[VoIP] Cnet tandem stacking

Jayson Smith ratguy at bellsouth.net
Thu Mar 22 11:54:51 CST 2007


Hello,
I can't find any problem on this end, and tried it through another shell
account and got through. Try again.
Jayson.

----- Original Message ----- 
From: "Steph Kerman" <stfkerman at jps.net>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Cc: "Jayson Smith" <ratguy at bellsouth.net>
Sent: Thursday, March 22, 2007 1:50 PM
Subject: Re: [VoIP] Cnet tandem stacking


>
>
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>
>  The page cannot be displayed
> The page you are looking for is currently unavailable. The Web site
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>
>
> Jayson Smith wrote:
> > Should be fixed now. Sorry about that.
> >
> > ----- Original Message ----- 
> > From: "Steph Kerman" <stfkerman at jps.net>
> > To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> > Cc: "Jayson Smith" <ratguy at bellsouth.net>
> > Sent: Thursday, March 22, 2007 1:37 PM
> > Subject: Re: [VoIP] Cnet tandem stacking
> >
> >
> >
> >> Forbidden
> >> You don't have permission to access /stack.wav on this server.
> >>
> >>
>
>> -------------------------------------------------------------------------
-
> >>
> > ------
> >
> >> Apache Server at www.bluegrasspals.com Port 80
> >>
> >>
> >> Jayson Smith wrote:
> >> Hi all,
> >> There's now a file up on my website which demonstrates the tandem stack
I
> >> made. Here are the steps:
> >> 1. Pick up any Cnet outbound phone and dial +1 349-8688. You will get a
> >> dialtone from Lee's SXS.
> >> 2. Dial 9 from that, and you'll get a dialtone from Lee's Asterisk
system.
> >> 3. Dial 622-6245 in my case, to get back to my voicemail. Since the SXS
is
> >> an analog component, Lee can't drop out of the circuit once the call is
> >> connected, since his SXS is holding the call.
> >> 4. Leave a message, then hang up. I assume what's happening here is
that
> >> when Lee's Asterisk system sees the disconnect, it signals the SXS line
to
> >> hang up which it does, When the SXS sees the hung up line, it hangs up
the
> >> trunk back to the Asterisk system. Thus, the far end of the connection
> >>
> > gets
> >
> >> to hear the click when the near end hangs up, and also the end of the
SXS
> >> hanging up. The file where I demonstrate this is here.
> >> http://www.bluegrasspals.com/stack.wav
> >> Jayson
> >>
> >> ----- Original Message ----- 
> >> From: "Lee Spenadel" <lee at spenadel.com>
> >> To: "'Jayson Smith'" <ratguy at bellsouth.net>; "'Voice Over IP Tandem for
> >> Analog Switches'" <voip at ckts.info>
> >> Sent: Thursday, March 22, 2007 11:22 AM
> >> Subject: RE: [VoIP] Cnet tandem stacking
> >>
> >>
> >>   Jayson,
> >>
> >> I've noticed the problem that you mention about not being able to break
> >> dialtone on my SxS.  I'm not sure yet why that is happening.  Each tone
to
> >> pulse converter is functioning properly on the switch, so it would
appear
> >> that something in the zap channel is not passing DTMF at that point.
> >>
> >> I have dialed into and out of my switches until all circuits have been
> >>     tied
> >>   up (testing my all circuits busy intercept) and this is where I get
the
> >> problem.  I curious if anyone has an idea on why this is happening.
> >>
> >> Lee
> >>
> >> -----Original Message-----
> >> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf
Of
> >> Jayson Smith
> >> Sent: Thursday, March 22, 2007 10:07 AM
> >> To: Voice Over IP Tandem for Analog Switches
> >> Subject: [VoIP] Cnet tandem stacking
> >>
> >> Hi,
> >> I just did an experiment where I dialed into Lee's SXS, then out to his
> >> Asterisk DISA, then back to my own voicemail where I left a message.
When
> >>     I
> >>   got the message, I could hear myself hanging up, then probably Lee's
SXS
> >> hanging up too. Neat! I did try going to SXS, then to DISA, then back
to
> >> SXS, since he does have two talk paths not reserved for coin phones,
but
> >> although I did get a second SXS dialtone, my DTMF digits weren't
getting
> >> through, so I couldn't get back over to Asterisk again! Talk about a
> >>     tandem
> >>   stack! Now if only other people had analog switches with trunks to
> >>     Asterisk
> >>   systems, we could really create some long calls, with some actual
analog
> >> routing!
> >> Jayson
> >>
> >> _______________________________________________
> >> VoIP mailing list
> >> VoIP at ckts.info
> >> http://lists.ckts.info/mailman/listinfo/voip
> >> Project Web Page: http://www.ckts.info/
> >>
> >>
> >>
> >> _______________________________________________
> >> VoIP mailing list
> >> VoIP at ckts.info
> >> http://lists.ckts.info/mailman/listinfo/voip
> >> Project Web Page: http://www.ckts.info/
> >>
> >>
> >>
> >>
> >
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> >
> >
>



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