FreePBX is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), an open source communication server.

1. Zack Chou
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.

9. SIP Operation

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

13. Asterisk Core Architecture

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