1/17

Asterisk @ CMU

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

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!

3/17

A Picture Paints A Thousand Words

4/17

Asterisk: The Optimal Solution

Internet

Olympus-running Dialog Systems

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

6/17

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 …”

7/17

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

8/17

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 …

9/17

Asterisk Lingo

10/17

Asterisk Lingo

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

11/17

Asterisk Lingo

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

13/17

Asterisk Lingo

14/17

Asterisk Lingo

15/17

Asterisk Lingo

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

Asterisk

Questions?