CN100568896C - The interactive system of IP phone voice answer-back and method thereof - Google Patents
The interactive system of IP phone voice answer-back and method thereof Download PDFInfo
- Publication number
- CN100568896C CN100568896C CNB2006100532776A CN200610053277A CN100568896C CN 100568896 C CN100568896 C CN 100568896C CN B2006100532776 A CNB2006100532776 A CN B2006100532776A CN 200610053277 A CN200610053277 A CN 200610053277A CN 100568896 C CN100568896 C CN 100568896C
- Authority
- CN
- China
- Prior art keywords
- server end
- user side
- user
- additional channel
- processor
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Expired - Fee Related
Links
- 230000002452 interceptive effect Effects 0.000 title claims abstract description 16
- 238000000034 method Methods 0.000 title claims abstract description 16
- 239000000463 material Substances 0.000 claims abstract description 15
- 230000005540 biological transmission Effects 0.000 claims abstract description 14
- 230000011664 signaling Effects 0.000 claims description 23
- 238000007906 compression Methods 0.000 claims description 11
- 230000006835 compression Effects 0.000 claims description 11
- 238000006243 chemical reaction Methods 0.000 claims description 7
- 238000005538 encapsulation Methods 0.000 claims description 7
- 230000006837 decompression Effects 0.000 claims description 6
- 230000004044 response Effects 0.000 claims description 5
- 239000000284 extract Substances 0.000 claims description 4
- 230000006854 communication Effects 0.000 description 15
- 238000004891 communication Methods 0.000 description 10
- 239000003795 chemical substances by application Substances 0.000 description 8
- 238000001514 detection method Methods 0.000 description 3
- 230000009977 dual effect Effects 0.000 description 3
- 230000006870 function Effects 0.000 description 3
- 230000000977 initiatory effect Effects 0.000 description 3
- 230000005236 sound signal Effects 0.000 description 3
- 238000011161 development Methods 0.000 description 2
- 238000010586 diagram Methods 0.000 description 2
- 238000005516 engineering process Methods 0.000 description 2
- 230000010354 integration Effects 0.000 description 2
- 238000012546 transfer Methods 0.000 description 2
- 230000009471 action Effects 0.000 description 1
- 230000009286 beneficial effect Effects 0.000 description 1
- 238000004364 calculation method Methods 0.000 description 1
- 238000013144 data compression Methods 0.000 description 1
- 230000007812 deficiency Effects 0.000 description 1
- 238000001914 filtration Methods 0.000 description 1
- 239000012634 fragment Substances 0.000 description 1
- 230000008676 import Effects 0.000 description 1
- 239000000203 mixture Substances 0.000 description 1
- 230000008569 process Effects 0.000 description 1
- 238000012545 processing Methods 0.000 description 1
- 238000004088 simulation Methods 0.000 description 1
Images
Landscapes
- Telephonic Communication Services (AREA)
- Data Exchanges In Wide-Area Networks (AREA)
Abstract
A kind of interactive system of IP phone voice answer-back, comprise and reply user side, IP-based switching's server and answering server end, reply user side and be provided with user side additional channel processor, the data channel that is used to manage special transmission dtmf signal, the parameter of definition user additional channel; The answering server end is provided with server end additional channel processor, the data channel that is used to manage special reception dtmf signal, the parameter of definition server additional channel; Hear the voice suggestion of language material database as the user after, will select the button input of user interface section as required, user side additional channel processor sends this key assignments code; After receiving the key assignments code signal, answering server end additional channel processor is passed to service agent module with the key assignments code that receives.And the exchange method that proposes a kind of IP phone voice answer-back.The present invention is simple in structure, can effectively avoid situation, the reliability height of mistake identification.
Description
(1) technical field
The present invention relates to the application in computer telephone integration (Computer Telephone Integration) field, specifically be used in interactive system and the method for using phone and voice suggestion to reply in the IP telephony system.
(2) background technology
In traditional public switched telephone network system (PSTN), once typical communication process comprises 3 stages: the stage 1 is " call setup ", and the stage 2 is " voice communication ", and the stage 3 is " line disconnection ".Stage 1 wherein and stage 3 are signaling control procedure (as the SS7 signaling), and the stage 2 is actual communication process.We imagine a scene of making a phone call, user A picks up terminal equipment PA (as telephone set) (this action is referred to as off-hook), telephone number (user A is the calling party) by button input the other side, so just initiated call request one time, route switching by stored-program control exchange, this request signal is sent to the terminal equipment PB (such as telephone set) that is connected on user B (user B the is the callee) circuit, after ring detection module among the PB detects the bell signal that imports into, produce ringing tone to remind user B, user B hangs on, begin conversation, stage this moment 1 finishes, and enters the stage 2.Behind the end of conversation, both sides put down microphone (on-hook), finish the stage 3 rapidly.In the stage 2, voice convert sound wave to the signal of telecommunication, and transmit by telephone line by acoustoelectric sensor.
Along with popularizing rapidly and application of the development of compunication, particularly Internet technology, the IP telephony system (VoIP) of communication Network Based has appearred.Once typical communication process in VoIP and be similarly, just the mode difference that realizes in traditional telephone relation process.Comprise 3 stages equally, communication process also is based on switch, and is referred to as " soft switch " (being realized by program fully).The signaling system of using the most widely has SIP, H323 and MGCP etc. at present.VoIP system, the transmission of any data comprises control signaling and speech data, all is based on IP network.
For telephone system, main application is voice communication.But along with The development in society and economy, a lot of value-added services also having occurred uses, interactive automatic speech service system (IVR) wherein is to use a kind of very widely, can be used in as customer service calls center (callcenter), the automatic stock exchange trading system of phone, fields such as telephone bank's self-aid system.In these were used, the user was undertaken by the mode of input button according to recipient's machine talk prompting alternately.Recipient's terminal equipment is made corresponding operation according to receiving the pairing dual tone multi-frequency dtmf signal of button and discerning its respective value.This is the cardinal principle of IVR system works.Dual tone multi-frequency dtmf (Dual Tone Multi-Frequency) signaling is used in worldwide at present on touch-tone telephone, because of it provides higher dialing speed, has replaced the dial impulse signaling that traditional rotating disc type telephone set uses.Each key 0-9 on telephone keypad, #, *, A-D, all corresponding waveform that constitutes by the audio signal stack of two frequencies.The frequency of these two audio signals is from two groups of preallocated group of frequencies: line frequency group or row are group frequently.Each is to numeral of the unique expression of such audio signal or symbol.Produce dtmf signal, the waveform that utilizes the sine-wave superimposed of two different frequencies to form later on exactly is that two sine-wave superimposed of 1209HZ and 697HZ form as " 0 " key by frequency.People's ear can be heard dtmf signal, but can't discern frequency component wherein, so can't discern corresponding key.
Transmit the 2nd stage that dtmf signal occurs in communication process in interactive automatic speech service system (IVR), this character that shows the DTMF waveform is identical with the character of the waveform of speaking, and all belongs to " data-signal ", but not " control signal ".In order to discern the pairing key assignments of dtmf signal, must use a kind of DTMF identification module at the IVR end.This identification module can be with hard-wired, can decode to dtmf signal as the MT8870 DTMF recipient that MITEL company produces, and realizes the separating filtering and the decoding function of dtmf signal, exports four parallel-by-bit binary codes of corresponding 16 kinds of combination of frequencies.Also can realize, need utilize the method for Digital Signal Processing with software, the existence of two sine waves of search in frequency domain, amount of calculation is bigger.
Traditional IVR system has had very widely and has used, but had following problem:
(1) support of DTMF identification module must be arranged;
(2) because DTMF utilizes speech channel to transmit, the composition of DTMF is arranged in our voice, even be easy to produce the waveform of DTMF, so the situation of the wrong identification of meeting occurs.
Because communications in IP network is to adopt the circuit switched (circuit switch) in packet switch (packet switch) rather than the traditional field and the unsettled characteristic of IP network, for the problems referred to above (2), situation is more serious in VoIP system.Main cause has two aspects, the use of first data compression protocol.In order to reduce the amount of voice data of transmission, before transmitting, data need data are compressed, and compression protocol commonly used has G711, G723 and G729 etc.These compression protocols are lossy compression method, and compression ratio does not wait, and dtmf signal will produce certain distortion behind decompress(ion), and compression ratio is high more, and the distortion situation behind the decompress(ion) can be more serious, thereby cause mistake identification.Its two, under the unfavorable situation of network service, have the phenomenon of data packet loss, also can produce the DTMF mistake and discern phenomenon.
The basis of IP telephony system is IP data communications, no matter is control signaling or speech data, all transmits by UDP or TCP grouping.Its dtmf signal of system that uses is to be sent by the UDP grouping through RTP encapsulation back at present.
(3) summary of the invention
For the complex structure of the interactive system that overcomes existing IP phone voice answer-back, the situation that has mistake identification, the deficiency that reliability is not high, the invention provides the interactive system and the method thereof of the high IP phone voice answer-back of a kind of situation, reliability simple in structure, that can effectively avoid mistake identification.
The technical solution adopted for the present invention to solve the technical problems is:
A kind of interactive system of IP phone voice answer-back, comprise and reply user side, VoIP soft switching server and answering server end, the described user side of replying comprises user side control unit, user side user interface section and user side network interface unit, described user side control unit comprises: the user side signal processor is used for generation, explanation and the conversion of managing signaling passage, also responsible signaling; The user side Media Processor is used to manage speech channel, and is responsible for the compression and decompression of speech data after the digitlization and the encapsulation and the dismounting of data; The user side dispatch processor, the Task Distribution that is used for setting is given signal processor and Media Processor; User side user interface section, user side network interface unit connect the user side control unit, and described user side network interface unit connects the VoIP soft switching server; Described answering server end comprises server end control unit, server end user interface section, server end network interface, language material database and service broker, described server end control unit comprises the server end signal processor, is used for generation, explanation and the conversion of managing signaling passage, also responsible signaling; The server end Media Processor is used to manage speech channel, and is responsible for the compression and decompression of speech data after the digitlization and the encapsulation and the dismounting of data; The server end dispatch processor, the Task Distribution that is used for setting is given signal processor and Media Processor; Server end user interface section, server end network interface unit Connection Service device end subscriber controller, the server end user interface section connects the language material database, and described server end network interface connects VoIP soft switching server and service agent module;
The described user side of replying also comprises: user side additional channel processor, the data channel that is used to manage special transmission dtmf signal, IP address, transport-type and the port of definition user side additional channel; Described answering server end also comprises: server end additional channel processor, the data channel that is used to manage special transmission dtmf signal, IP address, transport-type and the port of definition server end additional channel;
Hear the voice suggestion of language material database as the user after, will select the button input of user interface section as required, user side additional channel processor sends this key assignments code; After the answering server termination was received the key assignments code signal, server end additional channel processor was passed to service agent module with the key assignments code that receives.
As preferred a kind of scheme: the described user side of replying comprises that also the user side channel parameters that is used to set in advance user side additional channel parameter is provided with module, and described answering server end comprises that also the server end channel parameters that is used to set in advance server end additional channel parameter is provided with module.
As preferred another kind of scheme: in the data communication protocol module, comprise that the parameter that is used to be provided with user side additional channel, server end additional channel parameter is provided with module.
Described key assignments code is the ASCII character of numeral or symbol correspondence.
A kind of exchange method of IP phone voice answer-back may further comprise the steps:
(1), the user carries out dial-up operation, generates call signal, sends to the VoIP soft switching server by network interface, with this call forward to the answering server end;
(2), the answering server termination receives this call signal, as is operated in non-automatic response status, the loud speaker by user interface sends vibration, waiting answering; As be operated in the automatic-answering back device state, and simulate off-hook, and off-hook information is passed back to the user, enter talking state;
(3), the answering server end extracts voice snippet and playback from the language material database;
(4), after the user hears voice suggestion, selection key input as required, the key assignments code sends by the user side additional channel;
(5), the answering server end receives the key assignments code by the server end additional channel, and discerns this key assignments code;
(6), the key assignments code after will discerning passes to service agent module, realizes various service business.
As preferred scheme:, set in advance IP address, transport-type and the port of user's additional channel at the described user side of replying; At described answering server end, set in advance IP address, transport-type and the port of server additional channel.
As preferred another kind of scheme: in the data communication protocol module, set in advance IP address, transport-type and the port of user's additional channel, server additional channel parameter.
Described key assignments code is the ASCII character of numeral or symbol correspondence.
Technical conceive of the present invention is: set up a new transmission channel respectively replying user side and answering server end, called after additional channel AC (Additional Channel) transmits dtmf signal specially.In IP communication, the foundation of new transmission channel is very easy to, and just can determine a passage by IP address, transport-type (UDP/TCP) and port numbers tlv triple.VoIP system with the employing Session Initiation Protocol is an example, and system is made up of two transmission channels.One is signalling path, and default employing TCP mode is transmitted, and port is 5060.The 2nd, data channel, concrete parameter is consulted to determine by communicating pair by the control signaling that signalling path transmits.In the VoIP system of Session Initiation Protocol, the parameter of AC passage is determined, can adopt predefined, also can be by being determined by master control signaling negotiation method.
Information transmitted is dtmf signal at present in the AC passage, also can transmit other information in the future as required, the function of expanding system.
The dtmf signal that transmits in the AC passage, no longer based on the signal after two sine-wave superimposed, but the key assignments code can use the ASCII character of numeral or symbol correspondence to represent.
The dtmf signal form that transmits in the AC passage can define with the XML standard, and is as follows:＜dtmf〉1＜/dtmf 〉, expression enter key 1.
Beneficial effect of the present invention mainly shows: 1, omitted traditional DTMF detection module, simplified the structure; 2, the information of Chuan Songing is a character string, handles to be very easy to, the situation of having avoided flase drop to survey; 3, transmission reliability height.
(4) description of drawings
Fig. 1 is the theory diagram of the interactive system of IP phone voice answer-back.
Fig. 2 is the contrast schematic diagram of dtmf signal different expression form in the present invention and existed system.
(5) embodiment
Below in conjunction with accompanying drawing the present invention is further described.
With reference to Fig. 1, Fig. 2, a kind of interactive system of IP phone voice answer-back comprises that replying user side 1, second replys user side 4, VoIP soft switching server 2 and answering server end 3.
VoIP system with the employing Session Initiation Protocol is an example, replys user side 1, answering server end 3 among Fig. 1, replys voip user's terminal that second user side 4 all is based on CPU.They have identical basic structure.Answering server end 3 is operated in the automatic-answering back device state as the voice answer-back server, has comprised two additional components.
The described user side 1 of replying comprises user side controller unit 11, user side user interface section 12 and user side network interface unit 13.Described user side controller unit 11 comprises: user side signal processor 300 is used for generation, explanation and the conversion of managing signaling passage, also responsible signaling; User side Media Processor 400 is used to manage speech channel, and is responsible for the compression and decompression of speech data after the digitlization and the encapsulation and the dismounting of data; User side dispatch processor 200, the Task Distribution that is used for setting is given signal processor 300 and Media Processor 400.User side user interface section 12, user side network interface unit 13 connect user side controller unit 11, and described user side network interface unit 13 connects VoIP soft switch 2.Described answering server end 3 comprises server end controller unit 31, server end user interface section 32, server end network interface unit 33, language material database 6 and service broker 5, described server end controller unit 31 comprises server end signal processor 1300, is used for generation, explanation and the conversion of managing signaling passage, also responsible signaling; Server end Media Processor 1400 is used to manage speech channel, and is responsible for the compression and decompression of speech data after the digitlization and the encapsulation and the dismounting of data; Server end dispatch processor 1200, the Task Distribution that is used for setting is given signal processor and Media Processor.Server end user interface section 32, server end network interface unit 33 Connection Service device side controller unit 31, server end user interface section 32 connects language material database 6, and described server end network interface unit 33 connects IP network 10 and service agent module 5;
The described user side 1 of replying also comprises: user side additional channel processor 500, be used to manage the data channel of special transmission dtmf signal, IP address, transport-type and the port of definition user side additional channel are responsible for the transmission and the reception of DTMF code (non-waveform signal).These parts are core contents of this patent; Described answering server end 3 also comprises: server end additional channel processor 1500, the data channel that is used to manage special transmission dtmf signal, IP address, transport-type and the port of definition server end additional channel;
Hear the voice suggestion of language material database as the user after, will select the button input of user interface section 12 as required, user side additional channel processor 500 sends this key assignments code.After receiving the key assignments code signal, server end additional channel processor 1500 is passed to service agent module 5 with the key assignments code that receives.
The described user side 1 of replying comprises that also the user side channel parameters that is used to set in advance user side additional channel parameter is provided with module, and described answering server end 3 comprises that also the server end channel parameters that is used to set in advance server end additional channel parameter is provided with module.Perhaps, in the data communication protocol module, comprise that the parameter that is used to be provided with user side additional channel, server end additional channel parameter is provided with module.Described key assignments code is the ASCII character of numeral or symbol correspondence
User interface section 12 expression ordinary telephone set and the mutual parts of user comprise the modulus of receiver, loud speaker, keypad, LCDs (optional) and voice and digital-to-analogue conversion etc.; Network interface unit 13 is responsible for data are sent to IP network and the data of accepting to transmit from IP network; The control that controller unit 11 is realized user interface section 12 and network interface unit 13.
The user carries out dial-up operation by the keyboard of user side user interface section 12.Controller unit 11 generates call signal according to user's input by signal processor 300, sends to IP network end 10 by network interface unit 13, arrives soft switch 2.
IP network end 10 is IP network clouds.
The function of soft switch 2 is for being responsible for forward call to the called party.
Solicited message enters server end network interface unit 33.Network interface unit 33 is submitted to controller unit 31 with request.
Controller unit 31 will be asked notice user interface section 32, if answering server end 3 is operated in non-automatic response status, then user interface section 32 will send ring by loud speaker, wait for answering of user.If answering server end 3 is operated in the automatic-answering back device state, controller unit 31 will be controlled the simulation off-hook, and off-hook information is sent to IP network end 10 by network interface unit 33, arrive at and reply user side 1.Finished establishment of connection this moment.Reply user side 1 and answering server end 3 all enters talking state.
In language material database 6, deposit the sound bite of prerecording in advance, as: " welcoming to use the phone stock exchange trading system ", " by 1 Chinese prompt, by 2 English prompts " etc.
After the user hears voice suggestion, selection key input as required.Institute's key value is obtained by user interface section 12, and submits to controller unit 11.If adopt traditional DTMF occupation mode, controller unit 11 will indicate Media Processor 400 to be responsible for generating the signal of corresponding two sine-wave superimposed, and send out by speech channel, controller unit 31 must be accepted this signal at speech channel, and, there is the situation of mistake identification by an extra DTMF detection module identification key assignments.In this patent, controller unit 11 will indicate AC processor 500 to send the pairing code of this key, and form is＜dtmf〉x＜dtmf 〉.Transmit by the AC passage, rather than send by speech channel, controller unit 31 is accepted at the AC passage, is discerned by 31 AC processor 1500, because the information that transmits is a character string, handles being very easy to, and has the possibility of mistake identification hardly.
Controller unit 31 is passed to service agent module with the key that receives, and service agent module is service broker's parts.Can realize various service business.
Present embodiment adopts AC channel transfer dtmf key sign indicating number, needing to have avoided the requirement of special detectors by speech channel transmitting DTMF waveform, while has also been avoided the situation of flase drop survey, and is simple and practical, thereby a kind of implementation method of novel interactive voice response is provided.
With reference to Fig. 1, Fig. 2, a kind of exchange method of IP phone voice answer-back may further comprise the steps:
(1), the user carries out dial-up operation, generates call signal, sends to VoIP soft switching server 10 by network interface, with this call forward to answering server end 3;
(2), answering server end 3 receives these call signals, as is operated in non-automatic response status, the loud speaker by user interface sends vibration, waiting answering; As be operated in the automatic-answering back device state, and simulate off-hook, and off-hook information is passed back to the user, enter talking state;
(3), answering server end 3 extracts voice snippet and playback from language material database 6;
(4), after user side hears voice suggestion, selection key input as required, the key assignments code sends by user's additional channel;
(5), server end receives the key assignments code by the server end additional channel, and discerns this key assignments code;
(6), the key assignments code after will discerning passes to service agent module 5, realizes various service business.
At the described user side 1 of replying, set in advance IP address, transport-type and the port of user's additional channel; At described answering server end 3, set in advance IP address, transport-type and the port of server additional channel.Or, in the data communication protocol module, set in advance IP address, transport-type and the port of user's additional channel, server additional channel parameter.Described key assignments code is the ASCII character of numeral or symbol correspondence.
Claims (6)
1, a kind of interactive system of IP phone voice answer-back, comprise and reply user side, VoIP soft switching server and answering server end, the described user side of replying comprises user side control unit, user side user interface section and user side network interface unit, described user side control unit comprises: the user side signal processor is used for generation, explanation and the conversion of managing signaling passage, also responsible signaling; The user side Media Processor is used to manage speech channel, and is responsible for the compression and decompression of speech data after the digitlization and the encapsulation and the dismounting of data; The user side dispatch processor, the Task Distribution that is used for setting is given signal processor and Media Processor; User side user interface section, user side network interface unit connect the user side control unit, and described user side network interface unit connects the VoIP soft switching server; Described answering server end comprises server end control unit, server end user interface section, server end network interface, language material database and service broker, described server end control unit comprises the server end signal processor, is used for generation, explanation and the conversion of managing signaling passage, also responsible signaling; The server end Media Processor is used to manage speech channel, and is responsible for the compression and decompression of speech data after the digitlization and the encapsulation and the dismounting of data; The server end dispatch processor, the Task Distribution that is used for setting is given signal processor and Media Processor; Server end user interface section, server end network interface unit Connection Service device end subscriber controller, the server end user interface section connects the language material database, and described server end network interface connects VoIP soft switching server and service agent module; It is characterized in that:
The described user side of replying also comprises: user side additional channel processor, the data channel that is used to manage special transmission dtmf signal, IP address, transport-type and the port of definition user side additional channel; Described answering server end also comprises: server end additional channel processor, the data channel that is used to manage special transmission dtmf signal, IP address, transport-type and the port of definition server end additional channel;
Hear the voice suggestion of language material database as the user after, will select the button input of user interface section as required, user side additional channel processor sends this key assignments code.After the answering server termination was received the key assignments code signal, server end additional channel processor was passed to service agent module with the key assignments code that receives.
2, the interactive system of IP phone voice answer-back as claimed in claim 1, it is characterized in that: the described user side of replying comprises that also the user side channel parameters that is used to set in advance user side additional channel parameter is provided with module, and described answering server end comprises that also the server end channel parameters that is used to set in advance server end additional channel parameter is provided with module.
3, the interactive system of IP phone voice answer-back as claimed in claim 1 or 2 is characterized in that: described key assignments code is the ASCII character of numeral or symbol correspondence.
4, the exchange method of the interactive system of a kind of usefulness IP phone voice answer-back as claimed in claim 1 realization may further comprise the steps:
(1), the user carries out dial-up operation, generates call signal, sends to the VoIP soft switching server by network interface, with this call forward to server end;
(2), the answering server termination receives this call signal, as is operated in non-automatic response status, the loud speaker by user interface sends vibration, waiting answering; As be operated in the automatic-answering back device state, and simulate off-hook, and off-hook information is passed back to the user, enter talking state;
(3), the answering server end extracts voice snippet and playback from the language material database;
(4), after the user hears voice suggestion, selection key input as required, the key assignments code sends by the user side additional channel;
(5), the answering server end receives the key assignments code by the server end additional channel, and discerns this key assignments code;
(6), the key assignments code after will discerning passes to service agent module, realizes various service business.
5, the exchange method of a kind of IP phone voice answer-back as claimed in claim 4 is characterized in that: at described user side, set in advance IP address, transport-type and the port of user's additional channel; At described server end, set in advance IP address, transport-type and the port of server additional channel.
6, as the exchange method of claim 4 or 5 described a kind of IP phone voice answer-backs, it is characterized in that: described key assignments code is the ASCII character of numeral or symbol correspondence.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNB2006100532776A CN100568896C (en) | 2006-09-05 | 2006-09-05 | The interactive system of IP phone voice answer-back and method thereof |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CNB2006100532776A CN100568896C (en) | 2006-09-05 | 2006-09-05 | The interactive system of IP phone voice answer-back and method thereof |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN1937673A CN1937673A (en) | 2007-03-28 |
| CN100568896C true CN100568896C (en) | 2009-12-09 |
Family
ID=37954927
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CNB2006100532776A Expired - Fee Related CN100568896C (en) | 2006-09-05 | 2006-09-05 | The interactive system of IP phone voice answer-back and method thereof |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN100568896C (en) |
Citations (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1407445A (en) * | 2001-08-24 | 2003-04-02 | 华为技术有限公司 | Method for realizing interactive answer/speech mailbox facility by software |
| CN1457180A (en) * | 2002-05-10 | 2003-11-19 | 北京艾尼通科技有限公司 | Method for realizing interacting voice response in IP network |
| CN1809038A (en) * | 2005-01-19 | 2006-07-26 | 上海嵌威信息技术有限公司 | Exchange IP address and its calling method and intelligent IP telephone set |
-
2006
- 2006-09-05 CN CNB2006100532776A patent/CN100568896C/en not_active Expired - Fee Related
Patent Citations (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1407445A (en) * | 2001-08-24 | 2003-04-02 | 华为技术有限公司 | Method for realizing interactive answer/speech mailbox facility by software |
| CN1457180A (en) * | 2002-05-10 | 2003-11-19 | 北京艾尼通科技有限公司 | Method for realizing interacting voice response in IP network |
| CN1809038A (en) * | 2005-01-19 | 2006-07-26 | 上海嵌威信息技术有限公司 | Exchange IP address and its calling method and intelligent IP telephone set |
Also Published As
| Publication number | Publication date |
|---|---|
| CN1937673A (en) | 2007-03-28 |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| JP3101768B2 (en) | Voice Call Notification and Message Control by Data Path | |
| CN100559802C (en) | Be used between IP and analog telephone, realizing the method and apparatus of parallel work-flow | |
| CN102065187B (en) | Method and system for providing a call answering service between a source telephone and a target telephone | |
| US7593514B1 (en) | Methods and apparatuses for communicating between messaging and telephony systems | |
| CN100521631C (en) | System and method for generating ringbace tone | |
| US20030228010A1 (en) | Caller information display methods and systems | |
| CN101001287A (en) | Voice server and voice answer method | |
| CN101610465A (en) | Communication method and communication system for converting text information into voice information | |
| Keagy | Integrating voice and data networks | |
| CN101072260A (en) | Method for realizing voice short message based on network telephone and voice short-message system | |
| US8675638B2 (en) | Method and apparatus for enabling dual tone multi-frequency signal processing in the core voice over internet protocol network | |
| CN100426826C (en) | Method for realizing message-leaving lamp and communication system | |
| KR20020011668A (en) | Internet based telecommunication service method | |
| US20010018656A1 (en) | Method and server for setting up a communication connection via an IP network | |
| US11374985B1 (en) | Communications approach and implementations therefor | |
| CN1976376B (en) | Method for calling session, IP telephone system and IP telephone terminal | |
| CN100542196C (en) | The method and system that a kind of toll operator of realization inserts | |
| CN100568896C (en) | The interactive system of IP phone voice answer-back and method thereof | |
| KR20000014120A (en) | Telephone switching system | |
| CN1491006A (en) | Method for realizing no answer transferring business by calling based on internet speech gate | |
| CN101742010A (en) | Method and device for network extension to dial traditional extension and computer program product thereof | |
| US8837459B2 (en) | Method and apparatus for providing asynchronous audio messaging | |
| US20110123010A1 (en) | Method and system for transmitting caller identification information in a conference call | |
| CN1071071C (en) | Method and device for common telephone set to have the function of direct connection to Internet without agent | |
| CN100440911C (en) | A method for adjusting IP telephone volume |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| C06 | Publication | ||
| PB01 | Publication | ||
| C10 | Entry into substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| C14 | Grant of patent or utility model | ||
| GR01 | Patent grant | ||
| CF01 | Termination of patent right due to non-payment of annual fee |
Granted publication date: 20091209 Termination date: 20140905 |
|
| EXPY | Termination of patent right or utility model |