Telephone number detection device and telephone number detection method

A VoIP-based system for IP networks identifies transferee numbers by analyzing voice data, addressing compatibility issues and maintaining accurate telephone number lists.

JP2025171667APending Publication Date: 2025-11-20JINTEC
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Patent Information

Application Number
JP2024077247
Authority / Receiving Office
JP · JP
Patent Type
Applications
Current Assignee / Owner
Filing Date
2024-05-10
Publication Date
2025-11-20

AI Technical Summary

Technical Problem

Existing telephone number cleaning devices are not compatible with IP networks, as they rely on ISDN infrastructure, and there is no direct method to obtain transferee numbers from IP networks, making it impossible to keep telephone number lists up to date.

Method used

A computer system connected to an IP network using VoIP procedures to send calls to a SIP server, analyze voice data for relocation guidance, and identify transferee numbers through amplitude value comparisons in voice data.

Benefits of technology

Enables automatic detection and updating of transferee numbers in IP network-connected fixed-line telephones, ensuring accurate and efficient management of telephone number lists.

✦ Generated by Eureka AI based on patent content.

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Abstract

To automatically obtain a relocation destination number for a fixed-line telephone connected to an IP network.SOLUTION: A telephone number detection device 1 is configured by a computer system and includes a control unit 11, a memory unit 12 that stores telephone numbers, and a communication interface 13b that mediates data communication between the control unit and an IP network 100. When a code corresponding to the relocation of a destination number is sent from a SIP server 103 at the time of sending a call whose destination number is a telephone number to be investigated and that is supposed not to be received to the SIP server, the control unit receives and records the audio data of relocation guidance sent from within the IP network by sending a call for voice communication with the destination number to the SIP server, analyzes the recorded audio data to identify a relocation destination number included in the relocation guidance, and stores the relocation destination number in the memory unit.SELECTED DRAWING: Figure 3
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Description

[Technical Field]

[0001] The present invention relates to a telephone number detection device and a telephone number detection method. [Background technology]

[0002] In various telephone-based businesses, customer phone number lists are extremely important information, and it is necessary to manage the information from time to time so as not to decrease its value, such as by deleting unnecessary phone numbers from the list. For example, if a phone number registered on the list becomes invalid because it is no longer in use or has been changed to a new phone number, such invalid phone number must be deleted or changed from the list.

[0003] The following Patent Document 1 describes a telephone number list cleaning method in which a number investigation computer connected to ISDN distinguishes valid telephone numbers in a list of telephone numbers to be cleaned, and the following Patent Document 2 describes a telephone number list cleaning device that is made up of a computer connected to ISDN and that, when a "call setup" message is sent to a network with a telephone number read from a telephone number list as the called number and the content of the cause indicator transferred from the network in response to the message is "change of destination subscriber number," picks up a new telephone number entered in a diagnostic information field attached to the cause indicator. Also, the following Non-Patent Document 1 provides an explanation of the specifications of SIP (Session Initiation Protocol) in relation to the present invention, and the following Non-Patent Document 2 describes the specifications of SDP (Session Description Protocol). [Prior art documents] [Patent documents]

[0004] [Patent Document 1] Patent No. 3462196 [Patent Document 2] Patent No. 3140967 [Non-patent literature]

[0005] [Non-Patent Document 1] Information and Communications Technology Committee, "Technical Report on Session Initiation Protocol (SIP)," [online], [Retrieved April 22, 2024], Internet<URL:https: / / www.ttc.or.jp / download_file / 3941 / 301> [Non-patent document 2] tex2e / rfc-translater, "RFC 8866 - SDP: Session Description Protocol Japanese Translation," [online], [Retrieved April 22, 2024], Internet<URL:https: / / tex2e.github.io / rfc-translater / html / rfc8866.html> Summary of the Invention [Problem to be solved by the invention]

[0006] The telephone number cleaning devices described in Patent Documents 1 and 2 both assume the use of integrated digital communication services via ISDN provided by Nippon Telegraph and Telephone Corporation (NTT), a telecommunications carrier. However, as is well known, as of January 1, 2024, the network for fixed-line telephone services provided by NTT will be IP (Internet Protocol), and the Public Switched Telephone Network (PSTN), including ISDN, will be completely migrated to an IP network. In other words, fixed-line telephone services have been switched from conventional telephone exchanges to the well-known Voice Over Internet Protocol (VoIP) system, which relays voice via IP. Therefore, in order to keep the telephone number list up to date by using a computer to determine the active / inactive status and transfer status of individual telephone numbers listed in the telephone number list, and to identify the transferee number, as in the telephone number cleaning devices described in Patent Documents 1 and 2, a new system compatible with IP networks is required. However, when it comes to telephone number transfers, it is not possible to directly obtain the data of the transferee telephone number from an IP network. In other words, it is fundamentally impossible to simply adapt the transferee number acquisition method described in Patent Document 2 to an IP network. Therefore, in order to obtain a ported number via an IP network, a method completely different from the conventional method for obtaining a ported number for ISDN is required.

[0007] SUMMARY OF THE INVENTION It is therefore an object of the present invention to provide a telephone number detection device and a telephone number detection method that can automatically obtain the ported number of a fixed-line telephone connected to an IP network. [Means for solving the problem]

[0008] In order to achieve the above object, the first aspect of the present invention is It is composed of a computer system that connects to an IP network and performs data communication using VoIP procedures. The device includes a control unit, a storage unit, and a communication interface, The storage unit stores telephone numbers to be investigated, the communication interface mediates data communication between the control unit and the IP network; The control unit a first call sending step of sending a call to a SIP server installed in the IP network, the call being the target telephone number stored in the storage unit as a destination number and not intended to be terminated at a telephone having the destination number; a second call originating step of originating a voice call to the destination number to the SIP server when a code corresponding to the transfer of the destination number is transmitted from the SIP server as a response to the first call originating step; a voice receiving step of receiving voice data of relocation guidance transmitted from within the IP network in response to the second call transmitting step; a guidance recording step of recording voice data of the relocation guidance; a relocation destination identification step of analyzing the recorded voice data to identify a relocation destination number included in the relocation guidance; a transferee number storage step of storing the identified transferee number in the storage unit; To execute It is a telephone number detection device.

[0009] The telephone number detection device may be configured such that, when a code indicating that the destination number is either a valid or invalid telephone number outside the IP network is returned as a response from the SIP server to the first call making step, the device associates this with the destination number and stores it in the memory unit.

[0010] When a code indicating that the destination number is either a valid or invalid telephone number outside the IP network is returned as a response from the SIP server to the first call sending step, the SIP server specifies communication outside the IP network and sends a call to the SIP server that will not be terminated at a telephone having the destination number; If the code returned from the IP network for the call is a code indicating validity, invalidity, or jurisdiction of another telecommunications carrier, the destination number is stored in association with the code, indicating that the destination number is a telephone connected to a metal line under the jurisdiction of the same telecommunications carrier as the IP network, that the destination number is invalid, or that the destination number is valid as a telephone under the jurisdiction of another telecommunications carrier. It can also be a telephone number detection device.

[0011] The storage unit stores sample data to be compared with the recorded voice data of the relocation guidance, The sample data describes time series changes in amplitude values ​​of a predetermined amount of data from the start of utterance in the relocation guidance for each of the wordings at the beginning, middle, and end of each of the message sentences of the plurality of types of relocation guidance and for each of the digits 0 to 9 that announce the telephone number in the relocation guidance; In the guidance recording step, when the voice data being received in the voice receiving step matches the sample data of the tail portion, recording of the voice data is stopped; In the relocation destination identification step, a relocation destination number included in the relocation guidance is identified based on a comparison between a transition of an amplitude value of the voice data recorded in the guidance recording step and the sample data. It can also be a telephone number detection device.

[0012] The scope of the present invention also includes a telephone number detection method for identifying a transferee number of a telephone number to be investigated using any of the telephone number detection devices described above. [Effects of the Invention]

[0013] According to the present invention, a telephone number detection device and a telephone number detection method are provided that can automatically obtain the ported number of a fixed-line telephone connected to an IP network. Other advantages will be made clear in the following description. [Brief explanation of the drawings]

[0014] [Figure 1] 1 is a diagram illustrating a configuration of an IP network including a telephone number detection device according to an embodiment. [Figure 2] 1 is a diagram illustrating a configuration of a telephone number detection device according to an embodiment. [Figure 3] 4 is a flowchart showing the operation of the telephone number detection device according to the embodiment. [Figure 4] 10 is a diagram showing an example of the contents described in the body of a SIP message that an INVITE control unit constituting the telephone number detection device according to the embodiment sends to the SIP server by the first call origination process. FIG. [Figure 5] 10 is a diagram showing an example of the contents written in the body of a SIP message that an INVITE control unit constituting the telephone number detection device according to the embodiment sends to the SIP server by the second call origination process. FIG. [Figure 6] 10 is a diagram showing voice data of a transfer message recorded by an RTP receiving unit that constitutes the telephone number detection device according to the embodiment. FIG. [Figure 7] FIG. 4 is a diagram showing sample data stored in a storage unit of the telephone number detection device according to the embodiment. [Figure 8] 10 is a diagram showing an example of the contents described in the body of a SIP message that an INVITE control unit constituting the telephone number detection device according to the embodiment sends to the SIP server by the third call origination process. FIG. DETAILED DESCRIPTION OF THE INVENTION

[0015] The present invention will be described with reference to the accompanying drawings. In the drawings used in the following description, the same or similar parts are designated by the same reference numerals, and redundant explanations may be omitted. If a part designated by a reference numeral in one drawing is unnecessary, the same reference numeral may not be designated in other drawings. ===Example=== The telephone number detection device of the embodiment is configured by a computer system, and targets fixed telephones (hereinafter referred to as "telephones") connected to an IP network (hereinafter referred to as "Hikari Telephone Network") for using NTT's IP telephone service "Hikari Telephone," to detect whether telephone numbers listed in the telephone number list have been transferred, and if a transfer is detected, it identifies the telephone number of the transferred number (hereinafter referred to as "transferred number"). <Network configuration> Figure 1 shows devices connected to a Hikari telephone network 100, including a telephone number detection device 1 according to an embodiment. The telephone line 2 drawn into the home of a user of a telephone 10 includes an optical fiber line 2a and a copper wire line 2b. The telephone 10 may be a conventional analog telephone 10a, a telephone 10b equipped with the functionality of a VoIP gateway 3 called an IP telephone, or a personal computer 10c equipped with dedicated software and functioning as an IP telephone.

[0016] For convenience, we will refer to a telephone 10 that makes calls via an optical fiber line 2a drawn into a home as an optical IP telephone, and a telephone 10 that makes calls via a metal line 2b as a metal IP telephone. When a conventional analog telephone 10a is used as a Hikari IP telephone, the telephone 10a is installed in the home and connected to the optical fiber line 2a via a VoIP gateway 3, a router 4, and an optical network unit (hereinafter sometimes referred to as "ONU 5"). To connect a Hikari IP telephone to the Hikari telephone network 100, a device called a "home gateway" is generally used, which aggregates all of the functions of the VoIP gateway 3, the router 4, and the ONU 5, or aggregates all of the functions excluding the ONU 5.

[0017] On the other hand, a metal IP telephone consisting of an analog telephone 10a is directly connected to a metal line 2b drawn into the home, and is also connected to the Hikari telephone network 100 via a metal accommodation device 6 converted from a local switch in the PSTN, and a conversion device 7 equivalent to a VoIP gateway. The telephone number detection device 1 according to the embodiment is a computer system that is equipped with the functionality of a VoIP gateway and is configured to be able to communicate data with the Hikari telephone network 100 using VoIP procedures, and is connected to the Hikari telephone network 100 via a router 4 and an ONU 5. In FIG. 1, the boundary between the inside and outside of the home of the user of telephone 10, or the inside and outside of the facility where telephone number detection device 1 is installed, is indicated by a dashed line 8.

[0018] The Hikari Telephone Network 100 is equipped with accommodation routers 101 and relay routers 102, which correspond to the accommodation switch and relay switch of the PSTN. There are many accommodation routers 101 and relay routers 102 within the Hikari Telephone Network 100. Furthermore, IP-based telephone networks such as the Hikari Telephone Network 100 are equipped with a unique feature not found in the PSTN: a SIP server 103 that performs call control based on SIP, which handles call control for outgoing, incoming, answering, and disconnecting calls. In the Hikari Telephone Network 100, calls (INVITE) from telephones 10 to destinations specified by telephone numbers are collected by the accommodation router 101 and reach the SIP server 103 via relay routers 102 and other devices. The SIP server 103 uses IP address resolution to obtain an IP address or the like corresponding to the destination telephone number (hereinafter sometimes referred to as the "destination number"), and then forwards the INVITE to the relay router 102 associated with the telephone 10 of the destination number. When the telephone 10 of the destination number responds to the INVITE, the call becomes possible via the Hikari telephone network 100. The Hikari telephone network 100 and the IP network 200 of the other telecommunications carrier are connected via an interconnection router 104.

[0019] The telephone number detection device 1 of the embodiment performs data communication with the SIP server 103, and if a telephone number listed in the telephone number list has been transferred based on the data communication with the SIP server 103, it obtains the telephone number of the transferred telephone number and updates the telephone number list. <Configuration of telephone number detection device> 2 shows a functional block configuration of a telephone number detection device 1 according to an embodiment. The telephone number detection device 1 can be configured by a computer equipped with, for example, a control unit 11 including a CPU, RAM, and ROM, a storage unit 12 configured as an external storage (SSD, HDD, etc.), and a communication interface 13 that complies with a predetermined standard (such as the IEEE 802.11 standard). The telephone number detection device 1 includes a main processing unit 111, an INVITE control unit 112, an RTP receiving unit 113, a voice analysis unit 114, and a Web server unit 115, which are realized when the control unit 11 executes a predetermined program.

[0020] The storage unit 12 stores a list of telephone numbers and various data that the telephone number detection device 1 generates while executing a predetermined program or that it acquires through data communication via the communication interface 13. The storage unit 12 also stores various data that is processed by the web server unit 115, such as data that is the source of generating a web page.

[0021] The communication interface 13 is provided with a user communication interface (hereinafter sometimes referred to as "user communication I / F 13a") provided for data communication with a computer (hereinafter sometimes referred to as "user terminal 40") operated by a user who views or edits the telephone number list, and a detection communication interface (hereinafter sometimes referred to as "detection communication I / F 13b") provided for communication with the Hikari telephone network 100. The detection communication I / F 13b is provided with eight virtual I / Fs per physical I / F, which serve as communication ports with the Hikari telephone network 100. In this embodiment, the telephone number detection device 1 is provided with a maximum of 12 physical I / Fs, so the telephone number detection device 1 can make calls in parallel to a total of 96 telephone numbers.

[0022] The web server unit 115 accepts instructions for telephone number research and the like via a web page and the browser of the user terminal 40. Alternatively, in response to an HTTP request from the user terminal 40, it returns a web page that enables, for example, uploading or deleting a telephone number list.

[0023] The main processing unit 111, the INVITE control unit 112, and the RTP receiving unit 113 are functional components that are in the relationship of parent process, child process, and grandchild process, respectively. In general terms, the main processing unit 111 has a function of forwarding telephone numbers read from a telephone number list to a child process while monitoring the availability of the detection communication I / F 13b. The INVITE control unit 112 performs data communication with the SIP server 103, and performs call control for telephone numbers received from the parent process and acquires various codes transferred from the SIP server 103 in response to calls. The RTP receiving unit 113 is activated when the INVITE control unit 112 starts call control, and receives audio data transmitted from the Hikari telephone network 100 in response to calls originated by the child process, based on the Real-time Transport Protocol (RTP), a protocol for playing data such as audio and video in real time. As described below, the voice analysis unit 114 has a function of analyzing the voice data of the guidance notifying the relocation (hereinafter referred to as "relocation guidance") received by the RTP receiving unit 113, for example, using well-known voice recognition technology. <Operation of the phone number detection device> FIG. 3 is a diagram showing the flow of operation of the telephone number detection device 1. Hereinafter, with reference to FIGS. 2 and 3, information processing and the like accompanying the operation of the telephone number detection device 1 will be described. The storage unit 12 stores a file of a telephone number list that lists telephone numbers to be investigated for transfers, etc. When a telephone number list to be investigated is specified based on an instruction from the user terminal 40, the main processing unit 111 reads the file of the telephone number list from the storage unit 12 (s1). Then, the main processing unit 111 sequentially reads out the telephone numbers listed in the list (s2). At this time, if there is a vacant line in the detection communication I / F 13b, the main processing unit 111 transfers the read telephone numbers to the INVITE control unit 112 (s3, s4 → s6). If there is no vacant line in the detection communication I / F 13b, the main processing unit 111 waits for a predetermined time (e.g., 100 ms) and continues to wait until there is a vacant line in the detection communication I / F 13b (s3, s4 → s5 → s4).

[0024] Upon receiving the telephone number from the main processing unit 111, the INVITE control unit 112 activates the voice data reception process (first RTP reception process) by the RTP receiving unit 113 (s6), and then performs a call (first call) using the received telephone number as the destination number to the telephone set 10 at that destination number in a manner that prevents the call from being received (s7). In summary, the INVITE control unit 112 specifies a data transfer method for the call to the SIP server that is different from a normal call (e.g., AVPF (Audio-Visual Profile with Feedback)) and transmits a connection request to the telephone set 10 at the destination number. As is well known, in VoIP, SIP messages sent by IP telephone sets 10b and the like to the SIP server 103 have a basic structure consisting of a start line, header information, and a body portion separated by a blank line, similar to HTTP. SIP defines only the basic structure, start line, and header information of this SIP message. Furthermore, in VoIP, session information such as the IP address and compression format of media streaming such as voice is included in the body portion of the SIP message using the SDP syntax. The SIP specifications are defined by RFC3261, and an explanation of the SIP specifications is provided in the above-mentioned Non-Patent Document 1. The SDP specifications are defined by RFC8866, and a translation thereof is provided in the above-mentioned Non-Patent Document 2. For reference, Fig. 4 shows an example of the content described in the body section of a SIP message for a first call in SDP syntax.

[0025] When a SIP message specifying a destination number is sent, the SIP server 103 returns a code indicating the status of the destination number to the sender of the message. If the status indicated by the code transferred from the SIP server as a SIP response to the first call (s7) is a status issued when the destination number is transferred, the INVITE control unit 112 starts the RTP receiving process (second RTP receiving process) again (s8 → s9 → s10 → s12) and makes a call (second call) to the same destination number using a SIP message for a normal voice call (s13). Figure 5 shows an example of the content described in the body of the SIP message in SDP syntax for the second call.

[0026] In SIP, the code 410 indicates "Gone" and is also called a "guidance connection error." This code is transferred when "Gone: The user was previously present but is no longer available here." In the Hikari telephone network 100, this code "410" corresponds to the code issued when the destination number is transferred. If a preset time has elapsed after the first call (s7) and a timeout occurs (s8: YES), or if the SIP response to the first call (s7) is a code other than 410 (s8 → s9 → s10: No), the first RTP receiving process is terminated (s11), and a result file containing the received code, information indicating that the timeout has occurred, and the destination number is created and stored in the storage unit 12 (s19). When the INVITE control unit outputs the result file (s19), the main processing unit 111 reads the next telephone number in the telephone number list as the target of investigation (s2).

[0027] On the other hand, as described above, if the INVITE control unit 112 receives a 410 code as a SIP response (s10: YES), it makes a second call (s13) to the same destination number as the first call (s7). Naturally, since this destination number has already been transferred, the call will not arrive at the telephone set 10 that has that destination number. Instead, the Hikari telephone network 100 sends audio data of relocation guidance to the telephone number detection device 1, informing the telephone number detection device 1 that the destination number has been transferred. The RTP receiving unit 113 receives the audio data of this relocation guidance (s14).

[0028] Specifically, the RTP receiving unit 113 writes received data, including voice data and an RTP header, into the created raw data file until the preset end timing of the relocation guidance. When recording of the received data is complete, the RTP receiving unit 113 converts the voice data in the raw data file into a message voice file in a predetermined format such as WAV format, and stores the message voice file in the storage unit 12. The RTP receiving unit 113 also transmits a signal (e.g., SIGUSR2) instructing a predefined call disconnection request to the INVITE control unit 112. In response to the disconnection request from the RTP receiving unit 113, the INVITE control unit 112 sends a "CANCEL" SIP message to the Hikari telephone network 100 to terminate the call, thereby terminating the call (s15→s16).

[0029] The timing at which the RTP receiving unit 113 ends recording the relocation guidance voice data can be, for example, the point in time after starting recording the voice data that is long enough to receive at least one instance of the relocation guidance repeatedly transmitted from the Hikari telephone network 100. The relocation guidance message text is not limited to one, but includes multiple types. A user who has ported their phone number selects one of the multiple relocation guidance texts when applying for the relocation number notification service. Therefore, the recording time for the relocation guidance can be set to a time that ensures one instance of the relocation guidance is recorded, regardless of the type of relocation guidance. By stopping reception after the elapse of that time, the reception time can be minimized. Of course, recording of the voice data can also be stopped when real-time voice recognition detects that the relocation guidance has been repeated. An example of real-time voice recognition of the relocation guidance will be described later.

[0030] When the INVITE control unit 112 ends the call (s16), it transfers the message voice file created by the RTP receiving unit 113 to the voice analysis unit 114. The voice analysis unit 114 analyzes this message voice file using voice recognition technology or the like (s17), extracts the voice data of the portion announcing the ported number, and identifies the ported number (s18). Then, a result file including the ported number is created and output (s19). The created result file is stored, for example, in the storage unit 12, and later used for telephone number list update processing, etc., by the main processing unit 111 or a function executed by another program. That is, the ported number is added to the field of the ported telephone number in the telephone number list, or the ported number is added to the telephone number list as a new telephone number. Of course, the ported telephone number may be replaced with the ported telephone number. The telephone number list update processing may be performed in parallel with the investigation of other telephone numbers when the ported number is acquired or when the result file is created. <Speech recognition method for porting number> The telephone number detection device 1 of this embodiment identifies the ported number based on the voice data of the relocation guidance returned from the Hikari telephone network 100. As described above, well-known voice recognition technology can be used to identify the ported number from the relocation guidance. However, if the content of the relocation guidance were to be analyzed point by point using advanced voice recognition technology that can handle unspecified natural languages, the processing load would be heavy. Therefore, since the voice message informing the ported number is standardized and the voice that needs to be identified is only the digits of the ported number, the telephone number detection device 1 of this embodiment identifies the ported number based on the time-series changes in the amplitude values ​​in the voice data.

[0031] Specifically, the voice used in the Japanese telephone network is G.711 (PCMU / 8000), which is expressed as one byte of numerical data per 1 / 8000 seconds, and the amplitude of the voice is indicated by a value between 0 and 255. This amplitude value consisting of a value between 0 and 255 is expressed in hexadecimal, and samples are taken of 16 consecutive bytes extracted from standard parts of voice messages or from voices corresponding to the digits 0 to 9, and each sample is stored in advance in storage unit 12 as a definition file. The definition file can be created by recording the voice data of a large number of voice guidances and identifying what is being spoken at which part of the time-series change in amplitude from the transition of the amplitude value of the voice signal. The message for the transfer guidance informing the caller of the transfer destination number uses "x" or "y" as a string of digits to indicate the phone number, such as "Dear caller x, your number has changed due to transfer. Your new number is y," "Dear caller x, your phone has been disconnected due to transfer. Your contact number is y," or "Dear caller x, your number has changed. Your new number is y." Audio data for all types of messages and audio data for each of the digits 0 to 9 that are common to all types of messages are recorded in advance, and the above definition file is created based on the recorded audio data.

[0032] Figure 6 shows a time series change 50 in amplitude values ​​of voice data for one of the message sentences in the relocation guidance, "Thank you for calling, x. Your phone has been disconnected due to relocation. Your contact number is y." Also, Figure 7 shows examples of sample data 151 corresponding to the beginning portion 51 of the relocation guidance, "Thank you for calling," sample data 152 corresponding to the middle portion 52 of "has been disconnected due to customer circumstances. Your contact number is," which is inserted between the relocation source number x and the relocation destination number y, sample data 153 corresponding to the end portion 53 of "is," and sample data for each of 0 to 9 (xy0 to sy9).

[0033] In the example of the amplitude transition of the entire relocation message shown in Figure 6, the relocation source number x is a 10-digit number and the relocation destination number y is an 11-digit number. It can also be seen that the beginning portion 51 of the relocation guidance, the relocation source number x, the middle portion 52 ("has disconnected the phone due to customer circumstances") 52a, the middle portion 52b ("The contact number is...") 52b, and the end portion 53 are intermittently arranged. It can also be seen that the beginning portion 51 of the relocation guidance is played for the second time when the relocation guidance is listened to to the end. Furthermore, as can be seen from Figure 6, in reality, the durations during which each digit in the telephone number (x, y), the beginning portion 51, the middle portion 52, and the end portion 53 are spoken are not uniform, and the amount of data varies. However, as shown in Figure 7, in the sample data, each portion (51-53) of the message and each digit in the telephone number (x, y) are written with a predetermined amount of data starting from the start of the speech.

[0034] The maximum recording time for the relocation guidance by the RTP receiving unit 113 is set to a time (for example, 45 seconds) that allows some leeway for the time when the second relocation guidance starts to be played, and then, when the INVITE control unit 112 makes a second call, the RTP receiving unit 113 starts recording the audio data of the relocation guidance with the set time as the maximum time, while the audio analysis unit 114 compares the audio data in real time with the sample data described in the definition file.

[0035] The voice analysis unit 114 increments a counter by "1" each time it detects voice data corresponding to the beginning portion 51, which is common to all types of relocation guidance, and outputs the number of times the beginning portion has been detected. When the counter reaches "2," the RTP receiving unit 113 terminates (exits) the voice data reception process and, as described above, sends a disconnection request (SIGUSR2) to the INVITE control unit 112. In response to this request, the INVITE control unit 112 sends a CANCEL to the SIP server 103 to terminate the call. This shortens the voice data recording time by the RTP receiving unit 113 compared to a preset time, thereby reducing the time required for the series of processes from when the main processing unit 111 reads the telephone number to when the RTP receiving unit 113 terminates reception of the voice data. In other words, it is possible to investigate more telephone numbers in a shorter time. To identify the relocated number, all recorded voice data of the relocation guidance is reanalyzed and the relocated number is identified based on the amplitude values ​​of the voiced digits between the middle portion 52 and the end portion 53 of the message. Furthermore, according to the telephone number detection device 1 of the embodiment, when identifying a transferee number, the voice data of a standard message text is analyzed only based on the transition of amplitude values ​​defined by a predetermined amount of data, thereby reducing the load on the voice analysis unit 114. ===About SIP responses=== As is well known, the SIP server 103 sends various codes as SIP responses in response to incoming outgoing calls. In other words, the codes returned in the SIP response to the first outgoing call are not limited to the above-mentioned 410. Some SIP responses are a combination of a code and a "warn-code" added to the warning header field. Therefore, the telephone number detection device 1 may detect a status related to the destination number other than porting, depending on a code other than the 410 code or a combination of a code and a warn-code.

[0036] For example, if the code or the combination of the code and the warn-code is 488 (warning code: none), it indicates that the destination number is the number of a telephone 10 that uses the Hikari Telephone service (Hikari Telephone valid), and if the code is 404, it indicates that "Hikari Telephone invalid." If the combination of the code and the warn-code is 488 (warning code: 302), it indicates that the destination number is a telephone number that belongs to the IP network 200 of another telecommunications carrier (other network).

[0037] Furthermore, details about destination numbers detected as "Hikari Telephone Invalid" or "Other Network" can also be detected. Specifically, for destination numbers detected as "Hikari Telephone Invalid" or "Other Network," a call (hereinafter sometimes referred to as a "third call") is sent to the SIP server 103 in which the SDP syntax content to be written in the body of the SIP message specifies a method of preventing the destination number from receiving calls and a call to an outside of the Hikari Telephone Network 100. Figure 8 shows an example of the SDP syntax content to be included in the SIP message for the third call. Then, for this third call, a code is returned from the interconnection router 104 or the conversion device 7, which is a gateway connecting the inside and outside of the Hikari Telephone Network 100. If the code is 500, it is determined that the destination number belongs to a metal IP phone under the jurisdiction of the same telecommunications carrier as the Hikari Telephone Network 100. If the code is 488 (warning code: 300), it is determined that the destination number is currently being used by another IP network 200 via number portability. ===Other Examples=== Although one embodiment of the present invention has been described above, the present invention is not limited to the above-described embodiment and includes various modifications. Furthermore, for example, the above-described embodiments have been described in detail to clearly explain the present invention, and the present invention is not necessarily limited to those including all of the described configurations. Furthermore, it is possible to add, delete, or replace part of the configuration of the telephone number detection device 1 according to the embodiment with another configuration.

[0038] For example, the telephone number detection device according to the above embodiment is configured with one computer, but it may be configured with multiple computers that are distributed over a LAN or the Internet and linked to each other, and the functions of the main processing unit 111, INVITE control unit 112, RTP receiving unit 113, voice analysis unit 114, etc. may be realized by individual computers. Alternatively, the functions of the telephone number detection device 1 may be appropriately combined and distributed among multiple computers, such as by implementing the function of the voice analysis unit 114 and other functions in separate computers.

[0039] In the above embodiment, when the INVITE control unit 112 terminates the call, it analyzes the audio data of the entire transfer message recorded by the RTP receiving unit 113 to identify the transferee number. However, when the call is terminated, it may output a result file indicating that the destination number has been transferred and audio data has been recorded, and after investigating all telephone numbers listed in the telephone number list, it may identify the transferee number.

[0040] The second call may be made at a different time from the first call. For example, the phone numbers that returned code 410 after the first call may be listed, and a second call may be made later with the phone number to be transferred as the destination number to identify the ported number.

[0041] If a call is made based on RFC-compliant specifications so that the destination number is not reached, the SDP syntax content included in the SIP message for the first call is not limited to that shown in Figure 4. The SDP syntax content included in the second and third call messages is also not limited to that shown in Figures 5 and 8.

[0042] In the above embodiment, first, a telephone number list containing multiple telephone numbers to be investigated is read, and telephone numbers are extracted from the list in order to be investigated, but it is also possible to read a specific telephone number as the investigation target from the storage unit based on, for example, an instruction from the user terminal 40. In any case, a large number of telephone numbers are stored in the storage unit 12, and it is sufficient to investigate whether or not a telephone number has been relocated using that telephone number as the destination number.

[0043] In the SIP response of the SIP server 103 of the Hikari telephone network 100 targeted by the telephone number detection device 1 according to the embodiment, code 410 indicates the transfer of the destination number, but for the SIP servers 103 of other IP networks, a code other than 410 may indicate the transfer. Therefore, when applying the telephone number detection device 1 according to the embodiment to other IP networks, it is necessary to investigate in advance the code of the SIP response corresponding to the transfer of the destination number, and if that code is returned in response to the first call outgoing, thereafter the transferred number can be identified in the same manner as in the above embodiment. [Explanation of symbols]

[0044] 1 phone number detector, 2 telephone lines, 4 routers, 5 Optical Line Terminal (ONU), 10 Telephone, 11 Control Unit, 12 Memory Unit, 13 Communication interface, 100 IP network (Hikari telephone network), 103 SIP server, 111 main processing unit, 112 INVITE control unit, 113 RTP receiver, 114 voice analyzer, 200 IP network of other telecommunications carriers

Claims

1. The system is configured by a computer system that connects to an IP network and performs data communication using VoIP procedures, The device includes a control unit, a storage unit, and a communication interface, The storage unit stores telephone numbers to be investigated, the communication interface mediates data communication between the control unit and the IP network; The control unit a first call sending step of sending a call to a SIP server installed in the IP network, the call being the target telephone number stored in the storage unit as a destination number and not intended to be terminated at a telephone having the destination number; a second call originating step of originating a voice call to the destination number to the SIP server when a code corresponding to the transfer of the destination number is transmitted from the SIP server as a response to the first call originating step; a voice receiving step of receiving voice data of relocation guidance transmitted from within the IP network in response to the second call transmitting step; a guidance recording step of recording voice data of the relocation guidance; a relocation destination identification step of analyzing the recorded voice data to identify a relocation destination number included in the relocation guidance; a transferee number storage step of storing the identified transferee number in the storage unit; To execute Phone number detection device.

2. 2. The telephone number detection device according to claim 1, wherein when a code indicating that the destination number is either a valid or invalid telephone number outside the IP network is returned as a response from the SIP server to the first call outgoing step, the device associates this with the destination number and stores it in the memory unit.

3. When a code indicating that the destination number is either a valid or invalid telephone number outside the IP network is returned as a response from the SIP server to the first call sending step, a call is sent to the SIP server specifying communication outside the IP network and not to be terminated at a telephone having the destination number; If the code returned from the IP network for the call is a code indicating validity, invalidity, or jurisdiction of another telecommunications carrier, the destination number is stored in association with the code, indicating that the destination number is a telephone connected to a metal line under the jurisdiction of the same telecommunications carrier as the IP network, that the destination number is invalid, or that the destination number is valid as a telephone under the jurisdiction of another telecommunications carrier.

3. The telephone number detection device according to claim 2.

4. The storage unit stores sample data to be compared with the recorded voice data of the relocation guidance, The sample data describes time series changes in amplitude values ​​of a predetermined amount of data from the start of utterance in the relocation guidance for each of the wordings at the beginning, middle, and end of each of the message sentences of the plurality of types of relocation guidance and for each of the digits 0 to 9 that announce a telephone number in the relocation guidance; In the guidance recording step, when the voice data being received in the voice receiving step matches the sample data of the tail portion, recording of the voice data is stopped; In the relocation destination identification step, a relocation destination number included in the relocation guidance is identified based on a comparison between a transition of an amplitude value of the voice data recorded in the guidance recording step and the sample data.

2. The telephone number detection device according to claim 1.

5. A telephone number detection method for identifying a ported number of a telephone number to be investigated using the telephone number detection device according to any one of claims 1 to 4.

Citation Information

Patent Citations

  • Transfer information database generating system and method therefor

    JP1997270851A

  • Telephone number investigation device, method, program, and information providing system

    JP6209762B1

  • Phone list cleaning device and cleaning method

    JP3140967B2

  • How to clean your phone number list

    JP3462196B2