170 likes | 314 Views
Asterisk @ CMU. Everything you needed to know to connect your dialog system to the world (but were afraid to ask). 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
E N D
Asterisk @ CMU Everything you needed to know to connect your dialog system to the world (but were afraid to ask)
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!
Asterisk: The Optimal Solution Olympus-running Dialog Systems Internet
Asterisk • Fully open source • Fully compliant with open standards • H.263 / RFCxyz / ulaw / … [Ignore most of this] • SIP • Allows a variety of setups
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 …”
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
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 • …
Asterisk Lingo • Trunks • Regular phone lines • Right now we only have one • Zaptel drivers make it work
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
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?
Asterisk • Questions?