TW200908692A - System extending conventional telephone switch to connect network telephone - Google Patents

Abstract

A system that extends a conventional telephone switch to connect network telephones is provided and includes a conventional telephone switch, an IVR unit, a SIP program, a number of network telephones and a SIP proxy server. When an incoming call intends to connect to one of the network telephones, the call passes through the conventional telephone switch to the IVR unit. Afterwards, the IVR unit connects the call to one of the network telephones following the SIP protocol through the SIP program and a network under the control of the SIP proxy server.

Description

200908692 九、發明說明： 【發明所屬之技術領域】 本發明有關於-種傳統交換機擴大連接網路電話的系 統，尤其是指傳統交換翻合互動式語音反應(IVR，200908692 IX. Description of the Invention: [Technical Field of the Invention] The present invention relates to a system for expanding a connection to a network telephone by a conventional switch, and more particularly to a conventional exchange-combined interactive voice response (IVR,

InteractiveInteractive

Voice Resp㈣e)及網路電話軌缺(sip，碰龜 Protocol)而擴大連接網路電話的系統。 【先前技術】 ;利用網路打電話其t聰為便宜，目此—般麵網路電話 應^亥日益普及。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 洁系統改裝摘路電_統，_阻礙了網路電話的普及進 度0 傳統電話交換機若線路太多，則十分昂貴，以台北科技大 二交換機為例說明，台北科技大學的電話交換機具有 \華電信公司所提供的電話外線，並具有1200條内線分 2 ’攻種電話交換機的售價約NT$_萬，價格十分昂貴。若 技大學翻僅具有條内線分機的電話交換機，、其售 職0萬，所不足的900條内線分機均使用便宜的網 路電話糸統，可以大量節省成本。 換機ί f餅_贼顧的敎τ，均將傳統交 網路電話，使成本降低，並_路電話普及，就 疋本發明的主題。 200908692 【發明内容】 的在提供-輯統交換機 擴大連接網路電話 本發明的目的在提 的系統，包含： 具有多條外線電話 ，並具有多條内線Voice Resp (4) e) and Internet phone track shortage (sip, turtle protocol) to expand the system to connect to the Internet phone. [Prior technology]; using the Internet to call its t-constrained is cheap, this is the case - the general face of the Internet phone should be increasingly popular. However, in fact, because the traditional telephones are laid out in various places, it is very convenient for customers to use them. Therefore, customers and telecom operators do not have to actively modify the traditional electric cleaning system to pick up roads and electricity. _ hinder the popularity of VoIP. 0 Tradition If there are too many lines on the telephone exchange, it is very expensive. Take the Taipei Science and Technology sophomore switch as an example. The telephone exchange of the Taipei University of Science and Technology has the telephone line provided by Hua Telecom, and has 1200 internal lines divided into 2 'seed telephone exchanges. The price is about NT$_10,000 and the price is very expensive. If a university has only a telephone exchange with an extension, it sells for 10,000, and less than 900 extensions use cheap network telephones, which can save a lot of money. The change of the machine f f _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ 200908692 SUMMARY OF THE INVENTION The present invention provides a system for the purpose of the present invention, comprising: a plurality of external telephones having multiple internal lines

SIP程式’依照網路電話通訊協定而設計，附掛於 一傳統交換機， 分機； 語音總機上； 多個網路電話及一 SIP代理伺服器 程式以網際網路互相連接；The SIP program is designed according to the VoIP protocol, attached to a traditional switch, extension; voice switchboard; multiple Internet phones and a SIP proxy server are connected to each other via the Internet;

與語音總機、SIP 當外線電話欲與多個網路電話其中之一連接，其經過 =統交換機後轉人語音顧，_語音總機刺其非接内線 刀，後’由5吾音總機藉著SIP程式經網際網路受训代理飼 服器的控_與多侧路電話其中之—依照網路電話通訊 協定而連接。 【實施方式】 SIP簡介 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（Resp0nse)兩種。請求 200908692 是由客戶端（Client)發送至SIP代理飼服器（p_se㈣ 之sn>訊息，並表達客戶端的目的；回應為sip代理饲服器發 送至客戶端之SIP訊息，用以回覆客戶端之請求。 SIP定義了六種請求方法，包括mviTE、CANCEL BYE、ACK、REGISTER 與 OPTIONS，如表丨所示。 __SIP之六種基本請求 請求方法 說 明 INVITE -------- 建立-個新的舰會談（mediasessiQn)，或改 變當則會談之媒體特性（re_INviTE);訊息主 體通节會伴隨INVITE出現，以描述此invjte 欲建立（或改變）之媒體特性。 CANCEL 向SIP代理伺服器（UAS)取消尚未完成建立 程序之會談（未收到最終回應），此時SIP代 理伺服器可能處於收尋或響鈐之狀態。 BYE 結束已成功建立之會談（已收到2xx最終回 應）’發出此請求者可為INVITE請求之發起 端或是接收端。 — ACK 客戶端（UAC ’ INVITE請求之發起端）收到 SIP代理伺服器發出之最終回應（Final Response)後，向sip代理伺服器做確認之請 求。 — REGISTER SIP用戶端（user agent)向SIP代理伺服器發 ---- 200908692With the voice switchboard, SIP, when the outside line phone wants to connect with one of the multiple network phones, it passes through the switch and then transfers the voice to the voice. The voice switchboard spurs it to be connected to the line cutter, and then passes by the 5 voice switchboard. The SIP program is controlled by the Internet Training Agent's feeder _ and the multi-way telephone is connected according to the VoIP protocol. [Embodiment] Introduction to SIP The SIP establishes a call connection message (Message) is the most basic unit of SIP. It can be divided into two types: request (Request) and response (Resp0nse). The request 200908692 is sent by the client (Client) to the SIP proxy server (p_se(4) sn> message, and expresses the purpose of the client; the response is the SIP message sent by the sip proxy server to the client to reply to the client. Requests SIP defines six request methods, including mviTE, CANCEL BYE, ACK, REGISTER, and OPTIONS, as shown in the table. __SIP's six basic request request methods description INVITE -------- Create - new The ship talks (mediasessiQn), or change the media characteristics of the talks (re_INviTE); the message body will accompany the INVITE to describe the media characteristics that this invjte wants to establish (or change). CANCEL to SIP proxy server (UAS) Cancel the negotiation that the program has not been completed (the final response is not received), at this time, the SIP proxy server may be in the state of receiving or ringing. BYE ends the successfully established meeting (received 2xx final response) 'issued this The requester can be the originator or the receiver of the INVITE request. — The ACK client (the originator of the UAC 'INVITE request) receives the most from the SIP proxy server. After the response (Final Response), do confirm the request to the proxy sip -. REGISTER SIP client (user agent) sent to the SIP proxy server ---- 200 908 692

OPTIONSOPTIONS

出之註冊請求，SIP 代理伺服器收到此請求將 訊痒口。 查詢對方的支援能力 SIP回應訊息為SIP代理飼服器回覆客戶端請求之訊息 如表2所示。 __類別 狀態碼 範圍 回應型態 ---_____ 說 明 100 〜199 (lxx) Informational SIP代理伺服器已收到請求，且該 睛求已被處理；但該請求尚未被 接受。 200-299 (2xx) Success SIP代理伺服器接受客戶端送來 之請求。 300-399 (3xx) Redirection 請求訊息需被重新導向至另一個 伺服端’而重新導向之伺服端之 URL將被表達於『Contact』標頭 之棚位。 400〜499 (4xx) Client Error 請求因客戶端之錯誤而無法處 理’比如說訊息未認證、媒體型 怨不被支援或查無此人...等專。 客戶端可依照回應訊息中之指示 200908692 產生新的請求訊,— 500-599 Server Error 請求訊息目 (5xx) 錯誤而無法處理，但客戶端可將 凊求訊息傳送至其它伺服器進行 處理。 600-699 Global Error 請求訊息因整 (6xx) 無法處理，而請求訊息不可傳送 至其他伺機器或重試。 SIP通訊實例 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代理飼服器（pr〇xy Servei>)， 皆使用真實IP位址。 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 14〇 124 4〇 η。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（DomainName)為咖.voipedutw。 下面說明圖1的流程: 200908692Upon request for registration, the SIP proxy server will receive this request and will scream. Querying the support ability of the other party The SIP response message is the message that the SIP proxy server responds to the client request as shown in Table 2. __Category Status Code Range Response Type ---_____ Description 100 to 199 (lxx) The Informational SIP Proxy Server has received the request and the request has been processed; however, the request has not been accepted. 200-299 (2xx) Success The SIP proxy server accepts requests from clients. 300-399 (3xx) Redirection request message needs to be redirected to another server' and the URL of the redirected server will be expressed in the "Contact" header. 400~499 (4xx) Client Error request cannot be processed due to client error 'For example, the message is not authenticated, the media type is not supported or the person is not found... The client can generate a new request message according to the instruction in the response message 200908692. The 500-599 Server Error request message (5xx) error cannot be processed, but the client can transmit the request message to other servers for processing. 600-699 Global Error The request message cannot be processed due to the whole (6xx), and the request message cannot be transmitted to another server or retry. SIP Communication Example This section describes the message exchange for the complete SIP communication process. The example given in this section is a successful SIP call flow (ie containing 2xx responses), and both the UAC and the Receiver (UAS) and the SIP Proxy (pr〇xy Servei>) use real IP bits. site. 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 its SIP-URI is sip:[email protected], where 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 14〇 124 4〇 η. Both the sender and the receiver have successfully registered with the SIP proxy server. The domain name (DomainName) of the SIP proxy server is coffee. voipedutw. The flow of Figure 1 is explained below: 200908692

Ml:用戶Hsing想與用戶Hsf建立SIP連線，用戶Hsing由 位址140.124.43.145發出INVITE請求透過SIp代1伺: 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。 M2: Sff代理祠服器（ntut.v〇ip edu tw)轉發取乂瓜請求鈴 戶 Hsf。 、口 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)。 M4: SIP代理飼服器（ntut.voip.edu.tw)轉發「本地端響鈴 之訊息至用戶Hsing，其回覆路徑依照標頭之記錄， 回覆至 IP 位址 140.124.43.145。 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。 M6:SIP代理祠服器（ntut v〇ip edu.tw)轉發「本地端已接聽」 之5凡息至用戶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。 M8: SIP代理伺服器（ntut v〇jp edu.tw)轉發ACK請求至用戶 Hsing。 M9:用戶Hsf主動結束通話，並送出BYE請求至SIP代理伺 服斋（ntut.voip.edu.tw )。 M10: SIP代理伺服器（ntut v〇ip e£ju.tw)轉發BYE請求至用 200908692 戶 Hsing。 M11.用戶Hsing收到bye請求，並回覆γ2〇〇 〇κ」至sip代 理飼服器（ntut.voip.edu.tw)。 SIP代理伺服器（ntut.voip_edu.tw)轉發「200OK」至用 户 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 outl〜outN ’並具有300條内線分機extl〜ext3〇〇。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。 若外線電話意圖與内線分機extl〜ext3〇〇其中之一連接， δ吾音總機2則以一種互動式語音反應(IVR，Ml: The user Hsing wants to establish a SIP connection with the user Hsf. The user Hsing sends an INVITE request by the address 140.124.43.145 to the user Hsf via the SIp proxy: (ntut.voip.edu.tw), and the message is indicated in the message. Its RTP address is 140.124.43.145: 49170. M2: The Sff proxy server (ntut.v〇ip edu tw) forwards the request to the user Hsf. Port M3: User Hsf replies to the message "local ringing" to the SIp proxy server (ntut.voip.edu.tw). M4: The SIP proxy server (ntut.voip.edu.tw) forwards the message from the local ring to the user Hsing. The reply path is replied to the IP address 140.124.43.145 according to the record of the header. M5: User Hsf Reply to the message "Local Received" to the SIP proxy server (ntut.voip.edu.tw) and indicate that the RTP connection address is 140.124.40.11:3456. M6: The SIP proxy server (ntut v〇ip edu.tw) forwards the "local end answered" to the user Hsing. 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: The SIP proxy server (ntut v〇jp edu.tw) forwards the ACK request to the user Hsing. M9: The user Hsf actively ends the call and sends a BYE request to the SIP proxy server (ntut.voip.edu.tw). M10: The SIP proxy server (ntut v〇ip e£ju.tw) forwards the BYE request to 200908692 Hsing. M11. User Hsing receives the bye request and replies γ2〇〇 〇κ” to the sip agent feeder (ntut.voip.edu.tw). The SIP proxy server (ntut.voip_edu.tw) forwards "200OK" to the user Hsf. Please refer to FIG. 2' as a system diagram for expanding the connection of a conventional switch to a network telephone according to the present invention. The conventional switch 1 has a plurality of external telephones outl~outN' provided by Chunghwa Telecom, and has 300 extensions extl to ext3. The traditional switch 1 is attached to a voice switchboard. 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. If the external call is intended to be connected to one of the extensions extl to ext3, the δ Myspeaker 2 is an interactive voice response (IVR,

Interactive VoiceInteractive Voice

Response)的方式引導外線電話撥内線分機號碼，隨即將外線 電話引導至300條内線分機exti〜ext3〇〇其中之一。 右外線電話意圖與内線分機ext3〇i〜ext900其中之一連 接，本發明則將内線分機ext3〇l〜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，Sessi〇nResponse) guides the outside line to dial the extension number, and then the outside line is directed to one of the 300 extensions exti~ext3〇〇. The right outside line phone is intended to be connected to one of the extension lines ext3〇i~ext900. The present invention arranges the extension lines ext3〇l~ext900 into the type of the network telephone' interactive voice response in the voice switchboard 2 (ivr, Interactive) Voice Response) based on the above-mentioned VoIP protocol (SIp, Sessi〇n

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 200908692 路電話ext301〜ext900就是本發明將傳統交換機擴充的部分’ 其與300條内線分機extl〜ext300 —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outl〜outN的來電，這 樣就專於擴充了傳統交換機的容量。 圖2中的閘道器(Gateway) 5是一種傳統裝置，若外界的 行動電話意圖與内線分機exti~ext3〇〇其中之一連接，則通過 閘道器5、傳統交換機1、語音總機2與内線分機extl〜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、傳統交換機卜 曰總機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ext3〇1〜ext9〇〇其中之一連接。 内線分機extl〜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f話Gutl〜GutN其中之—而打出。 内線分機extl〜ext300若欲與網路電話ext3〇1〜ext9〇〇其中 之一連接’則可通過傳統交換機卜閘道器5進人網際網路3， ^後透過SIP代理飼服器4的控制而與網路電話ext3〇i〜ex_ 其中之一連接。 網路電話ext3G1〜ext_若欲與内線分機秦饮麵連 則^際網路3受SIP代糊服器4的控制，經由間道 =、傳統交換機卜語音總機2而與内線分機饮心湖 網路電話e)ct3〇l 3受SIP代理词服器 〜ext900若欲與外界連接，則經網際網路 4的控制’經由閘道器5而與外界連接。 12 200908692 語音總機2加上SIP程式2〗就是本發明的重點，可以讓 外線電話⑽〗〜outN (—般市内電話或行動電話）通過傳統交 換機卜語音總機2、SIP程式21進入網際網路3，秋後透過 SIP代理伺服器4的控制而與網路電話加綱〜加其中之 一連接。 下逃申請專利範圍，不受限於上述 本發明的範圍僅受限於 的特例。 【圖式簡單說明】 圖1為SIP通訊流程範例圖。 圖2為本發明將傳統交換機擴大連接網路電話的系統圖。 【主要元件符號說明】 1 傳統交換機 2 語音總機 3 網際網路 4 SIP代理伺服器 5 閘道器 瓤 13The SIP program 21 is designed and the voice switchboard 2 is connected to the Internet 3, so that the voice switchboard 2 is controlled by the SIP program 21 via the Internet Protocol 3 by the SIP proxy server (Proxy Server) 4. The network phones ext301 to ext900 interact in accordance with the above-mentioned Internet Telephony Protocol (SIp). The network 200908692 road telephone ext301~ext900 is the part of the invention that expands the traditional switch', and has the extension number with the 300 internal extensions extl~ext300, and can receive the call of the external telephone 2 outl~outN under the control of the voice switchboard 2, This is designed to expand the capacity of traditional switches. 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 exti~ext3, the gateway 5, the conventional switch 1, and the voice switch 2 are connected. One of the extensions extl to ext300 is connected. If the external mobile phone is intended to be connected to one of the Internet phones ext301 to ext900, then the gateway 3, the traditional switch switchboard 2, the SIP program 21 enter the Internet 3, and then pass the control of the sip proxy server 4. And connected to one of the Internet phone ext3〇1~ext9〇〇. If the extensions extl to ext300 are to be connected to the outside world, they are connected to the external line f via Gutl~GutN according to the conventional procedure. If the internal extension extl~ext300 wants to connect with one of the Internet phones ext3〇1~ext9〇〇, then it can enter the Internet 3 through the traditional switch gateway 5, and then through the SIP proxy feeder 4 Control is connected to one of the Internet phones ext3〇i~ex_. Internet phone ext3G1 ~ ext_ If you want to connect with the internal extension Qin drink face, then the network 3 is controlled by the SIP generation paste device 4, via the intermediation =, the traditional switch, the voice switchboard 2 and the internal extension drink center lake The network telephone e) ct3〇l 3 is connected to the outside world via the gateway device 5 if it is to be connected to the outside world by the SIP proxy word server ~ext900. 12 200908692 The voice switchboard 2 plus the SIP program 2 is the focus of the present invention, allowing the outside line (10) 〖outN (the general city phone or mobile phone) to enter the Internet through the traditional switch, the voice switchboard 2, and the SIP program 21. 3. After the fall, it is connected to the VoIP add-on by adding the control of the SIP proxy server 4. The scope of the patent application for the escape is not limited to the specific examples in which the scope of the present invention is limited only. [Simple diagram of the diagram] Figure 1 is a sample diagram of the SIP communication process. 2 is a system diagram of the invention for expanding a traditional switch to connect to a network telephone. [Main component symbol description] 1 Traditional switch 2 Voice switchboard 3 Internet 4 SIP proxy server 5 Gateway 瓤 13

Claims (1)

200908692 、申請專利範圍： 1. 一種傳統交換機擴大連接網路電話的系統，包含： 分機 •-傳統交換機’具有多條外線電話，並具有多條内線 -語音總機，晴於傳統交換機上，料線電話(泛指 :般市話與行動餘)欲财條_分機其巾之—連接，其 經過傳統交賴後’都會轉人語音總顧受縣式語音反應 的控制而與多條内線分機其中之一連接； θ μ 一 SIP程式，依照網路電話通訊協定而 語音總機上； w、 多個網路電話及，代理伺服器，與語音總機、SIP 私式以網際網路互相連接； 當外線_欲與多個網路電話其中之一連接， 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 ^機號碼非傳統交換機之内線分機後，由語音總機观 網際網路受SIP代理伺服器的控制而與多‘路電 邊其中之-依照網路電話通訊協定而連接。 細第1項之舰交換機擴大連接網路電話的 若外續其介於外線電話與傳統交換機之間，· 圖與内線分機其中之—連接，則通過閘道器、 、冼又換機、浯音總機與内線分機其中之 話意圖與網路電·^豆中 j ，右外線電 紗中之—連接，騎·道H、傳统交換 ^吾音總機、依據SIP程式而進入網際網路，然後透過灿 代理伺服器的控制而與網路電話其中之一連接。 14 200908692 系統，月内專^:第2 =讀統交換機擴大連接網路電話的 話其中之一連接，則:二=機= 進入網際網路，然ςτρ& 哥、兄父換機、閘道％ 話其令之—連接。 \理舰器的控制而與網路電 《如申請專利範圍第2項 系統，網路電話若欲與内線分機盆中接網路電話的 路受孙代理伺服器的控制，經，驗網際網 音總機而與内線分機Μ °、專統父換機、語 年、纽日， 中之一連接，網路電話若欲鱼外線雷 蝴’則經網際網路受SIP代理伺 二： 益而與外線電話連接。 役U閘道200908692, the scope of application for patents: 1. A traditional switch to expand the connection to the network phone system, including: Extensions - - Traditional switch ' has multiple external lines, and has multiple internal line - voice switchboard, clear on the traditional switch, the line The telephone (general refers to: general city and action), the _ extension, its towel-connection, after the traditional reconciliation, the voice will be transferred to the county-style voice response control and multiple extensions. One connection; θ μ a SIP program, in accordance with the VoIP protocol, on the voice switchboard; w, multiple VoIP phones, proxy servers, and voice switchboards, SIP private networks connected to each other through the Internet; _To connect with one of the multiple Internet telephony, the traditional switch is transferred to the voice switchboard, and the voice switchboard is used to identify the extension number of the non-traditional switch. The voice switchboard is connected to the Internet by the SIP proxy. The control of the server is connected to the multi-channels, which are connected according to the VoIP protocol. If the ship switch of the first item is expanded to connect to the Internet phone, it is between the external line and the traditional switch. · The connection between the picture and the extension of the internal line is connected through the gateway, the switch, and the switch. The tone switchboard and the extension line are intended to be connected to the network power, the bean, the right outer wire, the ride, the H, the traditional switch, the voice switch, the SIP, and then enter the Internet. Connected to one of the Internet phones through the control of the proxy server. 14 200908692 System, monthly special ^: 2nd = read one switch to expand the connection network phone, one of them connected, then: two = machine = enter the Internet, then ς ρρ & brother, brother change, gateway% Let it be made - connected. \The control of the ship and the network power "such as the second system of the patent application scope, if the network telephone wants to connect with the internal telephone in the extension of the internal extension, the server is controlled by the server, and the network is checked. The tone switchboard is connected to one of the internal extensions 、 °, the exclusive father changer, the language year, the New York, and the Internet phone. If the Internet phone wants to fish outside the line, the butterfly is served by the SIP agent. An external telephone connection. U gate