[Asterisk-Users] Fw: Predictive dialer

[C. Maj]

On 8 Apr 2003, Karl Putland waxed:

> On Tue, 2003-04-08 at 10:49, Hemant Kumar wrote:
> > Hi Michiel,
> >
> > I went thru this site about the Real Time Spectrum Analyser and other
> > things.
> >
> > http://www.techmind.org/audio/
> >
> > But it is not open source, Do you know about some open source tools which
> > can be plugged in for the Frequency Analyses.
> >
>
>
> http://www.speech.kth.se/snack/
>
> Might fit in with what I have planned.

I have used the Tcl snack for a custom * voice mail project, and it works
great, with an extremely simple API.  I didn't use any of the frequency
analysis stuff, tho.

However, I am working on a predictive dialer, and right now it's in Tcl.
It doesn't use sample.call, but the manager interface.  One side effect
being that I'm also writing a bunch of Tcl wrappers for the manager
interface, so that's probably a good thing.

During the course of this work, over the past month or two on and off, I
have had to make a few modifications to *, mostly to spit out more info
in the manager functions.  But I also did major changes to chan_agent,
hacking in a display mechanism for ADSI phones.  At first I was trying
to do everything on the ADSI phones, and that didn't exactly work out,
due to 5-7 second delays in display on the screen phone (during which
time the person you called is going "hello, hello, click" since the
phone is in data mode instead of voice.)

So, I adopted a two-line approach.  You log in as an agent with both a
regular analog phone and an ADSI phone.  Then you keep your ear on the
analog phone, since that's what you talk on, and the ADSI phone is used
just for the screen display of the person's name, number, and address
that you are talking to.  It sort of works, but there's some echo
problems from delays in my code, and once it in a while it just locks
up and dies.  I am therefore now working on scrapping the two-line in
one-piece-of-code approach, in favor of separating the display mechanism
out into something else more generic.  This way you could plug in a web
browser or even a SIP phone for the display.

Everything is backed up by a PostgreSQL database, with some functions
like "get_next_number_to_autodial" and a table to log results of the
call.  It isn't really "predictive" yet, since I need to add some more
manager commands for calculating the length of the call.  Basically, it
just waits until there is an available agent before it starts dialing.

But it is a start, and I am interested in sharing the code when it
becomes a little more stable and less like spaghetti.  One thing I'm
really pumped about is the 0 second handover from the time a person
answers their phone until the agent hears them.  I already dropped a
couple of jaws from people saying, "Those machines cost $70,000 used and
they don't even transfer that fast!"  And then I add, "Yeah, it does
calling cards, voice mail, internet phone calls, and cooks eggs sunny
side up, too."  I digres...

* rulez

--Chris


-- 

Chris Maj <cmaj_hat_freedomcorpse_hot_info>
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