JP2013021436A - Voice mail server and voice mail system - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a voice mail service convenient and user-friendly for a user who reproduces and listens to a recorded voice message and tries to talk with the user who has recorded the voice message.SOLUTION: After message reproduction processing of reproducing the recorded voice message by calling from a telephone terminal 70A to a voice mail server 10, a request for a direct call is accepted and a call is originated to a telephone terminal 70B of the recording-source user of the message. This enables seamless transition to the direct call with the telephone terminal 70B of the recording-source user without disconnection of the call. This allows for the voice mail service convenient and user-friendly.

Description

  The present invention relates to a voice mail server device and a voice mail system, and more particularly to processing after reproduction of voice mail recorded using a telephone terminal such as a mobile phone terminal.

  A Voice Mail (hereinafter referred to as “voice mail”) service is known in which voice messages can be transmitted without calling the other party's mobile phone. When using a voice mail service via a mobile telephone network, the following processing is performed. That is, when recording a voice message, a specific number for using the voice mail service is added to the head of the telephone number of the other party (destination) to which the voice mail is to be sent. This specific number is a specific number for each network (carrier network) operated by the communication carrier. For example, in the case of the voice mail service described in Non-Patent Document 1, “* 2020” is added to the head and “* 2020090BBBBBBBB” is transmitted (B is an arbitrary number). In the case of the voice mail service described in Non-Patent Document 2, “1612” is added to the head and “16120090BBBBBBBB” is transmitted (B is an arbitrary number). By making such a call, the call is connected to the voice mail server and a voice message can be recorded. In the case of the answering machine service described in Non-Patent Document 3, a voice message can be recorded from a mobile phone in another carrier network.

  Further, when a voice message is recorded, a notification to that effect is sent to the destination. Select the specific part (for example, the part displayed as "Play") included in the notification at the destination that received this notification or specify it differently from when recording a voice message A voice message can be reproduced by adding a number (for example, “* 2021” in the case of Non-Patent Document 1) to the beginning and making a call.

"Voice Delivery Service" NTT DoCoMo, [Search May 31, 2011], Internet <URL: http://www.nttdocomo.co.jp/service/communication/koe_no_takuhaibin/> “Voice Mail” Au KDDI Corporation, [Search May 31, 2011], Internet <URL: http://www.au.kddi.com/voice_mail/index.html> "How to use answering machine" SoftBank, [May 31, 2011 search], Internet <URL: http://mb.softbank.jp/mb/support/3G/voice_mail/06-02.html>

In order to reproduce the recorded message using the voice mail service described above, a call is made to the voice mail server. Then, if the user who listened to the message plays and wants to talk with the user who recorded the message, he / she temporarily disconnects the call for the reproduction and inputs the other party's phone number from his telephone terminal again. I need to make a call. This is not convenient and has a problem in terms of service convenience.
The present invention has been made for the purpose of solving the problems of the background art described above, and provides a voice mail server device and a voice mail system capable of realizing a voice mail service that is more convenient and easy to use. It is.

A voice mail server device according to an aspect of the present invention is a message reproduction unit (for example, in FIG. 1) that reproduces a recorded message by a first call from a first telephone terminal (for example, telephone terminal 70A in FIG. 2). A message reception function of the media control scenario 6) and a call reception unit (for example, FIG. 1) that receives a call request to the recording source of the message in a state where the first call is continued after the message reproduction processing by the message reproduction unit. When the call acceptance unit accepts a call request, a call is made to the second telephone terminal (for example, telephone terminal 70B in FIG. 2) that is the recording source of the message. A calling unit (for example, a connection request transfer function of the call processing scenario 2 in FIG. 1), and makes a call between the first telephone terminal and the second telephone terminal. Characterized in that it current.
By configuring the voice mail server device in this way, it is possible to seamlessly shift from the message reproduction processing to the direct call with the recording source user, and to realize a highly convenient and easy-to-use voice mail service.

  The voice mail server device according to one aspect of the present invention mixes the voice of the first call between the first telephone terminal and the own apparatus with the voice of the second call from the own apparatus to the second telephone terminal. Thus, a conference unit (for example, a conference function of the media processing unit 4 in FIG. 1) that realizes a call between the first telephone terminal and the second telephone terminal may be further included. By configuring the voice mail server device in this way, the voice of two calls can be mixed by the conference function, and seamless transition can be made from the message playback process to the direct call with the message recording source, which is highly convenient and easy to use. A good voicemail service can be realized.

  Further, in the voice mail server device according to one aspect of the present invention, the calling unit transmits a signal for calling to the second telephone terminal including the address of the first telephone terminal, A call between the first telephone terminal and the second telephone terminal may be realized. If the voice mail server device is configured in this way, a call between the first telephone terminal and the second telephone terminal can be realized, so that it is possible to seamlessly shift from a message reproduction process to a direct call without using the conference function. Highly convenient and easy-to-use voice mail service.

  Furthermore, the voice mail server device according to one aspect of the present invention provides a replay call for continuing message replay processing by the first call from the first telephone terminal when a disconnection signal is transmitted from the second telephone terminal. It is desirable to further include a continuation unit (for example, a reproduction call continuation function of the media control scenario 6 in FIG. 1). By configuring the voice mail server device in this way, it is possible to return to the original reproduction process after the direct call with the message recording source is completed, and to realize a user-friendly voice mail service.

  A voice mail system according to an aspect of the present invention includes any one of the above voice mail server apparatuses and a telephone terminal (for example, telephone terminal 70A in FIG. 2) that transmits the call request to the voice mail server apparatus. It is characterized by that. According to this voice mail system, it is possible to seamlessly shift to a direct call with the recording source user while continuing the call after the message reproduction process, and it is possible to realize a highly convenient and easy-to-use voice mail service.

According to the present invention, after a message reproduction process in which a recorded message is reproduced by a call from a telephone terminal, a direct call request is accepted and sent to the recording source of the message. It is possible to seamlessly shift to a direct call with the recording source user while continuing the call, and to realize a highly convenient and easy-to-use voice mail service.
Further, after the direct call with the recording source user is terminated, the message reproduction process by the call from the telephone terminal is continued, so that the original message reproduction process can be restored, and an easy-to-use voice mail service can be realized.

It is a block diagram which shows the function of the voice mail server apparatus by this Embodiment. It is a figure which shows the outline | summary of a process in the case of making a telephone call between the users of the telephone terminal of a self-carrier network using the voice mail server apparatus by this Embodiment. In the voice mail server apparatus by this Embodiment, it is a figure which shows the outline | summary of a process when the telephone terminal which is the recording origin of the reproduced | regenerated message is a telephone terminal of another carrier network. It is a sequence diagram which shows the processing content which calls to the telephone terminal of the recording origin of the message after the reproduction | regeneration processing of a voice message. It is a sequence diagram which shows the processing content in the case of making a telephone call between the users of telephone terminals after the voice message playback process and then continuing the playback process. It is a sequence diagram which shows the process in the case of utilizing the Conference function of a voice mail server apparatus. It is a sequence diagram which shows a process when not utilizing the Conference function of a voice mail server apparatus. It is a sequence diagram which shows transmission / reception in the voice mail server apparatus in the case of using a Conference function. It is a sequence diagram which shows signal transmission / reception in the voice mail server apparatus when not using a Conference function. It is a sequence diagram which shows a process in case a recording origin is a telephone terminal of another carrier network, and utilizes the Conference function of a voice mail server apparatus. It is a sequence diagram which shows a process in case a recording origin is a telephone terminal of another carrier network, and does not utilize the Conference function of a voice mail server apparatus.

Hereinafter, embodiments of the present invention will be described with reference to the drawings. In the drawings referred to in the following description, the same parts as those in the other drawings are denoted by the same reference numerals.
The following describes a case where the voice mail server device according to the present embodiment is provided in a mobile telephone network.
Also, for convenience of explanation, company D, which is a mobile phone operator, is its own company, and company A and company S, which are other mobile phone operators, are other companies. The own carrier network is a mobile phone network operated by the company, and the other carrier network is a mobile phone network operated by another company.

  Note that although an actual mobile telephone network is provided with devices such as a radio base station device and a base station control device, these devices are not shown and will not be described in order to simplify the description. Further, the illustration and description of the exchange provided in the telephone network are omitted as appropriate in order to simplify the description. In addition, a telephone terminal that appears in the following description is a device that has a telephone number and can perform a voice call. This telephone terminal includes a smart phone and a fixed telephone terminal in addition to a known mobile phone terminal.

(Function of voice mail server device)
The function of the voice mail server apparatus according to this embodiment will be described with reference to FIG. Referring to FIG. 1, a voice mail server device (hereinafter abbreviated as a VM server as appropriate) 10 of this embodiment includes a call processing unit 1, a call processing scenario 2, a media processing unit 4, and a media storage unit 5. And a media control scenario 6.

The call processing unit 1 notifies the media processing unit of a server interface function that realizes signal transmission and reception with an external device (such as an SMS transmission server 30 described later) and a media control scenario that is activated when a guidance is output. It has a media processing unit interface function and a call processing scenario interface function that activates a call processing scenario when a connection request signal is received.
The call processing scenario 2 is a program group activated by the call processing scenario interface function of the call processing unit 1. The call processing scenario 2 includes a program for realizing a connection request transfer function for calling a message recording source, and a program for starting a media control scenario when outputting voice guidance.

The media processing unit 4 records and reproduces a voice message by the user in response to a request from the call processing unit 1 and an exchange interface function that transmits a control signal for outputting a guidance message by voice to the telephone terminal. Call conferencing is realized by mixing the call processing unit interface function for starting the media control scenario 6 to be performed, the first call from the telephone terminal to the own apparatus, and the second call from the other telephone terminal to the own apparatus. And a conference function (Conference function).
The media storage unit 5 has a function of storing a voice message by the user. The voice message is stored in the media storage unit 5 in association with the telephone number of the telephone terminal of the user as the destination and the telephone number of the telephone terminal used for recording the message.

  The media control scenario 6 is a program group activated by the media processing unit 4. The media control scenario 6 includes a program for recording a voice message by a user to the media storage unit 5 and a program for reproducing a voice message stored in the media storage unit 5. The functions realized by the program of the media control scenario 6 include the following functions. That is, a message playback function for playing back a voice message stored in the media storage unit 5, a call reception function for receiving a call with the telephone terminal that recorded the message after playing the message, and a telephone terminal that recorded the message There is a replay call continuation function for continuing the message replay process when the call is terminated.

  The voice mail server device 10 is connected to the exchange 20 via a network. In this example, the C-plane (Control plane) side (control side) including the call processing unit 1 and the U-plane (User plane) side (user side) including the media processing unit 4 are respectively connected to the exchange 20. Yes. In this example, the protocol on the C-plane side is SIP (Session Initiation Protocol), and the protocol on the U-plane side is RTP (Real-time Transport Protocol).

By the way, the voice mail server device 10 can be connected to an SMS (Short Message Service) transmission server 30 which is an external device by the function of the call processing unit 1.
The SMS transmission server 30 is a server that transmits a message by the well-known SMS. By sending a message transmission request to the SMS transmission server 30, an SMS message is transmitted. That is, by sending a message transmission request from the call processing unit 1 to the SMS transmission server 30, an SMS message notifies that the voice mail has been recorded, or an SMS message indicates that the voice mail has been reproduced. Or can be notified.

  Note that the voice mail server device may be configured as a device in which the above-described units are collectively provided in one housing, or may be configured in which the above-described units are distributed and provided in a plurality of housings. In the case of a configuration in which the above units are distributed and provided in a plurality of cases, the units may be connected via a network. When realizing a voice mail service in a mobile telephone network, the C-plane and U-plane are often separated. In that case, the call processing unit and the media processing unit are realized by physically separate devices. Often done. Thus, not only when each unit is realized by a single device, but also when each unit is realized by a plurality of devices and each device is connected via a network, the “voice” in this specification is used. Corresponds to “mail server device”.

(Outline of processing by voice mail server)
Hereinafter, an outline of processing by the voice mail server device of this example will be described. FIG. 2 is a diagram showing an outline of processing when a call is made between telephone terminals of the own carrier network.
In FIG. 2, when an SMS message notifies that a message addressed to telephone terminal 70A has been recorded, and “play” is selected and pressed in the SMS message body, in FIG. When a call for message reproduction is made from the telephone terminal 70A to the VM server 10, a first call is formed between the telephone terminal 70A and the VM server 10 (step S1). The same applies when a call is made by pressing a special number for message reproduction (for example, “* 2021”).

  In the state where the first call is formed, the message stored in the media storage unit 5 of the VM server 10 is reproduced. Then, after the message reproduction process, the voice guidance is output from the VM server 10 to the telephone terminal 70A in a state where the first call is continued. As a result of the interaction with the user of the telephone terminal 70A who heard the voice guidance, that is, as a result of the interaction, the VM server 10 detects that a request for switching to a user-to-user call, that is, a call request for a message recording source is output from the telephone terminal 70A. (Step S2).

  Then, the VM server 10 makes a call to the telephone terminal 70B that is a recording source of the reproduced message (step S3). At this time, the calling signal includes the telephone number of the telephone terminal 70B as the called number information. The called number information is, for example, “090BBBBBBBB”. Further, the call signal includes the IP address of the VM server 10 or the IP address of the telephone terminal 70A.

  Thereafter, when there is an incoming call to the telephone terminal 70B of the same company D network and an operation by the user is performed, the call from the telephone terminal 70A to the VM server 10 is switched (step S4), and the telephone terminal 70A and the telephone terminal A call is made with 70B (step S5). Thereby, a direct call between the user of the telephone terminal 70A and the user of the telephone terminal 70B is realized.

  Processing when the message recording source is a telephone terminal of another carrier network will be described with reference to FIG. Even if the recording source of the message is changed from the own carrier network to another carrier network by Mobile Phone Number Portability (MNP), if a call is made to the telephone number of the telephone terminal 70B, the well-known MNP By the redirection processing (or transfer by the port-out source carrier), the call can be made again to the port-out destination carrier network. Here, the case where the telephone terminal 70B, which is the recording source of the reproduced message, is a telephone terminal of another carrier network (for example, the A company network or the S company network) in the processing in the case of FIG. 2 described above will be described. To do.

  As shown in FIG. 3, after the message reproduction process, the VM server 10 detects that a call request is output from the telephone terminal 70A (step S2), and proceeds to the telephone terminal 70B that is a recording source of the reproduced message. To make a call (step S3). At this time, the calling signal includes the telephone number of the telephone terminal 70B as the called number information. The called number information is, for example, “090BBBBBBBB”.

  Thereafter, when there is an incoming call to the telephone terminal 70B of the same company D network and an operation by the user is performed, the call from the telephone terminal 70A to the VM server 10 is switched (step S4), and the telephone terminal 70A and the telephone terminal A call is made with 70B (step S5). Thereby, a direct call between the user of the telephone terminal 70A and the user of the telephone terminal 70B is realized. That is, even if the telephone terminal 70B that is the recording source of the reproduced message is a telephone terminal of another carrier network, the telephone number of the telephone terminal 70B is included in the call signal from the VM server 10 as the called number information. Thus, as in the case of FIG. 2, a direct call between the user of the telephone terminal 70A and the user of the telephone terminal 70B is realized.

(Processing content of voice mail server device)
Next, the processing contents of the voice mail server apparatus will be described in more detail with reference to FIGS.
FIG. 4 is a sequence diagram showing processing contents to be called to the telephone terminal that is the recording source of the message after the voice message is reproduced. In FIG. 4, in a state where a call is formed between the telephone terminal 70A and the VM server 10 via the exchange 20, a call is in progress and a message reproduction process is performed (step S401). After the message reproduction process, an interaction with the user of the telephone terminal 70A is performed, and the VM server 10 receives a request for transition to a call between users, that is, a connection request signal (step S402). The connection request signal can be realized by, for example, PB sound or http Request based on a user input to the telephone terminal 70A. By receiving this connection request signal, the VM server 10 detects that a direct call is requested (step S403).

The VM server 10 that has detected that a direct call is requested performs outgoing call processing, that is, outgoing call to the telephone number of the telephone terminal 70B that is the recording source of the message (step S404). Thereby, the calling process for the telephone terminal 70B is activated (step S405), and a connection request signal is transmitted.
Then, in the destination network, it is determined whether or not the telephone terminal 70B has shifted to a contract with another carrier (that is, port-out) using the MNP system (step S406). The destination network is a company network (D company network) or another carrier network (A company network or S company network). Whether or not MNP has been performed is determined by referring to an MNP database (not shown) in the destination network.

If the result of determination in step S406 is that MNP has not been performed (step S406: NO), the telephone terminal 70B receives a connection request signal from the VM server 10 (steps S406 → S407). As a result, as described above, a call is formed between the telephone terminal 70A and the telephone terminal 70B, and a telephone call between users becomes possible.
On the other hand, if the result of determination in step S406 is that MNP has been performed (step S406: YES), the MNP destination (that is, the port-out destination) is determined by referring to the MNP database. Is notified (steps S406 → S408). The VM server 10 transmits a reconnection request signal to the notified MNP destination network (step S409). As a result, the telephone terminal 70B receives the connection request signal from the VM server 10 (steps S409 → S407), and as described above, a call is formed between the telephone terminal 70A and the telephone terminal 70B, and a call between the users Is possible.

Shifting to FIG. 5, in a state where a call is formed between the telephone terminal 70A and the telephone terminal 70B and a call is made between the users (steps S501a and S501b), the call is ended (step S502) and recording is performed. A call disconnection request is output from the original telephone terminal 70B (step S503).
The VM server 10 detects the end of the call process by receiving a call disconnection request from the recording telephone terminal 70B (step S504). Thereby, the call between the telephone terminal 70A and the VM server 10 remains connected by the reproduction call continuation function of the VM server 10, and the message reproduction process is continued (step S505). That is, the process returns to the state of step S401 in FIG. Note that when a disconnection request is output from the telephone terminal 70B instead of the recording source telephone terminal 70A, the process does not proceed to the message reproduction process.

(When the recording source is a telephone terminal of the company network)
Processing in the case where the recording source of the reproduced message is a telephone terminal of the company network will be described with reference to FIGS.
FIG. 6 is a sequence diagram illustrating processing when the conference function of the VM server 10 is used. In FIG. 6, when a call request to the recording source of the message is output from the telephone terminal 70A during the reproduction of the message (step S601), the VM server 10 is requested to shift to a call between users ( That is, it is detected that a direct call has been requested (step S602). Then, the VM server 10 outputs voice guidance to the telephone terminal 70A (step S603). This voice guidance is, for example, “I will call you”.

  Thereafter, the VM server 10 transmits a call signal to the telephone terminal 70B (step S604). This outgoing signal can be realized by, for example, a SIP INVITE message. The outgoing signal transmitted in step S604 includes “090BBBBBBBB”, which is the telephone number of the destination telephone terminal 70B, as the destination “To”. In addition, the outgoing signal transmitted in step S604 includes “090AAAAAAAAA”, which is the telephone number of the telephone terminal 70A, as the source “From”. Since the call is actually transmitted from the VM server 10, “From” may include the phone number of the caller VM server 10 instead of the phone number of the phone terminal 70A. Further, the SDP (connection destination IP address) of the D company VM server 10 is added to the outgoing signal transmitted in step S604.

The telephone terminal 70B that has received the outgoing signal via the exchange 20 first transmits a provisional response message (step S605). This provisional response message is received by the D company VM server 10 via the exchange 20. The provisional response message transmitted in step S605 can be realized by a SIP 183 message, for example.
Thereafter, the telephone terminal 70B transmits a calling message indicating that the calling is in progress (step S606). This ringing message can be realized by, for example, a SIP 180 message.

This ringing message is received by the D company VM server 10 via the exchange 20. Then, the VM server 10 mixes the ring back tone (hereinafter referred to as RBT) and the voice of the call between the telephone terminal 70A and the VM server 10 by the conference function (steps S607a and S607b). Thereby, since RBT can be output to the telephone terminal 70A, the user of the telephone terminal 70A can recognize that the (user's) call of the telephone terminal 70B is in progress. The RBT can be output from the exchange 20 or a server device (not shown).
Thereafter, the telephone terminal 70B transmits a response message (step S608). This response message is transmitted to the D company VM server 10. The response message transmitted in step S608 can be realized by, for example, a SIP 200 OK message.

  After the above processing, a call is established between the telephone terminal 70B and the VM server 10 of the D company carrier network (step S609a), and the call voice, the originating telephone terminal 70A and the VM server of the D company carrier network are established. 10 and the voice of the ongoing call (step S609b) are mixed by the Conference function of the D company VM server 10. With this conference function, a voice call can be realized between the telephone terminal 70A and the telephone terminal 70B. For example, it is possible to enjoy a conversation between the user of the telephone terminal 70A and the user of the telephone terminal 70B, such as “Hello, long time no see!”.

  Next, FIG. 7 is a sequence diagram illustrating processing when the conference function of the VM server 10 is not used. In FIG. 7, when a call request to the recording source of the message is output from the telephone terminal 70A during the reproduction of the message (step S701), the VM server 10 is requested to shift to a call between users ( That is, it is detected that a direct call has been requested (step S702). Then, the VM server 10 outputs voice guidance to the telephone terminal 70A (step S703). This voice guidance is, for example, “I will call you”.

  Thereafter, the VM server 10 transmits a call signal to the telephone terminal 70B (step S704). This outgoing signal can be realized by, for example, a SIP INVITE message. The outgoing signal transmitted in step S704 includes “090BBBBBBBB”, which is the telephone number of the destination telephone terminal 70B, as the destination “To”. Further, the transmission signal transmitted in step S704 includes “090AAAAAAAAA”, which is the telephone number of the telephone terminal 70A, as the transmission source “From”. Since the call is actually transmitted from the VM server 10, “From” may include the phone number of the caller VM server 10 instead of the phone number of the phone terminal 70A. Further, the SDP (connection destination IP address) of the D company VM server 10 is added to the outgoing signal transmitted in step S704.

The telephone terminal 70B that has received the outgoing signal via the exchange 20 first transmits a provisional response message (step S705). This provisional response message is received by the D company VM server 10 via the exchange 20. The provisional response message transmitted in step S705 can be realized by a SIP 183 message, for example.
Upon receiving the provisional response message in step S705, the company D VM server 10 transmits a switching signal to the telephone terminal 70A (step S706). The telephone terminal 70A that has received this switching signal transmits a response signal (step S707). The switching signal transmitted in step S706 can be realized by, for example, a SIP Re-INVITE message. The response signal transmitted in step S707 can be realized by a SIP 200 OK message, for example.

Here, the provisional response message transmitted in step S705 is a response message that requires arrival confirmation, and the arrival confirmation message is exchanged between the telephone terminal 70 and the D company VM server 10 (step S708).
Thereafter, the telephone terminal 70B transmits a calling message indicating that the calling is in progress (step S709). This ringing message can be realized by, for example, a SIP 180 message.

This ringing message is received by the D company VM server 10 via the exchange 20. Then, the RBT is output to the telephone terminal 70A (step S710). Thereby, the user of the telephone terminal 70A can recognize that the telephone terminal 70B (user's) is being called. The RBT can be output from the exchange 20 or a server device (not shown).
Thereafter, the telephone terminal 70B transmits a response message (step S711). This response message is transmitted to the D company VM server 10. The response message transmitted in step S711 can be realized by, for example, a SIP 200 OK message.

  After the above processing, a call is established between the telephone terminal 70A and the telephone terminal 70B (step S712). By establishing this call, a voice call can be realized between the telephone terminal 70A and the telephone terminal 70B. For example, it is possible to enjoy a conversation between the user of the telephone terminal 70A and the user of the telephone terminal 70B, such as “Hello, long time no see!”.

(Signal exchange within the VM server)
Next, signal transmission / reception of each unit in the VM server 10 will be described.
FIG. 8 is a sequence diagram illustrating signal transmission / reception in the VM server 10 when using the conference function described with reference to FIG. 6. In FIG. 8, the media control scenario 6 outputs a reproduction instruction for the post-reproduction menu (step S801). This reproduction instruction is input to the media processing unit 4 (step S802).
Then, the media processing unit 4 outputs voice menu guidance to the telephone terminal 70A (step S803). The menu guidance is, for example, “1 # for transition to inter-user call, 2 # for next message playback,...”.

  When the user who hears the guidance message presses the button on the telephone terminal 70A, the PB sound is input to the media processing unit 4 (step S804). Here, a case where “# 1” is pressed and the corresponding PB sound is input to the media processing unit 4 will be described. The media processing unit 4 notifies the media control scenario 6 of the input PB sound (step S805). With this notification, the media control scenario 6 detects a request for transition to a user-to-user call and notifies the call processing scenario 2 (steps S806 and S807). This notification is received by the call processing scenario 2 (step S808), and the call processing scenario 2 starts a conference mode (conference mode) by the conference function (step S809). When the conference mode activation signal is transmitted to the media control scenario 6 (step S810), the media control scenario 6 instructs the media processing unit 4 to secure U-Plane resources for the conference (steps S811, S812a, S812b). Then, when returning to message playback after the end of a user-to-user call, the media control scenario 6 holds the current playback state (step S813). For example, if the playback of up to the third message has been completed, the playback state is maintained.

After the current playback state is maintained by the media control scenario 6, the call processing scenario 2 outputs an instruction to transfer a connection request to the VM server 10 to the call processing unit 1 (steps S814, S815, and S816).
The call processing unit 1 transmits a call signal to the telephone terminal 70B via the exchange 20 (step S817). This outgoing signal can be realized by, for example, a SIP INVITE message. Thereafter, the call processing unit 1 receives a provisional response message via the exchange 20 (step S818). The provisional response message transmitted in step S818 can be realized by a SIP 183 message, for example. The call processing unit 1 that has received the provisional response message performs an update process by SDP (Session Description Protocol) on the media processing unit 4 in order to start real-time communication (step S819).

  Thereafter, the call processing unit 1 receives a response message via the exchange 20 (step S820). The response message transmitted in step S820 can be realized by, for example, a SIP 200 OK message. Receiving this response message, the call processing unit 1 notifies the call processing scenario 2 of the reception (step S821). Thereby, the call processing scenario 2 recognizes that there is a response from the telephone terminal 70B (step S822), and notifies the media control scenario 6 that there is a user response (step S823). Upon receiving this notification, the media control scenario 6 instructs the media processing unit 4 to start the conference mode (steps S824 and S825).

  Thereafter, a call is established between the telephone terminal 70B and the VM server 10 of the D company carrier network (step S826a), and the voice of this call and between the originating telephone terminal 70A and the VM server 10 of the carrier company D network are established. The voice of the ongoing call (step S826b) is mixed by the conference function of the media processing unit 4. With this conference function, a voice call can be realized between the telephone terminal 70A and the telephone terminal 70B. For example, it is possible to enjoy a conversation between the user of the telephone terminal 70A and the user of the telephone terminal 70B, such as “Hello, long time no see!”.

(When the Conference function is not used)
FIG. 9 is a sequence diagram illustrating signal exchange in the VM server 10 when the conference function is not used, which has been described with reference to FIG. In FIG. 9, the media control scenario 6 outputs a playback instruction for the post-playback menu (step S901). This reproduction instruction is input to the media processing unit 4 (step S902).
Then, the media processing unit 4 outputs voice menu guidance to the telephone terminal 70A (step S903). The menu guidance is, for example, “1 # for transition to inter-user call, 2 # for next message playback,...”.

  When the user who hears the guidance message presses the button on the telephone terminal 70A, a PB sound is input to the media processing unit 4 (step S904). Here, a case where “# 1” is pressed and the corresponding PB sound is input to the media processing unit 4 will be described. The media processing unit 4 notifies the input PB sound to the media control scenario 6 (step S905). By this notification, the media control scenario 6 detects a request for transition to a user-to-user call and notifies the call processing scenario 2 (steps S906 and S907). This notification is received by the call processing scenario 2 (step S908), and a connection request to the telephone terminal 70B is notified to the media control scenario 6 (step S909). When the connection request notification signal is transmitted to the media control scenario 6 (step S910), the media control scenario 6 instructs the media processing unit 4 to hold (release) the U-Plane (steps S911, S912a, S912b). Then, when returning to message playback after the end of a user-to-user call, the media control scenario 6 holds the current playback state (step S913). For example, if the playback of up to the third message has been completed, the playback state is maintained.

After the current playback state is maintained by the media control scenario 6, the call processing scenario 2 outputs an instruction to transfer a connection request to the VM server 10 to the call processing unit 1 (steps S914, S915, and S916).
The call processing unit 1 transmits a call signal to the telephone terminal 70B via the exchange 20 (step S917). This outgoing signal can be realized by, for example, a SIP INVITE message. Thereafter, the call processing unit 1 receives a provisional response message via the exchange 20 (step S918). The provisional response message transmitted in step S918 can be realized by a SIP 183 message, for example.

  Upon receiving the provisional response message in step S918, the call processing unit 1 transmits a switching signal to the telephone terminal 70A (step S919). The telephone terminal 70A that has received this switching signal transmits a response signal (step S920). The switching signal transmitted in step S919 can be realized by, for example, a SIP Re-INVITE message. The response signal transmitted in step S920 can be realized by, for example, a SIP 200 OK message.

Thereafter, the telephone terminal 70B transmits a response message, and the call processing unit 1 receives the response message via the exchange 20 (step S921). This response message can be realized by, for example, a SIP 200 OK message.
After the above processing, a call is established between the telephone terminal 70A and the telephone terminal 70B (step S922). By establishing this call, a voice call can be realized between the telephone terminal 70A and the telephone terminal 70B. For example, it is possible to enjoy a conversation between the user of the telephone terminal 70A and the user of the telephone terminal 70B, such as “Hello, long time no see!”.

(When the recording source is a telephone terminal of another carrier network)
When the recording source of the reproduced message is a telephone terminal of another carrier network, a call is made to the telephone terminal of another carrier network through the gateway exchange. Processing in this case will be described with reference to FIGS.
FIG. 10 is a sequence diagram illustrating processing when the conference function of the VM server 10 is used. In FIG. 10, when a call request to the recording source of the message is output from the telephone terminal 70A during the reproduction of the message (step S1001), the VM server 10 is requested to shift to a call between users ( That is, it is detected that a direct call is requested (step S1002). Then, the VM server 10 outputs voice guidance to the telephone terminal 70A (step S1003). This voice guidance is, for example, “I will call you”.

  Thereafter, the VM server 10 transmits a call signal to the telephone terminal 70B (step S1004). This outgoing signal can be realized by, for example, a SIP INVITE message. The outgoing signal transmitted in step S1004 includes “090BBBBBBBB”, which is the telephone number of the destination telephone terminal 70B, as the destination “To”. Further, the transmission signal transmitted in step S1004 includes “090AAAAAAAAA”, which is the telephone number of the telephone terminal 70A, as the transmission source “From”. Further, the IP address of the D company VM server 10 is added to the outgoing signal transmitted in step S1004.

  The gateway switch 80 that has received the call signal transmits the call signal to the telephone terminal 70A via the switch 20A of another carrier network (for example, the carrier network of company A) (step S1005). The outgoing signal transmitted in step S1005 includes “090BBBBBBBB”, which is the telephone number of the telephone terminal 70A as the destination, as the destination “To”. Further, the transmission signal transmitted in step S1005 includes “090AAAAAAAAA” as the transmission source “From”. The outgoing signal transmitted in step S1005 can be realized by, for example, an ISUP (ISDN user part) IAM (Initial Address Message) message.

  The telephone terminal 70B that has received the outgoing signal via the exchange 20 first transmits a provisional response message (step S1006). This provisional response message is sent to the gateway exchange 80 via the exchange 20A. Further, this provisional response message is received by the VM server 10 of the carrier company D network via the gateway exchange 80 (step S1007). The outgoing signal transmitted in step S1006 can be realized by, for example, an ISUP ACM (Address Complete Message) message. The outgoing signal transmitted in step S1007 can be realized by, for example, a SIP 183 message.

  Next, the telephone terminal 70B transmits a provisional response message (step S1008). This provisional response message is sent to the gateway exchange 80 via the exchange 20A. The gateway exchange 80 transmits a calling message indicating that it is calling (step S1009). The provisional response message transmitted in step S1008 can be realized by, for example, an ISUP CPG (Call Progress) message. The in-call message transmitted in step S1009 can be realized by a SIP 180 message, for example.

  This ringing message is received by the D company VM server 10 via the exchange 20. Then, the VM server 10 mixes the RBT and the voice of the call between the telephone terminal 70A and the VM server 10 by the conference function (steps S1007a and S1007b). Thereby, since RBT can be output to the telephone terminal 70A, the user of the telephone terminal 70A can recognize that the (user's) call of the telephone terminal 70B is in progress. The RBT can be output from the exchange 20 or a server device (not shown).

  Thereafter, the telephone terminal 70B transmits a response message (step S1011). This response message is received by the D company VM server 10 via the exchange 20 (step S1012). The response message transmitted in step S1011 can be realized by, for example, an SIP ANM (Answer Message) message. The response message transmitted in step S1012 can be realized by, for example, a SIP 200 OK message.

  After the above processing, a call is established between the telephone terminal 70B and the VM server 10 of the carrier company D (step S1013a). The voice of this call, the telephone terminal 70A of the caller, and the VM server of the carrier company D network 10 and the voice of the ongoing call (step S1013b) are mixed by the Conference function of the D company VM server 10. With this conference function, a voice call can be realized between the telephone terminal 70A and the telephone terminal 70B. For example, it is possible to enjoy a conversation between the user of the telephone terminal 70A and the user of the telephone terminal 70B, such as “Hello, long time no see!”.

  FIG. 11 is a sequence diagram illustrating processing when the conference function of the VM server 10 is not used. In FIG. 11, when a call request to the recording source of the message is output from the telephone terminal 70A during the reproduction of the message (step S1101), the VM server 10 is requested to shift to a call between users ( That is, it is detected that a direct call has been requested (step S1102). Then, the VM server 10 outputs voice guidance to the telephone terminal 70A (step S1103). This voice guidance is, for example, “I will call you”.

  Thereafter, the VM server 10 transmits a call signal to the telephone terminal 70B (step S1104). This outgoing signal can be realized by, for example, a SIP INVITE message. The outgoing signal transmitted in step S1104 includes “090BBBBBBBB”, which is the telephone number of the destination telephone terminal 70B, as the destination “To”. Further, the outgoing signal transmitted in step S1104 includes “090AAAAAAAAA”, which is the telephone number of the telephone terminal 70A, as the source “From”. Furthermore, the IP address of the D company VM server 10 is added to the outgoing signal transmitted in step S1104.

  The gateway switch 80 that has received the call signal transmits the call signal to the telephone terminal 70A via the switch A of the carrier network of the company A (step S1105). The outgoing signal transmitted in step S1105 includes “090BBBBBBBB”, which is the telephone number of the telephone terminal 70A as the destination, as the destination “To”. In addition, the transmission signal transmitted in step S1105 includes “090AAAAAAAAA” as the transmission source “From”. The outgoing signal transmitted in step S1105 can be realized by an ISUP IAM message, for example.

  The telephone terminal 70B that has received the outgoing signal via the exchange 20 first transmits a provisional response message (step S1106). This provisional response message is sent to the gateway exchange 80 via the exchange 20A. Further, this provisional response message is received by the VM server 10 of the carrier company D network via the gateway exchange 80 (step S1107). The outgoing signal transmitted in step S1106 can be realized by, for example, an ISUP ACM message. The outgoing signal transmitted in step S1107 can be realized by, for example, a SIP 183 message.

  Upon receiving the provisional response message in step S1107, the company D VM server 10 transmits a switching signal to the telephone terminal 70A (step S1108). The telephone terminal 70A that has received this switching signal transmits a response signal (step S1109). The switching signal transmitted in step S1108 can be realized by, for example, a SIP Re-INVITE message. The response signal transmitted in step S1109 can be realized by, for example, a SIP 200 OK message.

  Next, the telephone terminal 70B transmits a provisional response message (step S1110). This provisional response message is sent to the gateway exchange 80 via the exchange 20A. The gateway exchange 80 transmits a calling message indicating that the calling is in progress (step S1112). The provisional response message transmitted in step S1110 can be realized by, for example, an ISUP CPG message. The in-call message transmitted in step S1112 can be realized by a SIP 180 message, for example.

Here, the provisional response message transmitted in step S1107 is a response message that requires arrival confirmation, and the arrival confirmation message is exchanged between the gateway exchange 80 and the D company VM server 10 (step S1111).
The in-call message transmitted in step S <b> 1112 is received by the company D VM server 10 via the exchange 20. Then, RBT is output to the telephone terminal 70A (step S1113). Thereby, the user of the telephone terminal 70A can recognize that the telephone terminal 70B (user's) is being called. The RBT can be output from the exchange 20 or a server device (not shown).

Thereafter, the telephone terminal 70B transmits a response message (step S1114). This response message is transmitted to the D company VM server 10 (step S1115). The response message transmitted in step S1114 can be realized by a SIP ANM message, for example. The response message transmitted in step S1115 can be realized by, for example, a SIP 200 OK message.
After the above processing, a call is established between the telephone terminal 70A and the telephone terminal 70B (step S1116). By establishing this call, a voice call can be realized between the telephone terminal 70A and the telephone terminal 70B. For example, it is possible to enjoy a conversation between the user of the telephone terminal 70A and the user of the telephone terminal 70B, such as “Hello, long time no see!”.

(Voice mail system)
By providing the above voice mail server device in each carrier network, a voice mail system across the carrier network can be realized. Then, paying attention to the function of realizing a call with the recording source user after playing the message, the present system makes a call request to the recording source of the message in a state where the call continues with the voice mail server device. And a telephone terminal (for example, telephone terminal 70A in FIG. 2) that transmits to the voice mail server device. According to this voice mail system, it is possible to seamlessly shift to a direct call with the recording source user while continuing the call after the message reproduction process, and it is possible to realize a highly convenient and easy-to-use voice mail service.

(Summary)
According to the present invention, after a message reproduction process for reproducing a recorded message by a call from a telephone terminal to a voice mail server device, a direct call request is accepted and sent to the recording source of the message. After the playback process, it is possible to seamlessly shift to a direct call with the recording source user while continuing the call, and to realize a highly convenient and easy-to-use voice mail service.

Further, after the direct call with the recording source user is terminated, the message reproduction process by the call from the telephone terminal is continued, so that the original message reproduction process can be restored, and an easy-to-use voice mail service can be realized.
It should be noted that the scope of the present invention is not limited to the illustrated and described exemplary embodiments, but includes all embodiments that provide the same effects as those intended by the present invention. Further, the scope of the invention is not limited to the combinations of features of the invention defined by the claims, but may be defined by any desired combination of particular features among all the disclosed features. .

  The present invention can be used for realizing a call between a user who plays and listens to a message recorded by a voice mail service and a user who records the message.

DESCRIPTION OF SYMBOLS 1 Call processing part 2 Call processing scenario 4 Media processing part 5 Media storage part 6 Media control scenario 10, 10A Voice mail server apparatus 20, 20A Exchange 30 SMS transmission server 70A, 70B Telephone terminal 80 Gateway exchange

Claims (5)

  A message playback unit that plays back a recorded message by a first call from a first telephone terminal, and a call to the recording source of the message in a state in which the first call continues after message playback processing by the message playback unit A call accepting unit that accepts a request; and a calling unit that makes a call to a second telephone terminal that is a recording source of the message when the call accepting unit accepts a call request. A voice mail server device for realizing a call with a second telephone terminal.   The first telephone terminal and the second telephone terminal are mixed by mixing the voice of the first call between the first telephone terminal and the own apparatus and the voice of the second call from the own apparatus to the second telephone terminal. The voice mail server device according to claim 1, further comprising a conference unit that realizes a telephone call.   The calling unit transmits a call between the first telephone terminal and the second telephone terminal by transmitting a signal for calling to the second telephone terminal including the address of the first telephone terminal. The voice mail server device according to claim 1, wherein the voice mail server device is realized.   2. The replay call continuation unit further comprising a replay call continuation unit for continuing a message replay process by the first call from the first telephone terminal when a disconnection signal is transmitted from the second telephone terminal. Item 4. The voice mail server device according to any one of Items 3 to 3.   5. A voice mail system comprising: the voice mail server apparatus according to claim 1; and a telephone terminal that transmits the call request to the voice mail server apparatus.

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