the user device to which the call is to be directed, the IP address from which the request originated or an identifier of the source computer from which the callback request originated, the party from whom the call is requested) are tracked for each call. A decision to automatically block the requested callback is made based on one or more criteria based on a combination of these parameters, which may be a total over a period of time. In a second method, one or both of the parties to the callback is queried as to whether the callback is unwanted and, if so, one or more parameters associated with the request are blacklisted.

摘要翻译： 在用于阻止回呼系统中的不需要的呼叫的一种方法和装置中，与回叫请求相关联的参数（例如，呼叫所针对的用户设备，发起请求的IP地址或源计算机的标识符 从其发起回叫请求，请求呼叫的一方）为每个呼叫进行跟踪。 基于一个或多个基于这些参数的组合的标准来做出自动阻止所请求的回叫的决定，这些参数可以是一段时间内的总和。 在第二种方法中，查询回调的一方或双方是否回调是不需要的，如果是，则与请求相关联的一个或多个参数被列入黑名单。

公开(公告)号：US08081623B2

公开(公告)日：2011-12-20

申请号：US12270226

申请日：2008-11-13

申请人： Ian Halpern ,  John Piescik ,  Joseph Siegrist

发明人： Ian Halpern ,  John Piescik ,  Joseph Siegrist

IPC分类号： H04L12/66

CPC分类号： H04L12/66

摘要： In an emergency backup telephone system, members of an enterprise use their personal computers to log into an emergency communications web page. Upon logging in, software that enables the personal computer to act as a webphone is automatically downloaded. This software allows a person to initiate a call from personal computer to a conventional PSTN number destination using a PSTN gateway, or to another party's computer at a specified URL using VoIP telephony. Upon logging in, an authoritative index of employees reachable via the backup system updated to include information such as a phone number and/or IP address where the member can be reached in order to allow calls originating from the PSTN to be routed to the member's computer. The index is made available to other members of the enterprise via the enterprise's intranet, and, in some embodiments, to the public via a web page on the internet and/or email.

摘要翻译： 在紧急备用电话系统中，企业的成员使用其个人计算机登录紧急通信网页。 登录后，将自动下载使个人计算机充当网络电话的软件。 该软件允许个人使用PSTN网关从个人计算机发起呼叫到常规PSTN号码目的地，或者使用VoIP电话在指定的URL处发起另一方的计算机呼叫。 登录后，通过备份系统可访问的雇员的权威索引被更新为包括诸如可以到达成员的电话号码和/或IP地址的信息，以便允许从PSTN发起的呼叫被路由到该成员的计算机 。 该索引通过企业的内联网向企业的其他成员提供，并且在一些实施例中通过互联网和/或电子邮件上的网页向公众提供。

公开(公告)号：US07203188B1

公开(公告)日：2007-04-10

申请号：US10151036

申请日：2002-05-21

申请人： Joseph Siegrist ,  Michael Masinick

发明人： Joseph Siegrist ,  Michael Masinick

IPC分类号： H04L12/66 ,  H04L12/54 ,  H04L12/28 ,  H04L12/56 ,  H04M1/64

CPC分类号： H04M3/4938 ,  H04M7/003 ,  H04M7/0036

摘要： A communications system includes an automated voice processor such as an IVR (interactive voice response system) that can control a display associated with an end user calling the automated voice processor over a data network such as the Internet. In a preferred embodiment, the automated voice processor collects speech inputs from the end user, translates them into machine-readable symbols, stores the speech inputs in a specified format in a designated location, and generates DTMF tones which are detected by the end user terminal and cause the end user terminal to initiate a process to identify a web page to be downloaded based on the stored and translated speech inputs, and then to download desired web page. The web pages may be static (that is, the web pages may be predefined and contain static information), or may be dynamic web pages that are modified, possibly on a per-call basis.

摘要翻译： 通信系统包括诸如IVR（交互式语音响应系统）的自动语音处理器，其可以通过诸如因特网的数据网络来控制与终端用户呼叫自动语音处理器相关联的显示。 在优选实施例中，自动语音处理器从最终用户收集语音输入，将它们转换为机器可读符号，将指定格式的语音输入存储在指定位置，并产生由终端用户终端检测到的DTMF音 并使终端用户终端基于所存储和转换的语音输入来发起识别要下载的网页的处理，然后下载所需的网页。 网页可以是静态的（即网页可以是预定义的并且包含静态信息），或者可以是基于每个呼叫被修改的动态网页。

公开(公告)号：US08117084B2

公开(公告)日：2012-02-14

申请号：US11671663

申请日：2007-02-06

申请人： Joseph Siegrist

发明人： Joseph Siegrist

IPC分类号： G06Q30/00 ,  G06F17/30

CPC分类号： H04M7/003 ,  G06Q30/016 ,  G06Q30/0601 ,  G06Q30/0613 ,  G06Q30/0641 ,  H04M3/51 ,  H04M2201/39 ,  H04M2201/60

摘要： A method and apparatus for establishing a phone call between a potential customer viewing information on the internet and a sales agent includes the steps of receiving a message indicating that the customer wishes to be contacted and including contact information for the customer and other information, converting at least some of the information in the message to speech, establishing a phone call to the sales agents, playing the speech to the sales agent, establishing a phone call to the customer using the contact information in the message, and bridging the first call to the second call. In some embodiments, a text message such as email or SMS is sent in addition to the phone call to the sales agent or when the phone call to the sales agent cannot be completed.

摘要翻译： 一种用于在互联网上的潜在客户观看信息和销售代理之间建立电话呼叫的方法和装置包括以下步骤：接收指示客户希望被联系的消息，并且包括客户的联系信息和其他信息，转换为 将消息中的一些信息发送到语音，向销售代理建立电话，向销售代理播放语音，使用消息中的联系人信息建立到客户的电话呼叫，并将第一个呼叫桥接到 第二个电话。 在一些实施例中，除了向销售代理的电话呼叫或者当不能完成对销售代理的电话呼叫之外，还发送诸如电子邮件或短信的文本消息。

公开(公告)号：US20090067413A1

公开(公告)日：2009-03-12

申请号：US12270226

申请日：2008-11-13

申请人： Ian Halpern ,  John Piescik ,  Joseph Siegrist

发明人： Ian Halpern ,  John Piescik ,  Joseph Siegrist

IPC分类号： H04L12/66

CPC分类号： H04L12/66

摘要： In an emergency backup telephone system, members of an enterprise use their personal computers to log into an emergency communications web page. Upon logging in, software that enables the personal computer to act as a webphone is automatically downloaded. This software allows a person to initiate a call from personal computer to a conventional PSTN number destination using a PSTN gateway, or to another party's computer at a specified URL using VoIP telephony. Upon logging in, an authoritative index of employees reachable via the backup system updated to include information such as a phone number and/or IP address where the member can be reached in order to allow calls originating from the PSTN to be routed to the member's computer. The index is made available to other members of the enterprise via the enterprise's intranet, and, in some embodiments, to the public via a web page on the internet and/or email.

摘要翻译： 在紧急备用电话系统中，企业的成员使用其个人计算机登录紧急通信网页。 登录后，将自动下载使个人计算机充当网络电话的软件。 该软件允许个人使用PSTN网关从个人计算机发起呼叫到常规PSTN号码目的地，或者使用VoIP电话在指定的URL处发起另一方的计算机呼叫。 登录后，通过备份系统可访问的雇员的权威索引被更新为包括诸如可以到达成员的电话号码和/或IP地址的信息，以便允许从PSTN发起的呼叫被路由到成员的计算机 。 该索引通过企业的内联网向企业的其他成员提供，并且在一些实施例中通过互联网和/或电子邮件上的网页向公众提供。

公开(公告)号：US07941839B2

公开(公告)日：2011-05-10

申请号：US12110020

申请日：2008-04-25

申请人： Joseph Siegrist ,  Ryan Cruse

发明人： Joseph Siegrist ,  Ryan Cruse

IPC分类号： G06F15/16

CPC分类号： H04L63/0281 ,  H04L63/029

摘要： A streaming media application attempting to establish a streaming media connection first attempts to establish the connection directly using a format such as UDP. If no direct connection can be established, the media application attempts to establish a connection through a proxy server using proxy server information obtained from installed software components such as browsers that manage Internet connections. If necessary, an auto configuration web page is utilized to obtain the proxy server address. The invention also includes methods for blocking streaming media connections.

摘要翻译： 尝试建立流媒体连接的流媒体应用首先尝试使用诸如UDP的格式直接建立连接。 如果不能建立直接连接，则媒体应用程序将尝试通过代理服务器建立连接，使用从已安装的软件组件（如管理Internet连接的浏览器）获取的代理服务器信息。 如果需要，使用自动配置网页来获取代理服务器地址。 本发明还包括阻止流媒体连接的方法。