Abstract: A method for dialing between Internet extensions is disclosed. When dialing between Internet extensions, just dial the switchboard phone number of SIP proxy server plus “-” and then dial the extension phone number of the opposite Internet extension directly. It is not necessary to use a voice guidance for asking dialing of the extension phone number of the opposite Internet extension.

Abstract: The present invention provides a “Phone Log” in a telephone switchboard for recording the extension numbers, the phone numbers dialed by the extension numbers and the accumulated times. When there is an outside call, the telephone switchboard will first check the “Phone Log” whether the telephone number of the outside call is on the “Phone Log” or not; if the telephone number of the outside call is on the “Phone Log”, then the outside call (callback) is connected directly to the corresponding extension; otherwise the outside call will be connected to an auto-attendant for connecting. As the accumulated time of each of the records reaches a time limit, the related record will be cleared out. After the callback is picked up by the corresponding extension successfully, the related record is also cleared out.

Abstract: A method, an apparatus, and a computer program product thereof for enabling an Internet extension to ring a conventional extension are disclosed. The apparatus comprises an SIP proxy and an RTP relay. The SIP proxy receives a calling request from the Internet extension, substitutes an exchange number for a conventional extension number comprised in the calling request, and transmits the substituted calling request to a gateway, so that the substituted calling request can be transmitted to a voice automatic machine via a telephone exchange. The RTP relay transmits the conventional extension number in DTMF format to the gateway. Thus, this invention enables the Internet extension to ring the conventional extension with lower hardware costs.

Abstract: A method for dialing from Internet extension to conventional extension is disclosed. A VoIP gateway or an IP auto attendants is used for dialing from Internet extension to conventional extension. The phone number of the Private Branch Exchange and the voice guidance are not needed. The calling number of SIP message is interpreted directly and converted into DTMF (Dual-tone multi-frequency) messages for dialing into a conventional extension.

Abstract: A method for dialing between Internet extensions is disclosed. When dialing between Internet extensions, just dial the switchboard phone number of SIP proxy server plus “-” and then dial the extension phone number of the opposite Internet extension directly. It is not necessary to use a voice guidance for asking dialing of the extension phone number of the opposite Internet extension.

Abstract: The present invention provides an NAT (Network Address Translator) traversal method in Session Initiation Protocol (SIP) for solving the problems of SIP in Internet phone (VoIP) under current Internet environment. In other words, the present invention solves the SIP problems caused by NAT (Network Address Translator) that P2P (Peer to Peer) transmission cannot traverse the NAT firewall directly.

Abstract: A modified NAT firewall traversal method for SIP communication is based on the common SIP network phone communication protocol, and is aided by Interactive NAT Traversal (INT) and pre-established media session ideas to accomplish the object of transversing NAT firewall. Users of private IP located within different NAT firewalls can therefore directly transmit voice packets by means of peer-to-peer transmission without the need of any proxy server of voice packets.

Abstract: A method, an apparatus, and a computer program product thereof for enabling an Internet extension to ring a conventional extension are disclosed. The apparatus comprises an SIP proxy and an RTP relay. The SIP proxy receives a calling request from the Internet extension, substitutes an exchange number for a conventional extension number comprised in the calling request, and transmits the substituted calling request to a gateway, so that the substituted calling request can be transmitted to a voice automatic machine via a telephone exchange. The RTP relay transmits the conventional extension number in DTMF format to the gateway. Thus, this invention enables the Internet extension to ring the conventional extension with lower hardware costs.

Abstract: The present invention provides a flash disk of phone book, it comprises an autorun part, a phone book part and a free use part. The autorun part includes programs for displaying related information on a monitor of a computer when a subscriber plugs the flash disk of phone book into the computer. The phone book part comprises many webpages and phone numbers of many companies and individuals for being displayed on the monitor of the computer. XML or HTML can be used for developing webpages. The free use part is just the same as an ordinary flash disk, and can be used by the subscriber freely. The phone book part sets up an ActiveX component on each webpage of the companies and individuals. Each ActiveX component includes VoIP related software and a service phone number of a company or an individual.

Abstract: The present invention provides a warranty service system. It comprises a flash disk for warranty service, a computer, Internet and a registration center. The registration center is a computer server having a web address and a data base. A customer plugs the flash disk for warranty service into the computer and links with the registration center via Internet to achieve a warranty registration. The warranty service system of the present invention further comprises an information center, which is a computer server having a web address and a data base. The information center has two functions: one is to provide product advertising information to the customer; the other is to connect a service phone of a company for the customer.

Abstract: The present invention provides a web call method so that when a user uses a personal computer to make an Internet call, the user just inputs a web address through browser instead of key in telephone number by keyboard.

Abstract: A method for enabling an autodial via Internet comprises the steps of: designing a VoIP related software; packaging the VoIP related software into an Active-X element; buying CodeSign from Microsoft for being added into the Active-X element for a digital signature, and acquiring a certified ID from Microsoft; and installing the Active-X element in a webpage. After the webpage is sent to a user's computer display via the Internet, the Active-X element will appear as an icon at a specific position in the webpage for enabling the user to click, after clicking the Active-X element with the mouse, the VoIP related software will be installed in the user's computer and will automatically guide the user to a specific telephone number.

Abstract: The present invention provides an interactive NAT (Network Address Translator) traversal method, i.e. INT (Interactive NAT Traversal) method, for solving the problems of SIP (Session Initiation Protocol) in Internet phone (VoIP) under current Internet environment. In other words, the present invention solves the SIP problems caused by NAT (Network Address Translator) and private (virtual) IP, so that P2P (Peer to Peer) transmission can traverse the NAT firewall directly. The present invention uses the INT method on SIP, forming an interactive NAT traversal method in SIP (SIP with INT, SWINT), to solve the problems that SIP speech packets cannot transmit directly under NAT firewall.

Abstract: A modified NAT firewall traversal method for SIP communication is based on the common SIP network phone communication protocol, and is aided by Interactive NAT Traversal (INT) and pre-established media session ideas to accomplish the object of transversing NAT firewall. Users of private IP located within different NAT firewalls can therefore directly transmit voice packets by means of peer-to-peer transmission without the need of any proxy server of voice packets.

Abstract: A direct P2P transmission protocol between two virtual networks is disclosed in the invention, which is implemented on the Internet and includes three procedures: the login procedure, the testing Network Address Translation (NAT) procedure, and the direct data transmission procedure. Among them, the location server located in the real Internet is to support the login procedure and the testing NAT procedure, and then the direct data transmission procedure can be executed after the detection of NAT-based router parameters has been completed. The direct data transmission procedure, however, can do direct data packet transmission between two virtual IP networks without support from any proxy server. In addition, because the login procedure almost occupies none of the bandwidth resource and the testing NAT procedure takes very little connection time for completion, the direct data transmission between two virtual IP points can thus be executed with plenty of time and in large quantity.

Abstract: A system for converting a hyper text markup language (HTML) document to speech includes an HTML parser, an HTML to speech (HTS) control parser, a tag converter, a text normalizer and a TTS converter. The HTML parser receives data of an HTML formatted document and parses out content text, HTML text tags that structure the content text and control rules used only for translating the received data into sound. The HTS control parser parses control rules for converting the received data into sound. The HTS control parser modifies entries in one or more of a tag mapping table, an audio data table, a parameter set table, an enunciation modification table and a terminology translation table depending on each of the parsed control rules. The text normalizer modifies enunciation of each text string of the content text of the HTML document for which the enunciation modification table has an entry, according to an enunciation modification indicated in the respective enunciation table entry.