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?