[VoIP] Seattle Museum Switches

Duncan Smith duncan.b.smith at gmail.com
Mon Jun 4 00:21:47 CDT 2007


[ Crossposted to the VoIP list and the CO/XBAR list; please don't
  crosspost replies unless you have a good reason. :) ]

On Sun, Jun 03, 2007 at 09:33:37PM -0400, Steph Kerman wrote:
> Do you have listings of the source code and/or configuration data?
> Are you modifying the load image or what are you doing to get it
> working?

Right now we have a listing of most of the CBLM (Common Base Level
Monitor), which is the main call-processing loop that dispatches
program control.  It's essentially cooperative multitasking, where
each task is required by contract to take 5ms or less per go-around.

> It would also be good if you recorded the acoustic call completion
> sounds adjacent to the equipment.  When I visited the equipment I
> was surprised at how much the acoustic sounds overwhelmed what I
> heard in the handset.  Especially true of the motor and shaft noise
> around the Panel office.  It completely changed my perception
> relative to what I was accustomed to hearing in the handset from the
> days when I had service from or access to lines on working Panel,
> #1XB and #5XB offices.  IMO, it's instructive to be able to hear the
> sounds in the handset only, with no acoustic noise present, hear the
> acoustic only, and hear them both together with an appropriate
> balance between the two so that neither one masks the other.

My favorite idea for making recordings is to put the handset audio on
the left channel, and the office audio on the right channel.  Then the
synchronization is nearly perfect, and the listener can decide how to
mix it.

On my to-do list is writing a program to record such audio.  I have
this apparently disgusting piece of hardware in my laptop, called a
"winmodem".  It functions under Linux as a sound card with a "mute"
control that actually is a hookswitch control.  It's basically an FXO,
for which I have much more use than a modem.

> At such time as you have an * working at the museum, it might be
> interesting if you hosted remote ATAs which got dial tone directly
> off the museum switches without any active role by the *.  "Nailed
> up" connections in other words.  This would allow people to make
> calls as though connected directly to the switches regardless of
> where they were to hear what call completion sounds like.  Nothing
> would be masked by the * for calls initiated this way.

I think it would be reasonably easy to do this.  I think the best way
to connect it is to use a T1 concentrator, and (so far as I know) it
is possible to connect different channels of a single concentrator to
different switches.

What's a good single-T1 line interface card to use with Asterisk?
Digium's Wildcard TE120P looks good, but I'd say that it's a tad
expensive.  (I don't know what the budget is, but I don't think it
includes $600.  Sad face.)

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