[VoIP] Real-time dialling and tandem stacking

[Jayson Smith]

Hi,
Now that'd be cool! And I bet I've just found a use for the currently unused
0XX and 1XX office codes! These could be used somehow for routing to
interoffice trunks, rather than complete call termination. I'm sure most
people have 'NXXXXXX' in the appropriate places in their dial plans, thus
restriccting them to calling CNET office codes from 200-999, and leaving
codes 000-199 inaccessible. Of course anybody could modify these if they
really wanted to play with these routing codes.
Here's how I would picture this working. In this example, I have a Step
switch, which I don't really have, and I am trying to call Doug Alderdice's
busy signal via an interoffice trunk. So I pick up a Step line, then dial
some access code to get an outbound CNET trunk. I dial '366' into this
trunk, and it then picks up an interoffice trunk to Doug's switch. Now at
this point, I would need to know when it was safe to continue dialing. Then,
I dial '3214' and those digits are converted to SF, transmitted, then
converted to DP at Doug's end. Now I should be hearing a busy signal.
Jayson

----- Original Message ----- 
From: "ikjtel" <ikj1234i at yahoo.com>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Sunday, April 01, 2007 12:10 PM
Subject: [VoIP] Real-time dialling and tandem stacking


> In brief, here's what might be required to move
> forward on this...
>
> 1.  Establishment of direct interoffice trunk groups
> between offices.  From the perspective of the SxS
> office, this should be largely identical to how things
> would have been done in real offices in the old
> days...
>
> 2. For the outgoing leg of the trunk, the outgoing SxS
> selector levels would terminate on some sort of zaptel
> "FXS-like" instance within asterisk.  Asterisk would
> be set up to send SF signalling outward on these ports
> (essentially DC-to-SF conversion).  It's been a long
> time since I looked at this, but if I recall
> correctly, this is a standard Zaptel feature and would
> require no extra patching.  It's not immediately clear
> how the "S" lead signal would be generated back to the
> selector, but I'm sure someone has already solved that
> problem...
>
> 3. The asterisk routing would presumably be set up
> such that these calls would be specially tagged
> somehow to identify them as direct interoffice trunk
> calls rather than ordinary CNET user calls.  When
> siezed in the outgoing office, asterisk would
> automatically initiate a VOIP connection to the proper
> destination in the called office, based on the
> selector level that was selected.
>
> 4. Incoming from the VOIP world, the direct
> interoffice trunks would be mapped to zaptel FXO ports
> that would then terminate on incoming selector levels
> on the EM machine.  Presumably the EM office might be
> arranged not to return dial tone on these trunks.  The
> zaptel driver on this leg of the call would require my
> "2600/SF" patch in order to reconvert the pulses from
> SF to DC for application to the incoming selectors in
> the called office.  Multi-hop operation should not be
> a problem, since the SF patch is configured to notch
> the 2600 and not pass it forward to succeeding legs of
> the call...
>
> 5. When the caller initially connects to the first SxS
> office in the connection, it might be set up to simply
> give them access to a first selector which returns
> dial tone without any initially-dialled digits (i.e.,
> unlike in CNET today).  The caller would connect, hear
> the dial tone, and begin dialling using SF
> outpulsing...
>
> 6. Channel-bank FXO should be used throughout instead
> of X100P in order to provide a realistic listening
> experience...
>
> Max
>
>
>
>
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