FreePBX Application Introduce
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FreePBX is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), an open source communication server.
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FreePBX Application Introduce
- 2. • FreePBX Could Works For Private Branch Exchange (PBX) • Requirement: Hyper-V 2012 R2 VM Linux OS: FreePBX Standard PBX Service: Asterisk FreePBX Application
- 3. • Automatic Dialing • Automatic Switching • Call Transfer • Call Back FreePBX Basic Function
- 4. DUS works depends on VOIP (Voice over Internet Protocol): which is a technologies for delivery of Voice communication Multimedia session over Internet Protocol network. How FreePBX Works
- 5. • The Steps and Principles in originating • VoIP telephone calls: • Digitization of the analog voice signal • Packetized for these digital information • Encode audio and video with audio codecs: u-law , a-law , G-711 video codecs: H.263 What is Voice over IP
- 6. Network Integration Cluster Dial Up Server Cluster PSTN VOIP Gateway Corp Net DUS 1DUS 2DUS 3 Computer
- 7. VOIP Common Protocol • Session Initiation Protocol (SIP) • Real-time Transport Protocol (RTP) • Inter-Asterisk eXchange (IAX) • Skype protocol • H.263
- 8. Session Initiation Protocol • SIP design elements similar to the HTTP request/response transaction model. • Each transaction consists of a client request that invokes a particular method or function on the server within one response. • SIP reuses most of the header fields, encoding rules and status codes of HTTP, providing a readable text-based format.
- 10. Method Description INVITE Session setup request or media negotiation. Used also to hold &retrieve calls ACK Acknowledgement for an INVITE transaction completion OPTIONS Used as a query for remote's status & capabilities BYE Terminating a session CANCEL Used to cancel an on-going transaction REGISTER Registers a user with a Proxy/Register SIP Request
- 11. SIP Response Codes 100 Trying-Request has been received by a proxy/gateway 180 Ringing-The called party received the INVITE request, the phone is ringing 181 Call is being forwarded 182 Queued-Invite has been received and will be processed in a queue 183 Session Progress-Used to convey report of incoming early-media 200 OK-successful transaction completion 302 Moved Temporarily-Forward call to a given contact SIP Response
- 12. 305 Use Proxy-Repeats a me call setup using a given proxy 400 Bad Request-General error 401 Unauthorized-The request requires user authentication 404 Not Found-The user does not exist at the specified domain 408 Request Time out 486 Busy here 5XX Server Failure 6XX Global failure SIP Response
- 14. Digits 1209Hz 1336Hz 1477Hz 697Hz 1 2 3 770Hz 4 5 6 852Hz 7 8 9 941Hz * 0 # DTMF Meaning Dual-tone multi-frequency signaling (DTMF) is an in-band telecom signaling system using the voice-frequency band over telephone lines. That uses a set of eight audio frequencies transmitted in pairs to represent signals by digits
- 15. • Syntax dtmfmode=inband , rfc2833 , info and auto • Inband: This send tones as inband audio within the voice stream. If the phone is set for RFC2833 and asterisk is set for inband then you may not hear anything. • rfc2833: This is another inband method, that sends DTMF tones separately as specially encoded RTP packets, distinct from audio packets. DTMF Mode For SIP Configuration
- 16. • Syntax dtmfmode=inband , rfc2833 , info and auto • info :This is an out-of-band method that sends the DTMF signals within SIP on a separate network connection from the media streams. • auto: Asterisk use rfc2833 for DTMF relay by default but will switch to audio DTMF tones such as µ-law or a-law. If the remote side does not indicate support of rfc2833. DTMF Mode For SIP Configuration
- 17. • peer: A SIP point to which Asterisk sends calls (a SIP provider). The peer authenticates at registration. • user: A SIP entity which places calls through Asterisk . Users authenticate to reach services with their context. • friend: Asterisk will create two objects, one peer and one user, with the same name. Asterisk SIP Type
- 18. Q&A
Editor's Notes
- specifically refer to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).
- . IP電話通過把語音訊號經過数碼處理、壓縮編碼打包、透過網路傳輸、然後解壓、把数碼訊號還原成聲音，讓通話對方聽到 H.323的子協定多且複雜性高，在許多技術上的問題受限，不容易針對新的應用作擴充功能��因此，IETF (Internet Engineering Task Force)分別在1999年8月提出MGCP（Media Gateway Control Protocol）協定與1999年3月的SIP（Session Initiation Protocol）新架構，試圖簡化H.323的複雜性
- RTP用來定義在因特網上傳輸音頻和視頻的標準包格式，在RFC 1889中予以定義。它由音頻和視頻傳輸工作組制訂，並於1996年頒佈。 RTP和RTCP連接緊密，RTP實施實際資料的傳輸，RTCP則就服務質量做出反饋
- 會話發起協議（SIP）是IETF建立VoIP通話連接的協定標準。SIP是一種應用層控制協議，用於和一個或多個參與者創建、修改和終止會話。SIP的結構與HTTP（客戶－伺服器協議）相似。客戶機發出請求，並發送給伺服器，伺服器處理這些請求後給客戶機回送一個響應。請求與響應形成一次交換（transaction）。 SIP是會話初始化協定的縮寫，是一種用於建立、修改和終止VOIP電話通話的IP電話信號協定。SIP由IETF 制訂，並作爲RFC 3261頒佈。 媒體閘道器控制協議（MGCP）是由思科和Telcordia提議的VoIP協議，它定義了呼叫控制單元（呼叫代理或媒體閘道器）與電話閘道器之間的通訊服務。MGCP屬於控制協議，允許中心控制台監測IP電話和閘道器事件，並通知它們發送內容至指定位址。在MGCP結構中，智慧呼叫控制置於閘道器外部並由呼叫控制單元（呼叫代理）來處理。同時呼叫控制單元互相保持同步，發送一致的命令給閘道器。
- 撥打方電話發送邀請 接收方電話發出並送回應答180–正在撥打 當接收方電話發出應答200–OK 撥打方電話回以確認應答，以告知對方 當通過Media Session RTP傳輸資料時，就可以通話了 當撥打方挂上電話後，會向接收方電話發出一個再見呼叫 撥打方電話會回以應答200–OK
- user: A SIP entity which places calls through Asterisk (A phone which can place calls only).