Asterisk @ CMU

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Asterisk _at_ CMU

• Everything you needed to know to connect your
  dialog system to the world (but were afraid to
  ask)

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Dialog Systems

• Youve 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!

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A Picture Paints A Thousand Words
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Asterisk The Optimal Solution
Olympus-running Dialog Systems
Internet
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Asterisk

• Fully open source
• Fully compliant with open standards
• H.263 / RFCxyz / ulaw / Ignore most of this
• SIP
• Allows a variety of setups

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Asterisk Setup

• Its been done
• Asterisk_at_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

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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

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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 Lets Go!
• 400 Sublime

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Asterisk Lingo
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Asterisk Lingo

• Trunks
• Regular phone lines
• Right now we only have one
• Zaptel drivers make it work

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Asterisk Lingo
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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
• Youd need to implement SIP Transfer

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Asterisk Lingo
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Asterisk Lingo
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Asterisk Lingo
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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?

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Asterisk

• Questions?