Asterisk (1)shared by voip.com.vn

1. 1/17
Asterisk @ CMU
Everything you needed to know
to connect your dialog system to
the world (but were afraid to ask)

2. 2/17
Dialog Systems
 You’ve got it up and running – it works great!
 On your PC
 Now you decide to let anyone call it up
 Current approach: Gentner boxes
 Dialog server connected directly to phone line
 Old technology, many issues with audio quality
 Huge inertia in setting up new systems
 Many, many experience people will tell you:
 THIS IS A BAD SOLUTION!

3. 3/17
A Picture Paints A Thousand Words

4. 4/17
Asterisk: The Optimal Solution
Internet
Olympus-running Dialog Systems

5. 5/17
Asterisk
 Fully open source
 Fully compliant with open standards
 H.263 / RFCxyz / ulaw / … [Ignore most of this]
 SIP
 Allows a variety of setups

6. 6/17
Asterisk Setup
 It’s been done
 Asterisk@Home
 Self-contained Linux + Asterisk installation
 FX100P phone interface with Zaptel drivers
 Aka Voicemodem
 Pretty sucky quality
 Luckily, Asterisk does some echo cancellation
 Virtual digital assistant
 “Press 1 for email, 2 for schedule, 3 for …”

7. 7/17
Asterisk with Olympus
 What you need to do
 Read up on SIP
 Tell me about it
 Implement a SIP-compliant interface for Olympus
 Manages session stuff
 New call
 Hang up
 Transfer call?
 Manages Audio I/O

8. 8/17
Asterisk Lingo
 Extensions
 For us, these are all SIP
 These are equivalent to phone lines in the real
world
 One SIP extension per dialog system
 200 – Roomline
 300 – Let’s Go!
 400 – Sublime
 …

9. 9/17
Asterisk Lingo

10. 10/17
Asterisk Lingo
 Trunks
 Regular phone lines
 Right now we only have one
 Zaptel drivers make it work

11. 11/17
Asterisk Lingo

12. 12/17
Asterisk Lingo
 Wiring it all together
 Asterisk knows about
 SIP extensions (Sublime, RoomLine, etc.)
 Physical phone lines (1 so far)
 We need to tell it how to connect these up
 Fixed rules
 Time dependent
 Digital receptionist
 User choice dependent
 Could make an Olympus-based Digital receptionist
 You’d need to implement SIP Transfer

13. 13/17
Asterisk Lingo

14. 14/17
Asterisk Lingo

15. 15/17
Asterisk Lingo

16. 16/17
Asterisk
 Things you should know
 Asterisk server is speeg2.speech.cs.cmu.edu
 SIP works only on UDP, port 5060
 Ask me (jsherwan at andrew) to create extensions
for your dialog systems
 Things we need to figure out
 Voice codecs (preferably use raw audio)
 16-bit linear codec (128kbps)
 Echo cancellation
 Alex / Alan, 24-port T1 Digium board, perhaps?

17. 17/17
Asterisk
 Questions?