CN102045466A - Method for realizing enterprise voice over internet phone (VOIP) immediate call - Google Patents

Method for realizing enterprise voice over internet phone (VOIP) immediate call Download PDF

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CN102045466A
CN102045466A CN2010105696389A CN201010569638A CN102045466A CN 102045466 A CN102045466 A CN 102045466A CN 2010105696389 A CN2010105696389 A CN 2010105696389A CN 201010569638 A CN201010569638 A CN 201010569638A CN 102045466 A CN102045466 A CN 102045466A
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user
server
call
targeted customer
address
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CN102045466B (en
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曲旸
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DALIAN TIANYI SOFTWARE Co Ltd
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DALIAN TIANYI SOFTWARE Co Ltd
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Abstract

The invention discloses a method for realizing enterprise voice over internet phone (VOIP) immediate call, which is characterized by comprising the following steps: a) initialization is set; b) a fixed phone is dialed by a user, and a connection is established by a server; c) the user is further dialed and analyzed by the server; and the dialing information is corresponded with an addressing address stored in the server side of a calling user; d) target user name information is analyzed out by the server side of the calling user, and is sent to a target address server; e) the information is received by the target user server, and a corresponding extension number is extracted; f) a calling request is launched to a target user client, and simultaneously a calling request is sent out to an extension number of the target user; and g) the client of the target user responses, simultaneously the extension set of the target user rings, and a user B receives the request through the client and establishes the conversation or lifts own extension set to receive the request to establish the conversation. The method can realize video conference and fax in the same manner as the existing internet phone, and is free for all.

Description

A kind of method that realizes the VOIP of enterprise immediate
Technical field
The present invention relates to a kind of method that realizes the VOIP of enterprise immediate, relate in particular to a kind of digital equivalent in being the instant communicating method of separator addressing with the character.
Background technology
The employed communication modes of present most of user all is a traditional telephone system, and traditional telephone system comprises telephone exchange, telephone set, telephone wire.The user calls all to be needed to pay communication cost to operator, and especially for enterprise, these communication costs have increased a part of cost.
Along with the rise of VOIP technology, network phone system is by approval of a lot of enterprises and use, because it with respect to traditional telephone system, has reduced certain cost.But the software network telephone system also has its weak point, is example with Skype (is used network telephone software very widely), and it can only realize that software dials the conversation between landline telephone (or mobile phone), software and the software.Conversation between software and the software, VoP can be accomplished free by Internet transmission; Software is dialed landline telephone (or mobile phone), because the networking telephone and black phone is separate, if put through, networking telephone merchant need pay certain expense to black phone operator, and the user must pay certain expense (as: skype supplements with money) to networking telephone merchant; Skype can't realize that landline telephone is dialed soft phone and landline telephone is dialed landline telephone.
Along with development of internet technology, new solution is constantly proposed, VOIP solution with Cisco is an example, as: shown in Fig. 1 Cisco-voip, the hardware device that needs: router (Router), switch (Switch), telephone exchange (PBX), computer (PC), telephone set (Phone), netting twine, telephone wire.The computer expert crosses netting twine and is connected on the switch, and switch is connected on the router by netting twine, and router uses its speech interface to link to each other with PBX by telephone wire, and telephone set is connected on the PBX by telephone wire.Implementation: calling out company B with first company is example, the self-defined prefix number of first company management person, when before the first company personnel is calling, dialling this prefix number earlier, PBX can hand to this VoP the router of first company, and the router of first company finds the router (the call data transmission course is shown in Fig. 2 Cisco-viop process) of the company B that configures in advance by addressing.Do not have soft end in this cover solution, so can't realize dialling mutually between soft phone and landline telephone, soft phone and the soft phone, can only accomplish to dial mutually between the landline telephone free, because prefix number is self-defining by enterprise network management person, and enterprise network management person must be in advance does unified planning with the routing iinformation in the router of first, the company B of conversing mutually, so the outer router intercommunication with it of planning.The prerequisite of this solution is that these two companies are necessary between general headquarters and the branch in brief, and the PBX in this solution is necessary for the high-end specific speech interface that has, and therefore needs very high cost, is not suitable in the major part, small enterprise.
Development along with network hardware equipment, further development is being arranged aspect traditional VOIP solution, IPT solution with Cisco is an example, shown in Fig. 3 Cisco-IPT, the hardware device that needs: router (Router), switch (Switch), call center (Call Manager) (speech payloads is distributed extension to IP phone), IP phone (IP Telephone), computer (PC), netting twine.The computer expert crosses netting twine and is connected on the IP phone, and IP phone is connected on the switch by netting twine, and Call Manager is connected on the switch by netting twine, and switch is connected on the router by netting twine.Calling out company B with first company is example, the call data transmission course is shown in Fig. 4 Cisco-IPT process, when the employee of first company calls out the employee of company B, dial prefix number (the same in prefix number and the VOIP of the Cisco solution earlier, also need enterprise network management person's predefined), the audio call packet sends to switch, switch sends to CallManager, send to the router of first company after the CallManager deal with data by switch, the router of first company finds the router of company B by addressing.This solution is to have replaced traditional telephone exchange with Call Manager, increased soft end phone, the routing iinformation in the router of first, the company B that but prerequisite is needs to be conversed is mutually done unified planning, so router intercommunication with it that planning is outer, want just to accomplish that the enterprise that free charge is conversed mutually must be the relation of general headquarters and branch, and the cost of Call Manager and IP phone is quite expensive, and general medium-sized and small enterprises can't afford at all.
Summary of the invention
The present invention is directed to the proposition of above problem, and develop a kind of method that realizes the VOIP of enterprise immediate.Can in the existing hardware environment of enterprise, utilize the immediate method of this VOIP to carry out call communication.The technological means that the present invention adopts is as follows:
A kind of method that realizes the VOIP of enterprise immediate is characterized in that comprising the steps:
A) initializing set: distribute extension for the voice interface card of server, distribute extension, user name is bound with extension to the user; Set digital equivalent in targeted customer's addressing address, described addressing address is made up of targeted customer's name, separator and destination address three parts, and wherein targeted customer's name and destination address are separated by separator;
B) the user's off-hook that makes a call, at first dial the extension of self server voice interface card, the user that makes a call this moment triggers dialing signal, be processed into the analog signal that contains dialing information and be sent to the user's that makes a call PBX by telephone set, user's the PBX of making a call can send call request to the client server that makes a call according to the internal port correspondence table, and the user's that makes a call this moment server response receives request simultaneously and connects; The user that makes a call then then dials the extension of oneself, and the user's that makes a call telephone set is processed into the server that the analog signal that contains dialing information sends to the user that makes a call;
C) voice interface card that makes a call on the client server receives analog signal, and the client server that makes a call is decoded into digital signal to the analog signal of obtaining that contains dialing information; Server can convert this digital signal to corresponding packet, the dialing information of server by unpacking and can obtaining, and then obtain this user's specifying information; The user that makes a call then dials the addressing address digit that is equivalent to the targeted customer that has configured, and the user's that makes a call telephone set is processed into the server that the analog signal that contains dialing information sends to the user that makes a call; User's the server of making a call this moment is decoded into digital signal to the analog signal that contains dialing information obtained; The user that makes a call that the client server that makes a call obtains dials the addressing address dialing information that is equivalent to the targeted customer, and changes it into dial the corresponding targeted customer of numeral with the user that makes a call addressing address date bag;
D) make a call user's the destination address of server parses separator back, it is targeted customer's server address, if this targeted customer's server address is legal and existence, the user's that makes a call server is just issued the voice request packet targeted customer's server address of separator back by network addressing;
When e) targeted customer's server receives the voice request packet, can unpack and resolve the preceding targeted customer's name of separator, this moment, targeted customer's server can judge whether this targeted customer's name exists, if exist, targeted customer's server can extract corresponding extension number;
F) and to targeted customer's the client request of making a call send call request by targeted customer's PBX to targeted customer's extension number simultaneously;
G) this moment the targeted customer client end response, targeted customer's branch power ringing simultaneously, user's second can receive request by client and set up session or mention own extension set and receive and ask to set up session.
Dialing among the described step b is push-button dialing or phonetic dialing.
Separator among the described step b is meant the symbol that is used for distinguishing user name and destination address, i.e. separator and user name and destination address symbol dissimilar or inequality; Described destination address indicates the position of targeted customer's server, can be IP address, domain name or host name.
The present invention can oneself define the numeral of oneself liking and be equivalent to targeted customer's addressing address, and has significantly reduced the length of dialing, makes the user convenient; It is free that enterprise can accomplish that landline telephone is dialed landline telephone by the present invention, saves fund to enterprise; Traditional VOIP is a big server cluster, and all user profile are all stored on server cluster, and the present invention is distributed to each enterprise or third-party operator to server, has reduced the bearing capacity of server, has also reduced the cost of server; Utilize the present invention, can realize video conference, fax in the same way, and be free fully.
The present invention compares and has the following advantages with traditional telephone system, Skype, the VOIP of Cisco solution, Cisco's IPT solution:
Wherein: the field in the form is explained as follows:
Always-and minute: the representative enterprise of conversation mutually is general headquarters and branch.
Non-total-minute: the representative enterprise of conversation mutually is separate enterprise.
Gu Gu-: the expression landline telephone is dialed landline telephone.
Gu-soft: the expression landline telephone is dialed soft phone.
Soft-soft: the expression soft phone is dialed soft phone.
Gu soft-: the expression soft phone is dialed landline telephone.
Low: the expression cost is very cheap.
High: expression costs an arm and a leg.
0: the expression expense is 0, i.e. cost free.
*: expression can't realize.
Figure BDA0000035642340000041
Description of drawings
Fig. 1 is the system configuration schematic diagram of prior art Cisco-voip;
Fig. 2 is the call data transmission course schematic diagram of system shown in Figure 1;
Fig. 3 is the system configuration schematic diagram of prior art Cisco-IPT;
Fig. 4 is the call data transmission course schematic diagram of system shown in Figure 3;
Fig. 5 is the call flow diagram among the method for the invention embodiment;
Fig. 6 is the system configuration schematic diagram among the method for the invention embodiment;
Fig. 7 is the data transmission procedure schematic diagram among the method for the invention embodiment.
Embodiment
Hardware environment is made of telephone exchange (PBX), server (Server), switch (PBX), router (Router), voice interface card, computer (PC), telephone set (Phone), netting twine and telephone wire.The hardware connection state is shown in the deployment of Fig. 6 enterprise: the computer expert crosses netting twine and links on the switch, voice interface card is connected on the server, server links to each other with telephone wire with PBX by the interface of voice interface card, telephone set is connected on the PBX by telephone wire, server is connected on the switch by netting twine, and switch is connected on the router by netting twine.
The service end major function: (this technology is a technology commonly used in the prior art, does not therefore do too much description here for leading subscriber, the VoP that receives voice interface card, forwarding VoP.)。The client major function: set on the landline telephone digital equivalent in the spcial character be separator the addressing communication modes, dial, reply, hang up.
User's second of calling out B company with user's first of A company is that example is described the method for the invention below, and A, B company need be by above-mentioned hardware environment configuration.Here destination address adopts the domain name of company as destination address; The domain name of supposing A company is: example-A.com (also can adopt server address as destination address, for example: 192.168.1.10), the user that the network manager of A company distributes to user's first is called: jia, then his addressing address is: jia@example-A.com (separator symbolization @ here, when being provided with usually, can adopt can pound out except that the letter and number key come on the QWERTY keyboard character as :~! @# $ % ^﹠amp; * ()? |, { } " ' /+" "<〉:; Deng); The domain name of B company is: the user that example-B.com, the network manager of B company distribute to user's second is called: yi, then his addressing address is: yi@example-B.com.
Wherein: telephone set (Phone): user's dialing information and the sound by the receiver collection are changed into analog signal and transmit; The analog signal that receives is changed into sound and play by receiver.Telephone exchange (PBX): distribute extension; Receive (forwarding) analog signal or digital signal; Voice interface card: receive and Analog signals or digital signal.Server (Server): receiving digital signals, and digital signal broken into packet; Handle packet; Processing data packets is become digital signal and transmission.Switch (Switch): receive and transmit packet.Router (Router): packet is sent in the network by Route Selection.
Telephone exchange (PBX): be the dial exchange that computer continues by the program control of working out in advance, the full name stored program controlled telephone exch.SPC telephone exchange is made up of hardware and software: hardware comprises speech channel part, control section and input.Software comprises program part and data division.The present invention does not have special requirement to telephone exchange, and general telephone exchange can both satisfy.Server (Server): in the local area network (LAN), the computer that a kind of runs administrative software conducts interviews to network or Internet resources (disc driver, printer etc.) with control, and can provide resource that it is operated just as work station for the computer on network.The present invention needs voice interface card to link to each other with server, and the kind of interface of voice interface card is various, gets final product so server only needs the interface that matches with voice interface card.Such as: if voice interface card is a pci interface, so just need the integrated pci card groove of server; If voice interface card is the PCI-E interface, server needs integrated PCI-E draw-in groove so; If voice interface card is a USB interface, server needs the integration USB interface (this technology also is technology commonly used in the prior art, does not therefore do too much description here so.)。Switch (Switch): the aggregate of traffic bogey, switching stage, control and signaling equipment and other functional units on the network node.The exchange function subscriber's line, telecommunication circuit and (or) other functional units that will interconnect couple together according to the request of unique user.The present invention does not require switch, and all switches can both satisfy.Router (Router): connect the equipment of each local area network (LAN), wide area network in the internet, it can be selected and the setting route automatically according to the situation of channel, with optimal path, sends the equipment of signal in proper order by front and back.The present invention does not require switch, and all routers can both satisfy.Voice interface card: contain integrated circuit board or equipment that speech interface is used for Analog signals or digital signal.
After the hardware connection finished, PBX can detect the equipment in the connection, can set the extension number of phone, voice interface card by the PBX management.
A) initializing set:
The PBX of A company sets: the extension that network manager elder generation distributes to the voice interface card of server by the PBX management is: 100; The extension of distributing to user's first is: 601.
The PBX of B company sets: the extension that network manager elder generation distributes to the voice interface card of server by the PBX management is: 200; The extension of distributing to user's first is: 801.
The service end initializing set:
A company service end is set: the network manager adds user's first and is in the company service end: jia also binds its extension (being telephone number information) and is: 601.
B company service end is set: the network manager adds user's second and is in the company service end: yi also binds its extension and is: 801.
The client initializing set:
A corporate user first is used user name by client: jia lands the A corporate server, (this numeral is self-defined numeral by setting numeral 1 in the client software, figure place is fixed by user oneself) be equivalent to: yi@example-B.com (the addressing address of B corporate user second), this setting can be synchronized to server simultaneously, and server can be stored this setting.The user just can realize calling specific as follows after initializing set was finished:
B) user's first off-hook, at first dial the extension of A corporate server voice interface card: 100, this moment, the user first triggered dialing (can push-button dialing or phonetic dialing) signal, be processed into by telephone set that (telephone set is different, the mode of handling is also different, encode such as having plenty of by DTMF, have plenty of by FSK and encode ...) contain the analog signal of dialing information and be sent to the PBX of A company, the PBX of A company can send call request to target terminal according to the internal port correspondence table, and this moment, the server response reception request simultaneously of A company connected; User's first is then dialled the extension 601 of oneself then, and the telephone set of user's first is processed into the server that the analog signal that contains dialing information sends to A company.
C) voice interface card on the A corporate server receives analog signal, and the A corporate server is decoded into digital signal to the analog signal of obtaining that contains dialing information by DTMF or FSK or other code encoding/decoding modes (code encoding/decoding mode depends on the setting of telephone set and PBX); Server can convert corresponding packet to changing digital signal, the dialing information [601] that server can obtain by unpacking and then obtain this user's specifying information.User's first is then dialled the numeral 1 that has configured, and the telephone set of user's first is processed into the server (this sentence belongs to step b, and for the ease of understanding, feature is in steps d) that the analog signal that contains dialing information sends to A company; This moment, the server of A company was decoded into digital signal (this sentence belongs to step c, and for the ease of understanding, feature is in steps d) to the analog signal of obtaining that contains dialing information by DTMF or FSK or other code encoding/decoding modes; Server can convert this digital signal to corresponding packet, the dialing information [1] that server can obtain by unpacking, and to convert addressing address of equal value to by the numeral 1 that service end configures be the packet of yi@example-B.com.
D) server address of the server parses character @ back of A company: example-B.com, if this server address is legal and existence, the server of A company is just issued the server address of character @ back: example-B.com (server address of B company) to the voice request packet by DNS (name server) addressing.
When e) service end of B company receives the voice request packet, can unpack and resolve the preceding user name of character: yi, this moment, B company service end can judge whether this user name yi exists, if exist, the service end of B company can be extracted corresponding extension number: 801;
F) and to the client request of making a call of user's second send call request by PBX to 801 simultaneously.
G) this moment user second client end response, 801 fens power ringings simultaneously, user's second can receive request by client set up session, also can mention own extension set and receive and ask to set up session.It is the example explanation that the above example of (this calling procedure is shown in Fig. 5 enterprise call flow process) (speech data in the transmission in the hardware shown in Fig. 7 enterprise hardware transmission flow) is called out landline telephone (or PC) with landline telephone, simultaneously also can save step b, directly call out landline telephone (or PC) with PC; Make a call with PC, do not need to receive analog signal, directly enter steps d, in the steps d 1 do not extract from packet at this moment, but server is by obtaining the numeral [1] of user input, and being converted into the spcial character by service end is the packet of decollator (yi@example-B.com); Following step is constant.
The above; only be the preferable embodiment of the present invention; but protection scope of the present invention is not limited thereto; anyly be familiar with those skilled in the art in the technical scope that the present invention discloses; be equal to replacement or change according to technical scheme of the present invention and inventive concept thereof, all should be encompassed within protection scope of the present invention.

Claims (3)

1. a method that realizes the VOIP of enterprise immediate is characterized in that comprising the steps:
A) initializing set: distribute extension for the voice interface card of server, distribute extension, user name is bound with extension to the user; Set digital equivalent in targeted customer's addressing address, described addressing address is made up of targeted customer's name, separator and destination address three parts, and wherein targeted customer's name and destination address are separated by separator;
B) the user's off-hook that makes a call, at first dial the extension of self server voice interface card, the user that makes a call this moment triggers dialing signal, be processed into the analog signal that contains dialing information and be sent to the user's that makes a call PBX by telephone set, user's the PBX of making a call can send call request to the client server that makes a call according to the internal port correspondence table, and the user's that makes a call this moment server response receives request simultaneously and connects; The user that makes a call then then dials the extension of oneself, and the user's that makes a call telephone set is processed into the server that the analog signal that contains dialing information sends to the user that makes a call;
C) voice interface card that makes a call on the client server receives analog signal, and the client server that makes a call is decoded into digital signal to the analog signal of obtaining that contains dialing information; Server can convert this digital signal to corresponding packet, the dialing information of server by unpacking and can obtaining, and then obtain this user's specifying information; The user that makes a call then dials the addressing address digit that is equivalent to the targeted customer that has configured, and the user's that makes a call telephone set is processed into the server that the analog signal that contains dialing information sends to the user that makes a call; User's the server of making a call this moment is decoded into digital signal to the analog signal that contains dialing information obtained; The user that makes a call that the client server that makes a call obtains dials the addressing address dialing information that is equivalent to the targeted customer, and changes it into dial the corresponding targeted customer of numeral with the user that makes a call addressing address date bag;
D) make a call user's the destination address of server parses separator back, it is targeted customer's server address, if this targeted customer's server address is legal and existence, the user's that makes a call server is just issued the voice request packet targeted customer's server address of separator back by network addressing;
When e) targeted customer's server receives the voice request packet, can unpack and resolve the preceding targeted customer's name of separator, this moment, targeted customer's server can judge whether this targeted customer's name exists, if exist, targeted customer's server can extract corresponding extension number;
F) and to targeted customer's the client request of making a call send call request by targeted customer's PBX to targeted customer's extension number simultaneously;
G) this moment the targeted customer client end response, targeted customer's branch power ringing simultaneously, user's second can receive request by client and set up session or mention own extension set and receive and ask to set up session.
2. a kind of method that realizes the VOIP of enterprise immediate according to claim 1 is characterized in that the dialing among the described step b is push-button dialing or phonetic dialing.
3. a kind of method that realizes the VOIP of enterprise immediate according to claim 1, it is characterized in that the separator among the described step b is meant the symbol that is used for distinguishing user name and destination address, i.e. separator and user name and destination address symbol dissimilar or inequality; Described destination address indicates the position of targeted customer's server, can be IP address, domain name or host name.
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Cited By (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN102232291A (en) * 2011-05-25 2011-11-02 华为技术有限公司 Method, system and device for communication
CN103179553A (en) * 2011-12-21 2013-06-26 华为终端有限公司 Method for changing subsidiary phone name, subsidiary phone, base and communication system
CN103685791A (en) * 2012-09-06 2014-03-26 黄能富 Communication method and system thereof
CN103813036A (en) * 2014-02-21 2014-05-21 广东绿瘦健康信息咨询有限公司 Communication connection allocation method and system thereof
CN105791610A (en) * 2016-04-18 2016-07-20 Ubiix有限公司 Enterprise instant voice communication method, device and applied electronic equipment
CN106161454A (en) * 2016-07-25 2016-11-23 大连天亿软件有限公司 A kind of immediate method of VOIP
CN107005618A (en) * 2014-11-26 2017-08-01 微软技术许可有限责任公司 PBX calls are controlled via client application

Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN1390031A (en) * 2001-05-26 2003-01-08 三星电子株式会社 Route-selecting service method in network agreement voice bussiness system
US6721790B1 (en) * 2000-03-09 2004-04-13 Avinta Communications, Inc User settable unified workstation identification system
CN1795643A (en) * 2003-03-12 2006-06-28 个人软件公司 Extension of a local area phone system to a wide area network with handoff
CN1822625A (en) * 2005-12-07 2006-08-23 北京佳讯飞鸿电气有限责任公司 Call extension system and call processing method
CN101292499A (en) * 2005-08-24 2008-10-22 高通股份有限公司 Wireless voip/vip roaming to access point of different network type

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6721790B1 (en) * 2000-03-09 2004-04-13 Avinta Communications, Inc User settable unified workstation identification system
CN1390031A (en) * 2001-05-26 2003-01-08 三星电子株式会社 Route-selecting service method in network agreement voice bussiness system
CN1795643A (en) * 2003-03-12 2006-06-28 个人软件公司 Extension of a local area phone system to a wide area network with handoff
CN101292499A (en) * 2005-08-24 2008-10-22 高通股份有限公司 Wireless voip/vip roaming to access point of different network type
CN1822625A (en) * 2005-12-07 2006-08-23 北京佳讯飞鸿电气有限责任公司 Call extension system and call processing method

Cited By (11)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN102232291A (en) * 2011-05-25 2011-11-02 华为技术有限公司 Method, system and device for communication
CN103179553A (en) * 2011-12-21 2013-06-26 华为终端有限公司 Method for changing subsidiary phone name, subsidiary phone, base and communication system
CN103179553B (en) * 2011-12-21 2016-09-07 华为终端有限公司 Method, handset, pedestal and the communication system of handset title change
CN103685791A (en) * 2012-09-06 2014-03-26 黄能富 Communication method and system thereof
CN103685791B (en) * 2012-09-06 2016-03-02 黄能富 Communication method and system thereof
CN103813036A (en) * 2014-02-21 2014-05-21 广东绿瘦健康信息咨询有限公司 Communication connection allocation method and system thereof
CN103813036B (en) * 2014-02-21 2015-01-14 广东绿瘦健康信息咨询有限公司 Communication connection allocation method and system thereof
CN107005618A (en) * 2014-11-26 2017-08-01 微软技术许可有限责任公司 PBX calls are controlled via client application
CN105791610A (en) * 2016-04-18 2016-07-20 Ubiix有限公司 Enterprise instant voice communication method, device and applied electronic equipment
CN106161454A (en) * 2016-07-25 2016-11-23 大连天亿软件有限公司 A kind of immediate method of VOIP
CN106161454B (en) * 2016-07-25 2019-05-10 大连天亿软件有限公司 A kind of immediate method of VOIP

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