TWI381712B - Traditional switches expand systems that connect to Internet telephony - Google Patents

Traditional switches expand systems that connect to Internet telephony Download PDF

Info

Publication number
TWI381712B
TWI381712B TW96129902A TW96129902A TWI381712B TW I381712 B TWI381712 B TW I381712B TW 96129902 A TW96129902 A TW 96129902A TW 96129902 A TW96129902 A TW 96129902A TW I381712 B TWI381712 B TW I381712B
Authority
TW
Taiwan
Prior art keywords
telephone
internet
sip
proxy server
traditional switch
Prior art date
Application number
TW96129902A
Other languages
Chinese (zh)
Other versions
TW200908692A (en
Inventor
Shaw Hwa Hwang
Shun Chieh Chang
Chi Jung Huang
Hsiao Chang Chang
Yen Chun Huang
Sheng Yang Tsai
Chun Chi Fan
Yu Chen Tu
Yu Che Wang
Yao Chang Chung
Pan Chane Liu
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Application filed filed Critical
Priority to TW96129902A priority Critical patent/TWI381712B/en
Publication of TW200908692A publication Critical patent/TW200908692A/en
Application granted granted Critical
Publication of TWI381712B publication Critical patent/TWI381712B/en

Links

Landscapes

  • Telephonic Communication Services (AREA)

Description

傳統交換機擴大連接網路電話的系統Traditional switches expand systems that connect to VoIP

本發明有關於一種傳統交換機擴大連接網路電話的系統，尤其是指傳統交換機結合互動式語音反應(IVR,Interactive Voice Response)及網路電話通訊協定(SIP,Session Initiation Protocol)而擴大連接網路電話的系統。The invention relates to a system for expanding a connection network telephone by a traditional switch, in particular to a traditional switch that integrates an interactive voice response (IVR) and a SIP (Session Initiation Protocol) to expand a connection network. Telephone system.

利用網路打電話其費用較為便宜，因此一般預期網路電話應該日益普及。但事實上因為傳統電話的鋪設密佈各處，客戶使用上已十分方便，所以客戶與電信業者均不必積極將傳統電話系統改裝成網路電話系統，因而阻礙了網路電話的普及進度。It is cheaper to make calls using the Internet, so it is generally expected that Internet telephony should become increasingly popular. However, in fact, because the traditional telephones are densely packed and the customers are very convenient to use, the customers and the telecom operators do not have to actively convert the traditional telephone system into a network telephone system, thus hindering the progress of the Internet telephone.

傳統電話交換機若線路太多，則十分昂貴，以台北科技大學的電話交換機為例說明，台北科技大學的電話交換機具有38條中華電信公司所提供的電話外線，並具有1200條內線分機，這種電話交換機的售價約NT$800萬，價格十分昂貴。若台北科技大學購買僅具有300條內線分機的電話交換機，其售價只有NT$40萬，所不足的900條內線分機均使用便宜的網路電話系統，可以大量節省成本。Traditional telephone exchanges are very expensive if there are too many lines. Take the telephone exchange of Taipei University of Science and Technology as an example. The telephone exchange of Taipei University of Science and Technology has 38 telephone lines provided by Chunghwa Telecom and has 1200 extensions. The price of the telephone exchange is about NT$8 million and the price is very expensive. If Taipei University of Technology purchases a telephone exchange with only 300 extensions, its price is only NT$400,000, and less than 900 extensions use cheap VoIP systems, which can save a lot of money.

在大家仍然習慣使用傳統交換機的情況下，如何將傳統交換機擴大連接網路電話，使成本降低，並使網路電話普及，就是本發明的主題。In the case where everyone is still accustomed to using a conventional switch, how to extend the connection of a conventional switch to a network telephone, reduce the cost, and popularize the network telephone is the subject of the present invention.

本發明的目的在提供一種傳統交換機擴大連接網路電話的系統，包含：一傳統交換機，具有多條外線電話，並具有多條內線分機；一語音總機，附掛於傳統交換機上，當外線電話欲與多條內線分機其中之一連接，其經過傳統交換機後，都會轉入語音總機接受互動式語音反應的控制而與多條內線分機其中之一連接；一SIP程式，依照網路電話通訊協定而設計，附掛於語音總機上；多個網路電話及一SIP代理伺服器，與語音總機、SIP程式以網際網路互相連接；當外線電話欲與多個網路電話其中之一連接，其經過傳統交換機後轉入語音總機，透過語音總機辨別其非接內線分機後，由語音總機藉著SIP程式經網際網路受SIP代理伺服器的控制而與多個網路電話其中之一依照網路電話通訊協定而連接。The object of the present invention is to provide a system for expanding a connection network telephone by a conventional switch, comprising: a conventional switch having a plurality of external telephones and having a plurality of extension lines; a voice switchboard attached to a conventional switch, and an external line telephone To connect with one of the multiple extensions, after passing through the traditional switch, it will be transferred to the voice switchboard to receive interactive voice response control and connected to one of the multiple extensions; a SIP program, in accordance with the VoIP protocol The design is attached to the voice switchboard; multiple network phones and a SIP proxy server are connected to the voice switchboard and the SIP program via the Internet; when the outside line phone is to be connected to one of the plurality of network phones, After passing through the traditional switch, it is transferred to the voice switchboard, and after the voice switchboard discriminates its non-connected extension, the voice switchboard is controlled by the SIP proxy server via the SIP program via the SIP proxy server and one of the plurality of network phones. Connected by VoIP protocol.

SIP簡介SIP Introduction

SIP建立通話連線的訊息(Message)為SIP最基本的單位，可分為請求(Request)與回應(Response)兩種。請求是由客戶端(Client)發送至SIP代理伺服器(Proxy Server)之SIP訊息，並表達客戶端的目的；回應為SIP代理伺服器發送至客戶端之SIP訊息，用以回覆客戶端之請求。The message that SIP establishes a call connection is the most basic unit of SIP, and can be divided into two types: request (Request) and response (Response). The request is a SIP message sent by the client to the SIP proxy server (Proxy Server) and expresses the purpose of the client; the response is a SIP message sent by the SIP proxy server to the client to reply to the client's request.

SIP定義了六種請求方法，包括INVITE、CANCEL、BYE、ACK、REGISTER與OPTIONS，如表1所示。SIP defines six request methods, including INVITE, CANCEL, BYE, ACK, REGISTER, and OPTIONS, as shown in Table 1.

SIP回應訊息為SIP代理伺服器回覆客戶端請求之訊息，如表2所示。The SIP response message is a message that the SIP proxy server replies to the client request, as shown in Table 2.

SIP通訊實例SIP communication instance

本節介紹完整SIP通訊流程的訊息交換。本節所舉的例子是一個成功的SIP通話流程(即含有2xx回應)，且發話端(UAC)與受話端(UAS)以及SIP代理伺服器(Proxy Server)，皆使用真實IP位址。This section describes the exchange of messages for the complete SIP communication process. The example in this section is a successful SIP call flow (that is, containing a 2xx response), and both the UAC and the Receiver (UAS) and the SIP Proxy Server use real IP addresses.

圖1描述一個完整的SIP通訊過程，包括SIP訊息的交換與RTP媒體封包的傳輸。在本例中，發話端的用戶帳號為hsing，其SIP－URI為sip：[email protected]，所在IP位址為140.124.43.145。受話端帳號為hsf，其SIP－URI為sip：[email protected]，所在IP位址為140.124.40.11。發話端與受話端都已向SIP代理伺服器註冊成功，SIP代理伺服器的網域名稱(Domain Name)為ntut.voip.edu.tw。Figure 1 depicts a complete SIP communication process, including the exchange of SIP messages and the transmission of RTP media packets. In this example, the user account of the sender is hsing, and the SIP-URI is sip:[email protected], and the IP address is 140.124.43.145. The receiving end account is hsf, and its SIP-URI is sip:[email protected], and the IP address is 140.124.40.11. Both the sender and the receiver have successfully registered with the SIP proxy server. The domain name of the SIP proxy server is ntut.voip.edu.tw.

下面說明圖1的流程：M1：用戶Hsing想與用戶Hsf建立SIP連線，用戶Hsing由IP位址140.124.43.145發出INVITE請求透過SIP代理伺服器(ntut.voip.edu.tw)轉發給用戶Hsf，並於訊息中註明其RTP位址為140.124.43.145：49170。The following describes the flow of Figure 1: M1: User Hsing wants to establish a SIP connection with the user Hsf, and the user Hsing sends an INVITE request by the IP address 140.124.43.145 to the user Hsf through the SIP proxy server (ntut.voip.edu.tw). And indicate in the message that its RTP address is 140.124.43.145:49170.

M2：SIP代理伺服器(ntut.voip.edu.tw)轉發INVITE請求給用戶Hsf。M2: The SIP proxy server (ntut.voip.edu.tw) forwards the INVITE request to the user Hsf.

M3：用戶Hsf回覆「本地端響鈴」之訊息至SIP代理伺服器(ntut.voip.edu.tw)。M3: The user Hsf replies to the "local ringing" message to the SIP proxy server (ntut.voip.edu.tw).

M4：SIP代理伺服器(ntut.voip.edu.tw)轉發「本地端響鈴」之訊息至用戶Hsing，其回覆路徑依照Via標頭之記錄，回覆至IP位址140.124.43.145。M4: The SIP proxy server (ntut.voip.edu.tw) forwards the message "local ringing" to the user Hsing. The reply path is replied to the IP address 140.124.43.145 according to the record of the Via header.

M5：用戶Hsf回覆「本地端已接聽」之訊息至SIP代理伺服器(ntut.voip.edu.tw)，並註明其RTP連線位址為140.124.40.11：3456。M5: User Hsf replies to the message "Local Received" to the SIP proxy server (ntut.voip.edu.tw) and indicates that its RTP connection address is 140.124.40.11:3456.

M6：SIP代理伺服器(ntut.voip.edu.tw)轉發「本地端已接聽」之訊息至用戶Hsing。M6: The SIP proxy server (ntut.voip.edu.tw) forwards the message "Local Received" to the user Hsing.

M7：用戶Hsing送出ACK請求至SIP代理伺服器(ntut.voip.edu.tw)，確認「本地端已接聽」之訊息。M7: User Hsing sends an ACK request to the SIP proxy server (ntut.voip.edu.tw) to confirm the message "Local has been answered".

M8：SIP代理伺服器(ntut.voip.edu.tw)轉發ACK請求至用戶Hsing。M8: The SIP proxy server (ntut.voip.edu.tw) forwards the ACK request to the user Hsing.

M9：用戶Hsf主動結束通話，並送出BYE請求至SIP代理伺服器(ntut.voip.edu.tw)。M9: User Hsf actively ends the call and sends a BYE request to the SIP proxy server (ntut.voip.edu.tw).

M10：SIP代理伺服器(ntut.voip.edu.tw)轉發BYE請求至用戶Hsing。M10: The SIP proxy server (ntut.voip.edu.tw) forwards the BYE request to the user Hsing.

M11：用戶Hsing收到BYE請求，並回覆「200 OK」至SIP代理伺服器(ntut.voip.edu.tw)。M11: User Hsing receives the BYE request and replies "200 OK" to the SIP proxy server (ntut.voip.edu.tw).

M12：SIP代理伺服器(ntut.voip.edu.tw)轉發「200 OK」至用戶Hsf。M12: The SIP proxy server (ntut.voip.edu.tw) forwards "200 OK" to the user Hsf.

請見圖2，為本發明將傳統交換機擴大連接網路電話的系統圖。傳統交換機1具有多條中華電信公司所提供的外線電話out1~outN，並具有300條內線分機ext1~ext300。傳統交換機1附掛一個語音總機2，當外線電話的信號進入傳統交換機1時，都會轉入語音總機2接受辨別、控制。Please refer to FIG. 2, which is a system diagram of the invention for expanding a traditional switch to connect to a network telephone. The traditional switch 1 has a plurality of external telephones out1~outN provided by Chunghwa Telecom, and has 300 internal extensions ext1~ext300. The traditional switch 1 is attached to a voice switchboard 2. When the signal of the external line enters the traditional switch 1, it will be transferred to the voice switchboard 2 for identification and control.

若外線電話意圖與內線分機ext1~ext300其中之一連接，語音總機2則以一種互動式語音反應(IVR,Interactive Voice Response)的方式引導外線電話撥內線分機號碼，隨即將外線電話引導至300條內線分機ext1~ext300其中之一。If the external telephone is intended to be connected to one of the extensions ext1~ext300, the voice switchboard 2 will use an interactive voice response (IVR) to guide the outside line to dial the extension number, and then the outside line will be directed to 300. One of the extensions ext1~ext300.

若外線電話意圖與內線分機ext301~ext900其中之一連接，本發明則將內線分機ext301~ext900安排成為網路電話的型態，在語音總機2的互動式語音反應(IVR,Interactive Voice Response)中依據上述的網路電話通訊協定(SIP,Session Initiation Protocol)而設計SIP程式21，並將語音總機2與網際網路3連接，於是語音總機2藉著SIP程式21經網際網路3受SIP代理伺服器(Proxy Server)4的控制而與眾多網路電話ext301~ext900依照上述的網路電話通訊協定(SIP)而互動。網路電話ext301~ext900就是本發明將傳統交換機擴充的部分，其與300條內線分機ext1~ext300一樣都具有分機號碼，可以接受語音總機2的控制而接收外線電話out1~outN的來電，這樣就等於擴充了傳統交換機的容量。If the external telephone is intended to be connected to one of the extensions ext301~ext900, the present invention arranges the extensions ext301~ext900 into the type of the network telephone, in the interactive voice response (IVR) of the voice switchboard 2 The SIP program 21 is designed according to the above-mentioned SIP Initiation Protocol (SIP), and the voice switchboard 2 is connected to the Internet 3, so that the voice switchboard 2 is SIP-mediated by the SIP program 21 via the Internet 3. The control of the server (Proxy Server) 4 interacts with a plurality of network telephones ext301~ext900 in accordance with the above-mentioned Internet Telephony Protocol (SIP). The network phone ext301~ext900 is the extension part of the traditional switch of the present invention, and has the same extension number as the 300 internal extensions ext1~ext300, and can receive the control of the voice switchboard 2 and receive the incoming calls of the outside line out1~outN, thus It is equivalent to expanding the capacity of the traditional switch.

圖2中的閘道器(Gateway)5是一種傳統裝置，若外界的行動電話意圖與內線分機ext1~ext300其中之一連接，則通過閘道器5、傳統交換機1、語音總機2與內線分機ext1~ext300其中之一連接。若外界的行動電話意圖與網路電話ext301~ext900其中之一連接，則通過閘道器5、傳統交換機1、語音總機2、SIP程式21進入網際網路3，然後透過SIP代理伺服器4的控制而與網路電話ext301~ext900其中之一連接。The gateway 5 in FIG. 2 is a conventional device. If the external mobile phone is intended to be connected to one of the extensions ext1 to ext300, pass through the gateway 5, the conventional switch 1, the voice switchboard 2, and the extension line. One of ext1~ext300 is connected. If the external mobile phone is intended to be connected to one of the network phones ext301~ext900, the gateway 3, the traditional switch 1, the voice switchboard 2, the SIP program 21 enter the Internet 3, and then pass through the SIP proxy server 4. Control and connect to one of the network phones ext301~ext900.

內線分機ext1~ext300若欲與外界連接，則按傳統程序，經由傳統交換機1連接外線電話out1~outN其中之一而打出。If the extension extension ext1~ext300 wants to connect with the outside world, it will be connected to one of the external telephones out1~outN via the traditional switch 1 according to the conventional procedure.

內線分機ext1~ext300若欲與網路電話ext301~ext900其中之一連接，則可通過傳統交換機1、閘道器5進入網際網路3，然後透過SIP代理伺服器4的控制而與網路電話ext301~ext900其中之一連接。If the extension extension ext1~ext300 is to be connected to one of the network phones ext301~ext900, it can enter the Internet 3 through the traditional switch 1, the gateway 5, and then communicate with the VoIP through the control of the SIP proxy server 4. One of ext301~ext900 is connected.

網路電話ext301~ext900若欲與內線分機ext1~ext300連接，則經網際網路3受SIP代理伺服器4的控制，經由閘道器5、傳統交換機1、語音總機2而與內線分機ext1~ext300連接。If the network phone ext301~ext900 is to be connected to the extension extension ext1~ext300, it will be controlled by the SIP proxy server 4 via the gateway 3, via the gateway 5, the traditional switch 1, the voice switchboard 2 and the extension extension ext1~ Ext300 connection.

網路電話ext301~ext900若欲與外界連接，則經網際網路3受SIP代理伺服器4的控制，經由閘道器5而與外界連接。If the Internet phone ext301~ext900 is to be connected to the outside world, it is connected to the outside world via the gateway 5 via the Internet Protocol 3 under the control of the SIP proxy server 4.

語音總機2加上SIP程式21就是本發明的重點，可以讓外線電話out1~outN(一般市內電話或行動電話)通過傳統交換機1、語音總機2、SIP程式21進入網際網路3，然後透過SIP代理伺服器4的控制而與網路電話ext301~ext900其中之一連接。The voice switchboard 2 plus the SIP program 21 is the focus of the present invention, and the outside line telephone out1~outN (general city telephone or mobile phone) can enter the Internet 3 through the traditional switch 1, the voice switchboard 2, and the SIP program 21, and then through The control of the SIP proxy server 4 is connected to one of the network phones ext301 to ext900.

本發明的範圍僅受限於下述申請專利範圍，不受限於上述的特例。The scope of the present invention is limited only by the scope of the following claims, and is not limited to the specific examples described above.

1．．．傳統交換機1. . . Traditional switch

2．．．語音總機2. . . Voice switchboard

3．．．網際網路3. . . Internet

4．．．SIP代理伺服器4. . . SIP proxy server

5．．．閘道器5. . . Gateway

圖1為SIP通訊流程範例圖。Figure 1 is an example of a SIP communication process.

圖2為本發明將傳統交換機擴大連接網路電話的系統圖。2 is a system diagram of the invention for expanding a traditional switch to connect to a network telephone.

1．．．傳統交換機1. . . Traditional switch

2．．．語音總機2. . . Voice switchboard

3．．．網際網路3. . . Internet

4．．．SIP代理伺服器4. . . SIP proxy server

5．．．閘道器5. . . Gateway

Claims (4)

一種傳統交換機擴大連接網路電話的系統，包含：一傳統交換機，具有多條外線電話，並具有多條內線分機；一語音總機，附掛於傳統交換機上，當外線電話(泛指一般市話與行動電話)欲與多條內線分機其中之一連接，其經過傳統交換機後，都會轉入語音總機接受互動式語音反應的控制而與多條內線分機其中之一連接；一SIP程式，依照網路電話通訊協定而設計，附掛於語音總機上；多個網路電話及一SIP代理伺服器，與語音總機、SIP程式以網際網路互相連接；當外線電話欲與多個網路電話其中之一連接，其經過傳統交換機後轉入語音總機，透過語音總機辨別其欲連接之分機號碼非傳統交換機之內線分機後，由語音總機藉著SIP程式經網際網路受SIP代理伺服器的控制而與多個網路電話其中之一依照網路電話通訊協定而連接。A traditional switch expands a system for connecting to a network telephone, comprising: a traditional switch having a plurality of external telephones and having a plurality of extensions; a voice switchboard attached to a conventional switch, and an external telephone (referred to as a general telephone) And the mobile phone) wants to connect with one of the multiple extensions, after passing through the traditional switch, it will be transferred to the voice switchboard to receive the interactive voice response control and connected with one of the multiple extensions; a SIP program, according to the network Designed by telephone telephone protocol, attached to the voice switchboard; multiple Internet phones and a SIP proxy server, connected to the voice switchboard and SIP program via the Internet; when the outside line wants to connect with multiple Internet phones One of the connections, after the traditional switch, is transferred to the voice switchboard, and the extension number of the extension to be connected is determined by the voice switchboard. After the extension of the extension is not the traditional switch, the voice switchboard is controlled by the SIP proxy server via the SIP program via the Internet. And one of the plurality of Internet phones is connected in accordance with the VoIP protocol. 如申請專利範圍第1項之傳統交換機擴大連接網路電話的系統，並含一閘道器，其介於外線電話與傳統交換機之間；若外線電話意圖與內線分機其中之一連接，則通過閘道器、傳統交換機、語音總機與內線分機其中之一連接；若外線電話意圖與網路電話其中之一連接，則通過閘道器、傳統交換機、語音總機、依據SIP程式而進入網際網路，然後透過SIP代理伺服器的控制而與網路電話其中之一連接。For example, the traditional switch of claim 1 expands the system for connecting to a network telephone and includes a gateway between the external telephone and the conventional switch; if the external telephone is intended to be connected to one of the internal extensions, The gateway, the traditional switch, the voice switchboard and one of the extensions are connected; if the outside line is intended to be connected to one of the network phones, the gateway is accessed through the gateway, the traditional switch, the voice switchboard, and the SIP program. And then connected to one of the Internet phones through the control of the SIP proxy server. 如申請專利範圍第2項之傳統交換機擴大連接網路電話的系統，內線分機若欲與一般市話或行動電話連接，則按傳統程序，經由傳統交換機連接外線其中之一而打出；內線分機若欲與網路電話其中之一連接，則通過傳統交換機、閘道器進入網際網路，然後透過SIP代理伺服器的控制而與網路電話其中之一連接。For example, if the traditional switch of the second application of the patent scope expands the system for connecting to the Internet telephone, if the extension of the extension is to be connected to a general local telephone or a mobile telephone, it is connected to one of the external lines via a conventional switch according to a conventional procedure; To connect to one of the Internet telephony, it enters the Internet through a traditional switch, gateway, and then connects to one of the Internet telephony through the control of the SIP proxy server. 如申請專利範圍第2項之傳統交換機擴大連接網路電話的系統，網路電話若欲與內線分機其中之一連接，則經網際網路受SIP代理伺服器的控制，經由閘道器、傳統交換機、語音總機而與內線分機其中之一連接；網路電話若欲與外線電話連接，則經網際網路受SIP代理伺服器的控制，經由閘道器而與外線電話連接。For example, the traditional switch of the second application patent scope expands the system for connecting to the Internet telephone. If the network telephone wants to connect with one of the extensions, the Internet is controlled by the SIP proxy server, via the gateway, and the conventional The switch and the voice switchboard are connected to one of the extension lines; if the network phone wants to connect with the external line, the Internet is connected to the external line via the gateway through the control of the SIP proxy server.
TW96129902A 2007-08-14 2007-08-14 Traditional switches expand systems that connect to Internet telephony TWI381712B (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
TW96129902A TWI381712B (en) 2007-08-14 2007-08-14 Traditional switches expand systems that connect to Internet telephony

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
TW96129902A TWI381712B (en) 2007-08-14 2007-08-14 Traditional switches expand systems that connect to Internet telephony

Publications (2)

Publication Number Publication Date
TW200908692A TW200908692A (en) 2009-02-16
TWI381712B true TWI381712B (en) 2013-01-01

Family

ID=44723742

Family Applications (1)

Application Number Title Priority Date Filing Date
TW96129902A TWI381712B (en) 2007-08-14 2007-08-14 Traditional switches expand systems that connect to Internet telephony

Country Status (1)

Country Link
TW (1) TWI381712B (en)

Families Citing this family (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
TWI404387B (en) * 2010-08-13 2013-08-01 Chunghwa Telecom Co Ltd Communication system and method for using session initiation protocol (sip) on a converted ip address
TWI404386B (en) * 2010-08-13 2013-08-01 Chunghwa Telecom Co Ltd Communication system and method for using multi-tiered registration session initiation protocol (sip)
TWI419572B (en) * 2010-12-31 2013-12-11 Univ Nat Taipei Technology Method of telephone switchboard for connecting directly the callback to the extension of the original caller

Citations (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20030202503A1 (en) * 2002-04-29 2003-10-30 Masanori Inui Telephone communication system
US20060098803A1 (en) * 2003-12-18 2006-05-11 Sbc Knowledge Ventures, L.P. Intelligently routing customer communications
US20060227957A1 (en) * 1999-04-01 2006-10-12 Dolan Robert A Methods and apparatus for providing expanded telecommunications service
US20070058787A1 (en) * 2000-08-15 2007-03-15 Pointone, Inc. Centralized Feature Platform In A Packetized Network
TW200714008A (en) * 2005-09-28 2007-04-01 Yin-Kai Huang Method for integrating network phone with conventional phone and the system thereof
US20070092073A1 (en) * 2005-10-03 2007-04-26 Verizon Data Services Inc. PBX call management
US20070172051A1 (en) * 2004-02-11 2007-07-26 Klaus Hoffmann Setting up a packet-oriented multimedia connection using an interactive voice response system

Patent Citations (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20060227957A1 (en) * 1999-04-01 2006-10-12 Dolan Robert A Methods and apparatus for providing expanded telecommunications service
US20070058787A1 (en) * 2000-08-15 2007-03-15 Pointone, Inc. Centralized Feature Platform In A Packetized Network
US20030202503A1 (en) * 2002-04-29 2003-10-30 Masanori Inui Telephone communication system
US20060098803A1 (en) * 2003-12-18 2006-05-11 Sbc Knowledge Ventures, L.P. Intelligently routing customer communications
US20070172051A1 (en) * 2004-02-11 2007-07-26 Klaus Hoffmann Setting up a packet-oriented multimedia connection using an interactive voice response system
TW200714008A (en) * 2005-09-28 2007-04-01 Yin-Kai Huang Method for integrating network phone with conventional phone and the system thereof
US20070092073A1 (en) * 2005-10-03 2007-04-26 Verizon Data Services Inc. PBX call management

Also Published As

Publication number Publication date
TW200908692A (en) 2009-02-16

Similar Documents

Publication Publication Date Title
US8447019B2 (en) Method and system for providing call screening in a packet-switched network
TWI401927B (en) Method and computer-readable medium for associating a telephone call with a dialog based on a computer protocol such as sip
JP5275908B2 (en) Communication system, session control management server, and session control method
EP1758360A2 (en) Proxy independent hunt group function in a packet based network
KR100602638B1 (en) The method for VoIP-UMS system access
CN101099366A (en) Facilitating early media in a communications system
US20060018267A1 (en) IP telephone system, ENUM server and method for performing telephone conference
US8681199B2 (en) Method of providing video-call service using general voice-call terminal and private branch exchange for performing the method
TWI381712B (en) Traditional switches expand systems that connect to Internet telephony
RU2374777C2 (en) Processing of initial multimedia data i
WO2007093116A1 (en) A method and system for realizing the simulating service and the access signaling adaptive entity
US8249238B2 (en) Dynamic key exchange for call forking scenarios
US9578180B2 (en) Communication network system, calling terminal and voice call establishing method thereof
JP4864999B2 (en) Caller information notification request service system, session control server, call control method, and program
WO2009036801A1 (en) Methods and arrangements for a telecommunications system
CN101132349A (en) Method for extended implementing IMS connected number indication business by SIP signaling
US20080137647A1 (en) VoIP terminal and method for providing multi-call service
KR20050043597A (en) Method of visitor confirming and communication service using internet
KR100475187B1 (en) key phone system for enable session initiation protocol and method for call setup
WO2013029389A1 (en) Method and device for realizing coloring ring back tone service in ctd call service
WO2019208433A1 (en) Rtp monitoring device and rtp monitoring method
CN110035051A (en) Voice service system and voice service method
TWI488464B (en) Method for dialing from internet extension to conventional extension
CN103905669B (en) A kind of method for interchanging data, system and MGC
KR100923642B1 (en) Method of video phone call using a bridge

Legal Events

Date Code Title Description
MM4A Annulment or lapse of patent due to non-payment of fees