JP2008219621A - Telephone with call recording function, and its information processing method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a function of recording audio information without requiring a special device, and transferring the recorded audio information to other telephones so that a specific person can listen to the audio information. <P>SOLUTION: The telephone has a setting data storage part 219 for storing identification of a communication device for connecting via an IP network 15, an audio data control part 210 for recording a voice message from the arbitrary telephone connected in response to a termination call in a voice data storage part 208, and a central control part 214 for transferring the recorded voice message recorded in the voice data storage part 208 to the telephone corresponding to a specific IP address of the setting data storage part 219. <P>COPYRIGHT: (C)2008,JPO&INPIT

Description

  The present invention relates to a telephone with a call recording function and an information processing method thereof, and more particularly to a telephone with a call recording function for transferring recorded voice information to another apparatus and an information processing method thereof.

  Conventionally, in order to record a call, two call recording systems have been adopted. The first call recording system records directly on the telephone like an answering machine. The second call recording system is to record in a special device such as a call recording device, listen to data that is managed by a server and stored by a specific person on the server.

  As a first call recording system, a patent application has been filed for a telephone with a call recording function. According to this patent application, in order to be able to always store a certain continuous call data in the digital data storage unit, for example, in FIG. 1 and FIG. When storing the data in the data storage unit 170, the end position storage unit 250 is referred to, the data stored therein is stored as the storage start position in the digital data storage unit 170, and the value of the end position storage unit 250 is stored. Is updated to a value indicating the position where the digital data is stored next. When the value of the end position storage unit 250 exceeds the value of the area securing unit 210, the value of the end position storage unit 250 is set to the first value “0”. When the value of the end position storage unit 250 is the same as the value of the start position storage unit 240, the value of the start position storage unit 240 is updated. At this time, if the value of the start position storage unit 240 exceeds the value of the area securing unit 210, the value of the start position storage unit 240 is reset to the initial value “0” (see Patent Document 1).

  FIG. 14 is an example of a second call recording device system and is a configuration diagram showing a call recording system using a conventional IP telephone. In FIG. 14, the A site 61A is connected to the public network 63 via the public line 62A, and the B site 61B is connected to the public network 63 via the public line 62B. The A site 61A is connected to the IP network 65 via an access line 64A which is a dedicated line, and the B site 61B is connected to the IP network 65 via an access line 64A which is a dedicated line. The IP network 65 is connected to a data center 67 through an access line 66 that is a dedicated line.

  The A site 61A includes a plurality of IP telephones 601, a PSTN gateway 602, a call recording device 603, an MDF (main wiring board) 604, and a router 605. Inside the A site 61A, the IP telephone set 601, the PSTN gateway 602, the call recording device 603, and the router 605 are connected by a network.

  The B site 61B includes a plurality of IP phones 606, a PSTN gateway 607, a call recording device 608, an MDF 609, and a router 610. Inside the B base 61B, the IP telephone 606, the PSTN gateway 607, the call recording device 608, and the router 610 are connected by a network.

  The data center 67 has a SIP server 611, a call recording server 612, and a router 613. Within the data center 67, the SIP server 611, the call recording server 612, and the router 613 are connected via a network.

  Therefore, the A base 61A, the B base 61B, and the data center 67 are connected by the router 605, the router 610, and the router 613 through the access line 64A, the access line 64B, the access line 66, and the IP network 65, respectively. . The A site 61A and the B site 61B are connected by the MDF 604 and the MDF 609 via the public line 62A, the public line 62B, and the public network 63.

  Connections between the bases 61A and 61B and the public network 63 are drawn from the public lines 62A and 62B to the MDFs 604 and 609 of the bases 61A and 61B, and are used to interconnect the public network 63 and the IP network 65 from the MDFs 604 and 609. By connecting to IP telephones 601 and 606 on the network via PSTN gateways 602 and 607, a voice network is configured.

  The IP telephones 601 and 606 transmit call control information to the SIP server 611 in the data center, and the SIP server 611 exchanges the call control information with the destination telephone. After the call connection is established, a voice call is performed between the telephones. At this time, there are two methods for recording voice during a call with the public network 63.

The first method is a method of recording a voice branched from the MDF 604 by connecting the call recording device 603 to the MDF 604 like the A site 61A. The second method is a method of acquiring voice packets flowing on the network and recording them in the call recording device 608 as in the B site 61B. In the case of the second method, not only voice calls with the public network 63 but also extension calls can be recorded. The recorded voice data can be listened to by the specific person accessing the call recording devices 603 and 608 at each site as necessary.
Japanese Unexamined Patent Publication No. 2000-049980 (see paragraph numbers 0009 to 0011, FIG. 1 and FIG. 2)

  However, the first conventional call recording system does not require a special device for recording a call, but since data is stored only in individual telephones, the contents are heard only by the recorded telephones. I couldn't. In the second call recording system shown in FIG. 14, the recorded data can be collectively managed and listened to by a specific person. However, special devices such as call recording devices 603 and 608 are necessary. there were.

  The present invention is to solve the above-described conventional problems, and by recording voice information without requiring a special device and transferring the recorded voice information to another telephone, a specific person can The purpose is to be able to hear the audio information.

  In order to achieve the above object, a telephone with a call recording function according to the present invention includes a first storage means for storing identification information of a communication device connected via a communication line, and an arbitrary telephone connected in response to an incoming call. Recording control means for recording the voice information in the second storage means, and transferring the voice information recorded in the second storage means to the communication device corresponding to the specific identification information in the first storage means And a transfer control means.

  Therefore, according to the telephone with a call recording function of the present invention, the identification information of the communication device to be connected via the communication line is stored in the first storage means, and from any telephone connected in response to the incoming call. Since the voice information is recorded in the second storage means, and the voice information recorded in the second storage means is transferred to the communication device corresponding to the specific identification information of the first storage means, By recording voice information without requiring a special device and transferring the voice information to, for example, a communication device of a specific person, the specific person can hear the voice information.

  Further, in the telephone with a call recording function of the present invention, the transfer control means transmits the specific identification information of the first storage means to a server system connected via a communication line, and acquires the information acquired from the server system. The recorded voice information is transferred to a communication device having an address corresponding to specific identification information.

  Therefore, according to the telephone with a call recording function of the present invention, the specific identification information of the first storage means is transmitted to the server system connected via the communication line, and the specific identification information acquired from the server system. Since the recorded voice information is transferred to the communication device having the address corresponding to the address, for example, the recorded voice information can be transferred to a desired destination without directly holding the destination address. .

  Further, in the telephone with a call recording function of the present invention, the transfer control means, when a caller of the proxy response is set, the voice recorded on the caller of the callee related to the proxy response It is characterized by transferring information.

  Therefore, according to the telephone with a call recording function of the present invention, when a telephone of a callee of a proxy response is set, the recorded voice information is transferred to the telephone of the callee related to the proxy response. As described above, even when the proxy response is set, the recorded voice information can be transferred to the other party of the proxy response.

  In the telephone with a call recording function of the present invention, the recording control means stores predetermined additional information related to the voice information when the voice information is stored in the second storage means. To do.

  Therefore, according to the telephone with a call recording function of the present invention, when the voice information is stored in the second storage means, the predetermined additional information related to the voice information is stored. Advanced search and management can be realized by using information related to information.

  Further, in the telephone with a call recording function of the present invention, the transfer control means transfers the recorded voice information to a management device connectable from any telephone and is stored in a lump by the management device. It is characterized in that it receives a transfer of specific audio information from the existing audio information.

  Therefore, according to the telephone with a call recording function of the present invention, the recorded voice information is transferred to a management apparatus connectable from an arbitrary telephone, and the voice information stored in a batch by the management apparatus is stored. Since specific voice information is transferred from the inside, desired voice information can be acquired from a plurality of voice information stored on the management side.

  The telephone with a call recording function of the present invention has a recording means for recording a call, a storage means for storing voice data recorded by the recording means, and the voice data stored in the storage means is transferred to a predetermined destination. And a transfer means.

  Therefore, according to the telephone with a call recording function of the present invention, the voice information is recorded without requiring a special device, and the voice information is transferred to a specific person's communication device, for example. A person can hear audio information.

  According to the information processing method for the telephone with a call recording function of the present invention, the first step of storing the identification information of the communication device to be connected through the communication line in the first storage means is connected in accordance with the incoming call. A second step of recording voice information from an arbitrary telephone in the second storage means, and a communication corresponding to the specific identification information in the first storage means of the voice information recorded in the second storage means And a third step of transferring to the device.

  Therefore, according to the information processing method for a telephone with a call recording function of the present invention, the identification information of the communication device connected via the communication line is stored in the first storage means, and the arbitrary information connected in response to the incoming call is stored. The voice information from the telephone is recorded in the second storage means, and the voice information recorded in the second storage means is transferred to the communication device corresponding to the specific identification information in the first storage means. Therefore, by recording voice information without requiring a special device and transferring the voice information to, for example, a communication device of a specific person, the specific person can hear the voice information. .

  In the information processing method for a telephone with a call recording function according to the present invention, the third step transmits specific identification information of the first storage means to a server system connected via a communication line, and the server The recorded voice information is transferred to a communication device having an address corresponding to the specific identification information acquired from the system.

  Therefore, according to the information processing method for a telephone with a call recording function of the present invention, the specific identification information of the first storage means is transmitted to a server system connected via a communication line, and the information acquired from the server system is transmitted. Since the recorded voice information is transferred to the communication device of the address corresponding to the specific identification information, for example, the recorded voice information is transferred to the desired destination without directly holding the destination address. can do.

  Further, in the information processing method for a telephone with a call recording function according to the present invention, the third step is performed when the telephone of the caller of the proxy response is set, The recorded voice information is transferred.

  Therefore, according to the information processing method for a telephone with a call recording function of the present invention, when a caller of a proxy response is set, a voice recorded on the caller of the caller according to the proxy response is recorded. Since the information is transferred, the recorded voice information can be transferred to the destination of the proxy response even when the proxy response is set.

  In the information processing method for a telephone with a call recording function according to the present invention, the second step stores predetermined additional information related to the voice information when the voice information is stored in the second storage means. It is characterized by doing.

  Therefore, according to the information processing method for a telephone with a call recording function of the present invention, when storing the voice information in the second storage means, the predetermined additional information related to the voice information is stored. Advanced search and management can be realized by using information related to recorded voice information.

  In the information processing method for a telephone with a call recording function according to the present invention, the third step transfers the recorded voice information to a management apparatus connectable from an arbitrary telephone, and the management apparatus performs batch processing. Thus, it receives a transfer of specific voice information from the stored voice information.

  Therefore, according to the information processing method for a telephone with a call recording function of the present invention, the recorded voice information is transferred to a management apparatus connectable from an arbitrary telephone and is stored in a lump by the management apparatus. Since the transfer of specific voice information is received from the existing voice information, desired voice information can be acquired from the plurality of voice information stored on the management side.

  According to the telephone with a call recording function of the present invention, the identification information of the communication apparatus connected via the communication line is stored in the first storage means, and the voice information from any telephone connected in response to the incoming call Is recorded in the second storage means, and the voice information recorded in the second storage means is transferred to the communication device corresponding to the specific identification information in the first storage means. By recording voice information without requiring a device and transferring the voice information to, for example, a communication device of a specific person, an effect that the specific person can hear the voice information can be obtained.

  In addition, according to the information processing method for a telephone with a call recording function of the present invention, the identification information of the communication device connected via the communication line is stored in the first storage means, and the arbitrary information connected in response to the incoming call is stored. The voice information from the telephone is recorded in the second storage means, and the voice information recorded in the second storage means is transferred to the communication device corresponding to the specific identification information in the first storage means. So, by recording audio information without the need for a special device and transferring this audio information to, for example, a communication device of a specific person, the specific person can hear the audio information. An effect is obtained.

  Hereinafter, embodiments of the present invention will be described with reference to the drawings. FIG. 1 is a block diagram of a call recording system to which a telephone with a call recording function of the present invention is applied. In FIG. 1, the A site 11A is connected to the public network 13 via the public line 12A, and the B site 11B is connected to the public network 13 via the public line 12B. The A site 11A is connected to the IP network 15 via an access line 14A, which is a dedicated line, and the B site 11B is connected to the IP network 15 via an access line 14A, which is a dedicated line. A data center 17 is connected to the IP network 15 by an access line 16 which is a dedicated line.

  The A site 11A includes a plurality of IP telephones with automatic call recording function (hereinafter simply referred to as IP telephones) 101 and 102, a PSTN gateway 103, an MDF (main wiring board) 104, and a router 105. Inside the A base 11A, the IP telephones 101 and 102, the PSTN gateway 103, and the router 105 are connected by a network.

  The B site 11B includes a plurality of IP telephones 106 and 107, a PSTN gateway 108, an MDF 109, and a router 110. Inside the B base 11B, the IP telephones 106 and 107, the PSTN gateway 108, and the router 110 are connected by a network.

  The data center 17 includes a SIP server 111, a call recording server 112, and a router 113. Within the data center 17, the SIP server 111, the call recording server 112, and the router 113 are connected by a network.

  Therefore, the A site 11A, the B site 11B, and the data center 17 are connected by the router 105, the router 110, and the router 113 via the access line 14A, the access line 14B, the access line 16, and the IP network 15, respectively. . Further, the A base 11A and the B base 11B are connected by the MDF 104 and the MDF 109 via the public line 12A, the public line 12B, and the public network 13.

  Connections between the bases 11A and 11B and the public network 13 are drawn from the public lines 12A and 12B to the MDFs 104 and 109 of the bases 11A and 11B, and are used to interconnect the public network 13 and the IP network 15 from the MDFs 104 and 109. By connecting to IP telephones 101, 102, 106 and 107 on the network via PSTN gateways 103 and 108, a voice network is configured.

  The IP telephones 101, 102, 106, and 107 transmit call control information to the SIP server 111 of the data center 17, and the SIP server 111 exchanges call control information with the other party's telephone. After the call connection is established, a voice call is performed between the telephones. At this time, the type of the other party is confirmed from the call control information with the SIP server 111. For example, if the incoming call is an outside line, recording is performed, and the extension is not recorded. , 102, 106, 107 are recorded.

  As an example, it is assumed that data received by IP phone 101 and recorded as a call is transmitted to IP phone 102. In order to transmit the data recorded by the telephone call to the IP telephone set 102, the destination IP address is acquired from the SIP server 111 by inputting the extension number of the IP telephone set 102 of the data transmission destination from the incoming IP telephone set 101. By specifying FTP of the file transfer protocol in the body part on the message, the file is transmitted from the IP telephone set 101 to the IP telephone set 102 by the FTP procedure. The detailed operation sequence for acquiring the IP address of the transfer destination will be described in the description of FIG.

  When an incoming call to IP phone 101 is proxy-answered by IP phone 102 and the call is recorded, and the recorded data is transmitted from IP phone 102 to caller IP phone 101, the incoming call is received from the call control information with SIP server 111. The IP address of the original IP phone 101 is received, and a file is transmitted to the IP phone 101 by the FTP procedure.

  In addition, by previously setting the IP address of a device that can store data other than the IP phone in the IP phones 101, 102, 106, and 107, the recorded data is automatically transferred to the device having the IP address that has been set. Send. For example, after the call ends, a voice message is forcibly transmitted to the call recording server 112 and is not stored in the IP telephones 101, 102, 106, and 107, but the data recorded by the call recording server 112 is collectively managed. Security can be ensured by allowing only a specific person to access the call recording server 112 and listen to the recorded data as necessary. Furthermore, if data is sent to a unified message server or the like, a voice mail service function can be used.

  FIG. 2 is a functional block diagram of voice data transmission / reception of the IP telephone set 221 (101, 102, 106, 107). In FIG. 2, the connector 201 is connected to the LAN control unit 202. The LAN control unit 202 is connected to the DSP 203 and the audio data control unit 210. The DSP 203 is connected to the CODEC 204. The CODEC 204 is connected to the transmission / reception signal separation unit 205. The transmission / reception signal separation unit 205 is connected to the receiver 206. The voice storage unit 207 during a call is connected to the voice message management unit 209 and the voice data control unit 210. The voice data storage unit 208 is connected to the voice data control unit 210 and the voice message management unit 209. The voice message management unit 209 is connected to the clock unit 220. The audio data control unit 210 is connected to the CODEC 204 together with the output of the LAN control unit 202. The clock unit 220 is connected to the display data management unit 216 and the variable screen generation unit 217. The display data management unit 216 is connected to the basic screen data storage unit 218 and the setting data storage unit 219. The variable screen generation unit 217 is connected to the basic screen data storage unit 218 and the setting data storage unit 219. The screen data control unit 215 is connected to the basic screen data storage unit 218 and the setting data storage unit 219.

  The central control unit 214 is connected to the DSP 203, the CODEC 204, the voice message management unit 209, the voice data control unit 210, the clock unit 220, the screen data control unit 215, the display data management unit 216, and the variable screen generation unit 217. The key input unit 211, the display unit 212, and the hook detection unit 213 are connected to each other.

  In FIG. 2, a connector 201 is mainly an 8-pin modular RJ-45 connector, and is an interface with devices on the network. Data received from the network is subjected to transmission / reception control of a MAC (Media Access Control) frame in the LAN control unit 202, and is transferred to a DSP (Digital Signal Processor) 203 when reproducing the audio data, At the time of voice recording during a call, the voice data control unit 210 is passed. The DSP 203 performs audio signal compression / decompression (CS-ACELP, etc.), PB signal analysis, fluctuation control, level control, and echo canceller control. The CODEC 204 converts a digital signal and an analog signal, and the transmission / reception signal separation unit 205 separates the transmission signal and the reception signal transmitted on the communication line and transmits them to the receiver 206 to reproduce the voice data. Is done.

  The voice data is stored in the voice data control unit 210, the voice message management unit 209, the voice storage unit 207 during a call such as a flash memory, and the voice data storage unit 208. The voice data control unit 210 includes a buffer that prevents information from overflowing when reading data into the voice storage unit 207 and the voice data storage unit 208 during a call, and conversely, reading stored voice messages may not be in time. Control the output of input and output messages. The voice message management unit 209 displays the date and time of the voice message, the caller ID, the callee ID, the recording time information (recording start time, recording end time, call time, etc.), the file capacity of the recorded data, whether the message has been listened to or not listened to Message related information such as information is stored in the voice data storage unit 208 and managed together with the voice message. Further, message related information selected by the display data management unit 216 described later can be called from the voice data storage unit 208. In addition, the voice storage unit 207 during a call is a storage unit that temporarily stores until the end of the call when recording the voice during the call in real time. In addition to the voice data storage unit 208, by providing the voice storage unit 207 during a call as a buffer, the capacity of the storage unit becomes insufficient during recording, overwriting existing voice data, or recording stops. Make sure there is nothing. As described above, it is assumed that audio data is not automatically erased but is selected by the user. The voice data storage unit 208 stores voice messages and message related information as data files. The voice message recorded by the call is stored together with the message related information from the voice storage unit 207 during the call after the call ends. Data files exchanged with other devices are transmitted / received from the audio data storage unit 208.

  The central control unit 214 is a control unit that controls the IP telephone set 221, and the key input unit 211 receives user key input and transmits it to the central control unit 214. The display unit 212 performs various displays of the IP telephone set 221, and the hook detection unit 213 detects the on-hook and off-hook of the handset 206 by the user.

  The screen data includes a screen data control unit 215, a display data management unit 216, a variable screen generation unit 217, a basic screen data storage unit 218, and a setting data storage unit 219. The screen data control unit 215 controls output of display data displayed on the display unit 212. The display data management unit 216 manages display data displayed on the display unit 212. The variable screen generation unit 217 displays a variable screen (a screen whose contents change sequentially according to a change in stored contents such as a voice message) according to a display format using a screen that changes according to contents set by the user and temporary data including message-related information as elements. It is displayed on the display unit 212. The basic screen data storage unit 218 stores fixed screen display data in which screen display contents such as a list display screen data such as a main menu and a setting screen incorporated by default are not changed. The setting data storage unit 219 stores the contents set by the user, such as the setting of the incoming call type to be recorded, the setting of whether or not to record when receiving an incoming call, the setting of whether or not to record when calling, and the setting of the destination IP address.

  The clock unit 220 generates a clock to be supplied to the central control unit 214, the voice message management unit 209, and the display data management unit 216, and simultaneously manages time management of voice messages and the like performed by the voice message management unit 209 and the display data management unit 216. It is a clock control unit for enabling it, and is used as the date and time of the recorded voice message and the time display of the IP telephone.

  When there is an access from the user, the display data management unit 216 calls from the basic screen data storage unit 218 according to the data input from the key input unit 211, and the screen data control unit 215 sends the basic screen data to the display unit 212. To do. Further, the variable screen display data selected by the user is read by the variable screen generation unit 217 from the setting data storage unit 219 under the management of the display data management unit 216, or the audio data storage unit 208. By reading the message related information from the screen, a variable screen is created, and the screen data control unit 215 sends the variable screen data to the display unit 212. Note that the screen data control unit 215 sends the basic screen data and the variable screen created by the variable screen generation unit 217 to the display unit 212, and sends message related information and setting data to the voice data storage unit 208 and the setting data storage unit. A buffer is provided to prevent reading and writing from being in time when writing to 219, and display data input / output control is performed.

  FIG. 13 is a display screen of the IP telephone 221. (5-1) is a display screen during standby. Under the management of the display data management unit 216 in FIG. 2, the variable screen generation unit 217 reads the standby screen data from the basic screen data storage unit 218 and the setting data storage unit 219, and these pieces of information are read by the variable screen generation unit 217. The display screen is created as a standby display screen, and the screen data control unit 215 displays and outputs the standby display screen data to the display unit 212. The time of the IP telephone 221 is such that the variable screen generation unit 217 calls the time display data from the clock unit 220, creates a time display screen at any time in the variable screen generation unit 217, and displays it on the display unit 212. is there. As shown in (5-1) of FIG. 13, when a message arrives, for example, “There are three messages” is displayed on the display screen, and the lamp display of the IP telephone 221 informs the user.

  (5-2) in FIG. 13 is a display screen of the main menu. The display data management unit 216 shown in FIG. 2 calls the main new display screen from the basic screen data storage unit 218 and outputs it to the display unit 212. A number is assigned to the head of each item, and the user selects the number of the selected item by inputting a dial button of the IP telephone. Item No. 3 “Record message” is an item for recording one's voice as a message message and transmitting it to one or more people. In addition to selecting from the display screen, recording can be performed with a single button by providing a recording button on the IP telephone 211.

  In addition, (5-3) in FIG. 13 is a message list display screen stored in the IP telephone 221. Displayed when item number 1 to 4 is selected in the main menu of (5-2). Under the management of the display data management unit 216 in FIG. 2, the variable screen generation unit 217 reads the basic screen of the message list display screen from the basic screen data storage unit 218 and reads the requirements set by the user from the setting data storage unit 219. . Further, the voice message related information is read out from the voice data storage unit 208, the information is created as a message list display screen in the variable screen generation unit 217, and displayed on the display unit 212.

  From the left column of each item number, “*”, “#”, and “blank” represent listened messages that “*” has not been listened to, “#” has been sent, and “blank” has not been sent. The next column represents the subject line, such as urgent or faulty. When sending a voice message, in addition to selecting the subject set by default, the subject entered by the user individually is also displayed here. The next column displays caller information. If the caller number and the caller name are registered, the caller name is displayed. If not registered, the caller's external number or extension number is displayed. The next column represents the recording time.

  The row above each column is a key to be input when sorting. For example, if “*” is entered, unlistened messages are preferentially displayed and sorted by unlistened, listened, and untransmitted listened messages. “7 No.” means that an item number is input after 7. For example, if “71” is input, an urgent message is preferentially displayed and sorted by failure, circulation, and others. The same applies to “8 No.” and “9 No.”.

  (5-4) in FIG. 13 is a setting list display screen set in the IP telephone. Displayed when item number 5-6 is selected in the main menu (5-2). Under the management of the display data management unit 216 in FIG. 2, the variable screen generation unit 217 reads the basic screen of the setting list display screen from the basic screen data storage unit 218 and reads the contents set by the user from the setting data storage unit 219. These pieces of information are created as a setting list display screen in the variable screen generation unit 217 and displayed on the display unit 212. Item No. 4 “Destination” is an item for sending to a preset destination such as a call recording device or a unified message server without inputting the destination after recording the call. Item No. 5 “broadcast destination” is an item for transmitting a plurality of recorded voice messages by specifying a destination or a group. The item number 6 “time designation transmission” is an item for designating a time to send a recorded voice message to the other party. Item number 7 “non-response proxy recording” is an item for recording a message when a call is not answered for a certain period of time. Item number 8 “recording on behalf of absentee” is an item for recording a message when the extension is set to be absent. Item No. 9 “Password” is an item for setting a password when the main menu is displayed.

  The communication line described in the claims is the IP network 15, the communication device is the IP telephone 221 (101, 102, 106, 107) and the call recording server 112, the first storage means is the setting data storage unit 219, the second The storage means is the voice data storage unit 208, the identification information is the IP address, the recording control unit is the voice data control unit 210, the transfer control unit is the central control unit 214, the telephone is the IP telephone 221 (101, 102, 106, 107), the server The system corresponds to the SIP server 111, and the management device corresponds to the data center 17.

  FIG. 3 and FIG. 4 are normal sequence diagrams at the time of an incoming call, showing from the public network 13 of FIG. 1 to the IP telephones 101, 102, 106, and 107 with a call recording function for making a call and disconnecting. First, a sequence when a person on the public network 13 side receives an incoming call to the IP telephone 101 at the A base will be described.

  When a call is made from the public network 13 to the telephone number of the IP telephone 101 at the A site 11A, the PSTN gateway 103 receives an IAM message (address start message) and receives the telephone number (3-1). When the telephone number is normally received, an INVITE message is transmitted to the SIP server 111 (3-2), and the SIP server 111 transmits an INVITE message to the IP telephone 101 and makes a call request (3-3). At the same time, the SIP server 111 transmits a 100 Trying message to the PSTN gateway 103 to notify that processing is in progress (3-4). The incoming IP phone 101 transmits a 180 Ringing message to the SIP server (3-5), and the SIP server 111 transmits a 180 Ringing message to the PSTN gateway 103 to notify that it is calling (3-6). . After transmitting the ACM message (address completion message) from the PSTN gateway 103 to the public network 13 (3-7), a ringback tone (RBT) is transmitted (3-8).

  Next, a sequence from when the IP telephone 101 goes off-hook to disconnect the call will be described. When the IP telephone 101 goes off-hook, it transmits a 200 OK message indicating that the session establishment request processing with the public network 13 is completed to the SIP server 111 (3-9), and the SIP server 111 that has received the 200 OK message also receives the 200 OK message. Is transmitted to the PSTN gateway 103 (3-10). The PSTN gateway 103 transmits an ACK message indicating that the session establishment procedure with the public network 13 has been completed to the SIP server 111 (3-11), and the SIP server 111 that has received the ACK message sends an ACK message to the IP telephone set 101. Is transmitted (3-12). The PSTN gateway 103 transmits an ANM as a response message to the public network 13 (3-13), thereby establishing a session between the public network 13 and the IP telephone 101 and starting a voice call (3-14, 3-15). When the call is disconnected from the public network 13 after the call is finished, a REL message (disconnect message) is transmitted to the PSTN gateway 103 (3-16). As a response, the PSTN gateway 103 sends an RLC message (recovery) to the public network 13. (Completion message) is transmitted (3-17). Then, the PSTN gateway 103 transmits a BYE message to the SIP server 111 in order to end the session (3-18), and the SIP server 111 that has received the BYE message also transmits a BYE message to the IP telephone 101 (3- 19). Upon completion of the session termination processing, the IP telephone 101 transmits a 200 OK message to the SIP server 111 (3-20), and the SIP server 111 that has received the 200 OK message transmits a 200 OK message to the PSTN gateway 103 (3). -21). The process from the start to the end of the call is completed when a person on the public network 13 side is received by the IP telephone set 101 at the A base 11A.

  FIG. 4 is a normal sequence diagram at the time of outgoing call that shows a process from the IP telephones 101, 102, 106, and 107 with a call recording function shown in FIG. Here, a sequence when a call is made from the IP telephone set 101 at the A site 11A to a person on the public network 13 side will be described.

  When a call is made from the IP telephone set 101 at the A site 11A to the telephone number of the public network 13, an INVITE message is transmitted to the SIP server 111, and the SIP server 111 receives the telephone number (3-22). When the telephone number is normally received, the SIP server 111 transmits a 100 Trying message to the IP telephone 101 to notify that processing is in progress (3-23). At the same time, an INVITE message is transmitted to the PSTN gateway 103 to make a call request (3-24). The PSTN gateway 103 transmits a 100 Trying message to the SIP server 111, notifies that the processing is in progress (3-25), and then transmits an IAM message (address start message) to the public network 13 (3-26). When the address and the person on the public network 13 side are resolved, the PSTN gateway 103 receives an ACM message (address completion message) from the public network 13 (3-27), and 183 Session Progress message (session progress status). Is transmitted to the SIP server 111 (3-28). The SIP server 111 transmits a 183 Session Progress message to the IP telephone 101 (3-29), and then receives a ringback tone (RBT) from the public network 13 side via the PSTN gateway 103 (3-30). A ringback tone (RBT) from the network 13 side is transmitted to the SIP telephone set 101 (3-31).

  Next, a sequence from when the person on the public network 13 side goes off-hook to disconnect from the call will be described. When a person on the public network 13 side goes off-hook, the PSTN gateway 103 receives an ANM as a response message from the public network 13 (3-32). The PSTN gateway 103 transmits a 200 OK message indicating that the session establishment request processing with the public network 13 has ended to the SIP server 111 (3-33), and the SIP server 111 that has received the 200 OK message transmits the 200 OK message to the IP It transmits to the telephone 101 (3-34). The IP telephone set 101 transmits an ACK message indicating that the session establishment procedure with the public network 13 is completed to the SIP server 111 (3-35), and the SIP server 111 that has received the ACK message sends an ACK message to the PSTN gateway 103. Is transmitted (3-36) to establish a session between the public network 13 and the IP telephone 101 and start a voice call (3-37, 3-38). When the call is disconnected from the IP telephone set 101 after the call ends, the IP telephone set 101 transmits a BYE message to the SIP server 111 to end the session (3-39), and the SIP server 111 that receives the BYE message receives the PSTN. Similarly, a BYE message is transmitted to the gateway 103 (3-40). When the session termination process with the IP telephone 101 is completed, the PSTN gateway 103 transmits a 200 OK message to the SIP server 111 (3-41), and the SIP server 111 that has received the 200 OK message also sends the 200 OK message to the IP telephone 101. (3-42). At the same time, the PSTN gateway 103 transmits a REL message (disconnect message) to the public network 13 (3-43), and receives an RLC message (recovery completion message) from the public network 13 as a response thereto. The session is terminated (3-44). This completes the processing from the IP telephone set 101 at the A site 11A to the person on the public network 13 side, from the start to the end of the call.

  FIG. 5 and FIG. 6 show the process until the incoming call from the public network 13 of FIG. 1 to the IP telephones 101, 102, 106, and 107 with a call recording function is proxy-answered by another IP telephone and the call is disconnected. It is a sequence diagram at the time of a proxy response. Here, a sequence from the person on the public network 13 side to the IP telephone set 102 at the A base 11 </ b> A proxying the incoming call to the IP telephone set 101 and disconnecting the call will be described.

  In FIG. 5, when a call is made from the public network 13 to the telephone number of the IP telephone set 101 at the A site 11A, the PSTN gateway 103 receives the IAM message (address start message) and receives the telephone number (3-45). When the telephone number is normally received, an INVITE message is transmitted to the SIP server 111 (3-46), and the SIP server 111 transmits an INVITE message to the IP telephone 101 and makes a call request (3-47). At the same time, the SIP server 111 transmits a 100 Trying message to the PSTN gateway 103 to notify that processing is in progress (3-48). The incoming IP phone 101 transmits a 180 Ringing message to the SIP server (3-49), and the SIP server 111 transmits a 180 Ringing message to the PSTN gateway 103 to notify that it is calling (3-50). . After transmitting the ACM message (address completion message) from the PSTN gateway 103 to the public network 13 (3-51), a ringback tone (RBT) is transmitted (3-52).

  In a normal case, the process from sending the INVITE message to the SIP server 111 (3-46) and sending the ACM message from the PSTN gateway 103 to the public network 13 (3-51) is unnecessary. is there.

  Next, a sequence until IP telephone 102 completes acceptance of a proxy response will be described. Upon receiving the proxy response, the IP telephone 102 transmits a SUBSCRIBE message to the SIP server 111 (3-53). The SIP server that has received the SUBSCRIBE message transmits the SUBSCRIBE message to the IP telephone 101 and subscribes to the dialog event (3-54). The IP phone 101 transmits a 200 OK message indicating that the processing of the SUBSCRIBE message has been successful to the SIP server (3-55), and the SIP server 111 that has received the 200 OK message also transmits the 200 OK message to the IP phone 102 ( 3-56). The IP telephone 101 transmits a NOTIFY message to the SIP server 111 to notify that the transfer is in progress (3-57), and the SIP server 111 that has received the NOTIFY message transmits a NOTIFY message to the IP telephone 102 (3). -58). The IP phone 102 transmits a 200 OK message indicating that the transfer processing acceptance has been completed to the SIP server (3-59), and the SIP server 111 that has received the 200 OK message similarly transmits a 200 OK message to the IP phone 101 (3). -60).

  Next, a sequence until the handover from the IP telephone 101 to the IP telephone 102 is completed will be described. The IP telephone 102 transmits an INVITE Replace: IPTel SIP telephone 101 message indicating that the telephone to the IP telephone 101 is taken over to the SIP server 111 (3-61). INVITE Replace: The SIP server 111 that has received the IPTel SIP telephone 101 message transmits an INVITE message indicating that it is taking over to the IP telephone 102 to the PSTN gateway 103 (3-62). The PSTN gateway 103 that has received the INVITE message transmits a 200 OK message indicating acceptance of the takeover to the SIP server (3-63), and the SIP server 111 that has received the 200 OK message also transmits the 200 OK message to the IP telephone set 102. Then, the takeover is completed (3-64). After the handover is completed, the PSTN gateway 103 transmits a CANCEL message to terminate the session to the SIP server 111 (3-65), and the SIP server 111 that has received the CANCEL message transmits a CANCEL message to the IP telephone set 101. (3-66). Further, the SIP server 111 transmits a 200 OK message indicating that the session has been completed to the PSTN gateway 103 (3-67), and the IP telephone 101 that has received the CANCEL message similarly transmits a 200 OK message to the SIP server 111 ( 3-68).

  Further, in FIG. 6, the IP telephone set 101 transmits a 487 Request Terminated message indicating that the calling state is terminated by the CANCEL message to the SIP server 111 (3-69), and the SIP server 111 that has received the 487 Request Terminated message receives the 487 Request Terminated message. The 487 Request Terminated message is transmitted to the PSTN gateway 103 (3-70). Further, the SIP server 111 transmits an ACK message indicating completion of the call state termination procedure to the IP telephone 101 and the PSTN gateway 103 (3-71, 3-72).

  Next, a sequence from the end of transfer from the IP telephone set 101 to the IP telephone set 102 until the start / end of the voice call between the public network 13 and the IP telephone set 102 will be described. The IP phone 101 transmits a NOTIFY message indicating that the transfer has been successful to the SIP server 111 (3-73), and the SIP server 111 that has received the NOTIFY message transmits a NOTIFY message to the IP phone 102 (3- 74). The IP phone 102 transmits a 481 Dialogues Not Exist message to the SIP server 111 to establish a session with the public network 13 (3-75), and the SIP server 111 that has received the 481 Dialogues Not Exist message sends the 481Dialog to the IP phone 101. A Does Not Exist message is transmitted (3-76). Further, the IP telephone 102 transmits a 200 OK message indicating that the session establishment request processing with the public network 13 has been completed to the SIP server 111 (3-77), and the SIP server 111 that has received the 200 OK message also receives the 200 OK message. Is transmitted to the PSTN gateway 103 (3-78). The PSTN gateway 103 transmits an ACK message indicating that the session establishment procedure with the public network 13 has been completed to the SIP server 111 (3-79), and the SIP server 111 that has received the ACK message sends the ACK message to the IP telephone 102. Is transmitted (3-80). Then, the PSTN gateway (103) transmits an ANM as a response message to the public network 13 (3-81), thereby establishing a session between the public network 13 and the IP telephone 102 and starting the voice call (3). -82, 3-83). When the call is disconnected from the public network 13 after the call ends, a REL message (disconnect message) is transmitted to the PSTN gateway 103 (3-84), and as a response, the PSTN gateway 103 sends an RLC message (recovery) to the public network 13. (Completion message) is transmitted (3-85). Then, the PSTN gateway 103 transmits a BYE message to the SIP server 111 in order to end the session (3-86), and the SIP server 111 that has received the BYE message also transmits a BYE message to the IP telephone 102 (3- 87). Upon completion of the session termination processing, the IP telephone 102 transmits a 200 OK message to the SIP server 111 (3-88), and the SIP server 111 that has received the 200 OK message also transmits a 200 OK message to the PSTN gateway 103 (3). -89). The process from the start to the end of the call is thus completed.

  FIG. 7 is a sequence diagram until the voice message is transmitted from the IP telephones 101, 102, 106, and 107 with the call recording function of FIG. 1 to another device. Here, the sequence from the IP telephone set 102 recording the voice message at the time of normal response or proxy response to the transmission of the voice message to the IP telephone set 101 will be described. When the IP telephone 102 selects to transmit the voice message stored by the user and inputs the extension number of the destination IP telephone 101, or when the voice telephone is transmitted to the destination IP telephone 101 at the time of proxy response Then, the INVITE message is transmitted to the SIP server 111 to request the IP address of the destination IP telephone 101 (3-90). The SIP server 111 that has received the INVITE message transmits a 200 OK message to the IP phone 102 from the extension number entered by the user and the information of the IP phone 101 that is the call source stored at the time of the proxy response, and the destination IP phone 101 The IP address is notified (3-91). Here, the IP address notified by the SIP server 111 becomes the IP address of the transfer destination. Further, the information of the automatic transfer destination at the end of the call may be address information such as the IP address of the call recording server 112 or the IP telephone set 101 set in advance, and may be the called party number or the calling party number acquired during the call setting sequence. The associated address information may be used.

  From here, the sequence in which the IP telephone set 101 and the IP telephone set 102 directly transmit and receive messages and the voice message is transmitted by FTP will be described. First, the IP telephone 102 transmits a SYN message to the IP telephone 101 as a TCP / IP connection establishment request (3-92). Upon receiving this SYN message, the IP telephone set 101 transmits an ACK message as an establishment response, and a SYN message as a connection establishment request for the IP telephone set 101 from the IP telephone set 102 (3-93). An ACK message is transmitted from 102 to the IP telephone 101 (3-94), and the TCP / IP connection establishment phase ends. When the TCP / IP connection is established, a voice message is transferred from the IP telephone set 102 to the IP telephone set 101 (3-95). When the file transfer is completed, the IP telephone 101 transmits an ACK message indicating that the file reception is completed to the IP telephone 102 (3-96). Upon receiving the ACK message, IP telephone 102 transmits a FIN message to IP telephone 101 as a connection disconnection request (3-97). Upon receiving the FIN message, IP telephone 101 sends an ACK message to IP telephone 102 as a response (3-98), and further transmits a FIN message to IP telephone 102 (3-99). Upon receiving the FIN message, the IP telephone 102 transmits an ACK message to the IP telephone 101 as a response (3-100), and further transmits a BYE message indicating that the file transfer is completed to the SIP server 111 (3- 101). Upon receiving the BYE message, the SIP server 111 transmits a 200 OK message to the IP telephone 102 upon completion of the file transfer end acceptance process, and releases the stored information on the destination IP telephone 101 (3-102).

  As described above, the voice message is transmitted from the IP telephone 102 to the IP telephone 101, and the processing up to the end is completed. Further, when a voice message is transmitted by forcibly or selecting an IP address set to the IP telephone 102, the sequence from (3-92) to (3-100) is performed.

  FIG. 8 is a flowchart from the incoming call to the IP telephone 221 to the call recording. When the central control unit 214 of the IP telephone 221 detects an incoming call to itself (step S101), first, through the setting data storage unit 219, for example, a setting for recording all calls, or a specific external line, extension line, dedicated line, etc. Settings for recording line type, setting for recording presence / absence of incoming call, setting for recording presence / absence of recording for outgoing call, setting for recording specific caller ID, or setting for recording specific caller ID It is confirmed whether it is a setting for recording specific subaddress information, a setting for recording in response to a proxy response, or a setting for recording a specific extension number or a specific external line number to be recorded (step S102). The SIP message received from the SIP server 111 is compared with the setting contents of the data storage unit 219 to determine whether or not it is a recording target (step S103). If not, the process ends (step S104). If the incoming call is to be recorded, the central control unit 214 starts the call recording (step S106) when it detects off-hook through the hook detection unit 213 (step S105). The voice message recorded by the call is stored in the voice data storage unit 208 together with the message related information from the voice storage unit 207 during the call after the call is finished. In addition, the central control unit 214 stores the voice during a call in the voice storage unit 207 during a call (step S107). In addition, by temporarily storing the voice during a call in the voice storage unit 207 during a call according to the procedure described above, the capacity of the storage unit becomes insufficient during recording, and the existing voice data is overwritten or the recording stops. This is to prevent them from being lost. When the central control unit 214 detects on-hook through the hook detection unit 213 (step S108), the central control unit 214 ends the call recording (step S109), and proceeds to A shown in FIG. In the above, an example has been described in which automatic call recording is started based on predetermined conditions set in advance and stored together with message-related information after the call ends (after the handset is on-hook). Recording may be started by the recording instruction operation.

  FIG. 9 is a flowchart for recording a call to the IP telephone 221 at the time of outgoing call. The central control unit 214 of the IP telephone 221 detects off-hook through the hook detection unit 213 (step S110), and transmits the user's destination number from the IP telephone 221 in response to the input (step S111). First, the central control control unit 214, through the setting data storage unit 219, sets, for example, whether or not recording is performed when receiving a call, whether or not recording is performed when calling, and any of an outside line, an extension line, and a dedicated line. Recording is performed, or it is confirmed whether the dialed number is an extension number or an outside line number to be recorded (step S112). The central control unit 214 determines whether or not the call is a recording target call against the SIP message received from the SIP server 111 (step S113). The process ends (step S114). If the call is to be recorded, when the call is established, the call recording is started (step S115). Then, according to the procedure described above, the central control unit 214 stores the call voice in the voice storage unit 207 during a call (step S116). When the central control unit 214 detects on-hook through the hook detection unit 213 (step S117), the central control unit 214 ends the call recording (step S118), and proceeds to A shown in FIG.

  FIG. 10 is a flowchart showing a processing operation until the voice message recorded in the IP telephone 221 is stored. When the central control unit 214 of the IP telephone 221 finishes the automatic call recording, the voice message stored in the voice storage unit 207 during the call is a message such as the date and time of the voice message, the caller number, and information on whether or not it has been heard. It is confirmed whether or not there is a vacancy in the voice data storage unit 208 for saving in the voice data storage unit 208 together with the related information (step S119). If there is no space, the user is notified that there is no space in the audio data storage unit 208 (step S120). Then, the central control unit 214 inquires of the user whether or not the voice data is to be overwritten (step S121). If the user selects not to overwrite via the key input unit 211, the central control unit 214 terminates without saving the voice message stored in the voice storage unit 207 during a call in the voice data storage unit 208 (step S122). Note that the voice message stored in the voice storage unit 207 during a call remains until it is overwritten due to insufficient capacity, and if the voice data storage unit 208 becomes available, whether or not to save the voice message in the voice data storage unit 208 again. Or ask the user.

  When the user selects overwriting from the key input unit 211, the voice message to be deleted is selected from the voice messages stored in the voice data storage unit 208 (step S123). The central control unit 214 determines whether or not there is a free space for storing the voice message stored in the voice storage unit 207 during a call in the voice data storage unit 208 (step S124). After the message related information is added through the unit 209, the voice message is stored in the voice data storage unit 208 through the voice data control unit 210 (step S125). If there is no free space, the central control unit 214 notifies the user side again that there is no free space in the audio data storage unit 208 (step S120), and asks the user side whether to overwrite the audio data. (Step S121). It should be noted that audio data is not automatically erased in this way, but is performed by user selection. In step S125, the central control unit 214 stores the voice message in the voice data storage unit 208 through the voice data control unit 210, and then inquires of the user whether or not to transmit the stored voice message to another device. (Step S126). When the user selects not to transmit via the key input unit 211, the central control unit 214 ends without transmitting the voice message (step S127). If the transmission is selected, the central control unit 214 determines whether or not it is a voice message recorded as a proxy response (step S128). The user side is inquired whether or not to transmit (step S129). When the user selects to transmit to the destination IP phone 221 via the key input unit 211, the central control unit 214 checks the IP address of the destination IP phone 221 with respect to the SIP server 111 (step S130). ), And shifts to C shown in FIG. If it is determined in step S129 that transmission to the destination IP telephone 221 is not performed, or if it is determined in step S128 that the voice message is recorded other than the proxy response, the destination is designated and the voice message is transmitted. (Step S131), the process proceeds to B shown in FIG.

  Next, in FIG. 11, the central control unit 214 of the IP telephone 221 inquires of the user whether or not the transmission destination is designated by an extension number and transmitted to another apparatus (step S132). When the user selects the extension number via the key input unit 211 and selects transmission to another apparatus, the extension number is input (step S133). Next, the central control unit 214 transmits the input extension number to the SIP server 111, and confirms the IP address of the counterpart device (step S134). Then, the central control unit 214 determines whether or not the partner device's IP address has been received from the SIP server 111 (step S135). Is again input (step S133), and the input extension number is transmitted again to the SIP server 111 to check the IP address of the partner apparatus (step S134). Then, the central control unit 214 determines whether or not the IP address of the partner device has been received from the SIP server 111 (step S135).

  If the extension number is specified in step S132 and transmission to another device is not selected, the central control unit 214 selects the transmission destination set in the setting data storage unit 219 (step S136), and stores the setting data. The message of the main call content set in the unit 219 is selected from the items displayed on the display unit 212, or the message is directly input through the key input unit 211 (step S137), and the information is stored in the voice message management unit. After being added to the voice message through 209, the voice message is transmitted to the destination IP address by FTP communication through the voice data control unit 210 (step S138).

  After transmitting the voice message to the destination IP address, the user is inquired whether or not to delete the transmitted voice message from the voice data storage unit 208 (step S139). When the user selects to delete the transmitted voice message through the key input unit 211, the central control unit 214 deletes the voice message (step S140) and ends this processing operation (step S141). ). Even if it is selected not to delete the voice message transmitted in step S139, the process proceeds to step S141 in order to end this processing operation. If the IP address of the partner device is received from the SIP server 111 in step S135, the process proceeds to step S137.

  In addition, when the voice message is forcibly transmitted from the IP telephone 221 to another device, the central control unit 214, when the automatic call recording is finished, records with a proxy response as shown in the flow of FIG. It is determined whether or not the message is a voice message (step S142). If it is determined that the message is a voice message recorded other than the proxy response, the voice message is forcibly sent to the transmission destination set in the setting data storage unit 219. It transmits by communication (step S144). Then, after transmitting the voice message, the central control unit 214 does not save the transmitted voice message in the voice data storage unit 208, but deletes the data in the voice storage unit 207 during the call (step S145), thereby Security is ensured so that a voice message is not heard by a person, and this processing operation is terminated (step S146). If it is determined in step S142 that the voice message is recorded as a proxy response, the SIP server 111 confirms the IP address of the destination IP telephone 221 and transmits the voice message (step S143), and step S145. Migrate to In the above description, the case where the transfer destination information is set in advance from an IP telephone or the like has been described as an example. However, manual transfer means for specifying the extension number of the transfer destination and the address of the call recording server after the call recording is completed may be provided. Good. In addition, when manual transfer is performed, if the transfer destination is not input, the call recording data is deleted, the call recording data is transmitted to a specific transfer destination, or the call recording data is It may be stored in its own device.

  As described above, according to the present embodiment, the IP address of another IP telephone connected through the IP network 15 is stored in the setting data storage unit 219, and from any telephone connected in response to an incoming call. The voice message is recorded in the voice data storage unit 208, and the voice message recorded in the voice data storage unit 208 is transferred to the IP telephone corresponding to the specific IP address in the setting data storage unit 219. By recording a voice message without requiring a special device and transferring the voice message to, for example, a specific person's IP phone, the specific person can listen to the voice message.

  In addition, according to the present embodiment, the SIP server 111 connected through the IP network 15 is requested for the IP address of the specific IP telephone, and the IP address of the specific IP address acquired from the SIP server 111 is recorded to the IP telephone. Since the voice message is transferred, the recorded voice message can be transferred to a desired destination without directly holding the destination IP address.

  Also, according to the present embodiment, when the IP phone of the proxy response destination is set, the recorded voice message is transferred to the IP phone of the destination of the proxy response. Even when the proxy response is set, the recorded voice message can be transferred to the destination of the proxy response.

  According to the present embodiment, when a voice message is stored in the voice data storage unit 208, the predetermined additional information related to the voice message is also added and stored, so that the voice message is related to the recorded voice message. Using information, advanced search and management can be realized.

  Further, according to the present embodiment, the recorded voice message is transferred to the data center 17 connectable from an arbitrary IP telephone, and the voice messages stored in a batch by the data center 17 are transferred. Since a specific voice message can be selectively transferred from the inside, a desired voice message can be acquired from a plurality of voice messages stored on the data center 17 side.

  In the above-described embodiment, the voice message is automatically transferred. However, the voice message may be transferred manually by the user. Furthermore, the voice message may be automatically transferred or transferred by a user's manual operation, so that the voice message may be selected by setting the transfer mode.

  In the above embodiment, the IP telephone 221 is taken as an example to describe the telephone with a call recording function and the information processing method of the present invention, but the telephone of the present invention is not limited to the IP telephone. The identification information of the communication device connected via the communication line is stored in the storage means, the voice information from any telephone connected in response to the incoming call is recorded in the storage means, and the voice information recorded in the storage means is recorded. It is clear that the present invention can be applied to any telephone that can be transferred to a communication device corresponding to specific identification information in the storage means, that is, a general telephone, a multifunction telephone, a wireless telephone, and the like. . In other words, the above-described embodiment is to explain the present invention, and other embodiments and modifications that can be easily conceived by those skilled in the art are included in the present invention in accordance with the gist of the claims.

  According to the telephone with a call recording function of the present invention, the identification information of the communication apparatus connected via the communication line is stored in the first storage means, and the voice information from any telephone connected in response to the incoming call Is recorded in the second storage means, and the voice information recorded in the second storage means is transferred to the communication device corresponding to the specific identification information in the first storage means. By recording voice information without the need for a device and transferring the voice information to, for example, a communication device of a specific person, the specific person can hear the voice information. This is useful for IP telephones that accommodate and connect.

It is a figure which shows the communication system to which the telephone with a call recording function in embodiment of this invention is applied. It is a block diagram of an IP telephone with a call recording function in an embodiment of the present invention. It is a sequence diagram which shows the flow of the data of the IP telephone in embodiment of this invention. FIG. 4 is a sequence diagram showing a data flow of the IP telephone following FIG. 3. FIG. 5 is a sequence diagram showing a data flow of the IP telephone following FIG. 4. FIG. 6 is a sequence diagram showing a data flow of the IP telephone following FIG. 5. FIG. 7 is a sequence diagram showing a data flow of the IP telephone following FIG. 6. It is a flowchart which shows the information processing method of the IP telephone in embodiment of this invention. It is a flowchart which shows the information processing method of the IP telephone following FIG. FIG. 10 is a flowchart showing an information processing method of the IP telephone following FIG. 9. It is a flowchart which shows the information processing method of the IP telephone following FIG. FIG. 12 is a flowchart showing an information processing method of the IP telephone following FIG. 11. It is a figure which shows the display screen of the IP telephone in embodiment of this invention. It is a block diagram of the conventional call recording system.

Explanation of symbols

15 IP network (communication line)
17 Data center (management device)
101, 102, 106, 107, 221 IP telephone with call recording function (telephone with call recording function and communication device)
111 SIP server (system server)
113 Call recording server (communication device)
208 Audio data storage (second storage means)
210 Voice data control unit (recording control means)
214 Central control unit (transfer control means)
219 Setting data storage unit (first storage means)

Claims (11)

First storage means for storing identification information of a communication device connected via a communication line;
Recording control means for recording voice information from any telephone connected in response to an incoming call in the second storage means;
Transfer control means for transferring voice information recorded in the second storage means to a communication device corresponding to specific identification information in the first storage means;
A telephone with a call recording function, characterized by comprising:   The transfer control unit transmits the specific identification information of the first storage unit to a server system connected via a communication line, and transmits the specific identification information of the first storage unit to a communication device having an address corresponding to the specific identification information acquired from the server system. 2. The telephone with a call recording function according to claim 1, wherein the recorded voice information is transferred.   The transfer control means transfers the recorded voice information to the caller telephone set for the proxy response when the caller telephone set for the proxy response is set. A telephone with a call recording function described in 1.   2. The telephone with a call recording function according to claim 1, wherein the recording control means stores predetermined additional information related to the voice information when the voice information is stored in the second storage means. .   The transfer control means transfers the recorded voice information to a management device connectable from an arbitrary telephone, and transfers specific voice information from the voice information stored in a batch by the management device. The telephone with a call recording function according to claim 1, wherein the telephone with a call recording function is received. Recording means for recording calls,
Storage means for storing voice data recorded by the recording means;
Transfer means for transferring the audio data stored in the storage means to a predetermined destination;
A telephone with a call recording function, characterized by comprising: A first step of storing in a first storage means identification information of a communication device connected via a communication line;
A second step of recording voice information from any telephone connected in response to an incoming call in a second storage means;
A third step of transferring voice information recorded in the second storage means to a communication device corresponding to specific identification information in the first storage means;
An information processing method for a telephone with a call recording function, characterized in that:   The third step transmits the specific identification information of the first storage means to a server system connected via a communication line, and a communication device having an address corresponding to the specific identification information acquired from the server system 8. The information processing method for a telephone with a call recording function according to claim 7, wherein the recorded voice information is transferred to the telephone.   The third step is characterized in that, when a telephone set as a callee of a proxy response is set, the recorded voice information is transferred to the caller telephone set of the proxy response. 8. An information processing method for a telephone with a call recording function according to 7.   The said 2nd step memorize | stores the predetermined | prescribed additional information relevant to the said audio | voice information when memorize | storing audio | voice information in a said 2nd memory | storage means, The call recording function with Claim 7 characterized by the above-mentioned Information processing method for telephones. The third step transfers the recorded voice information to a management device connectable from an arbitrary telephone, and selects specific voice information from the voice information collectively stored by the management device. 8. The information processing method for a telephone with a call recording function according to claim 7, wherein the transfer is received.

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