[VoIP] Cnet tandem stacking

[Jayson Smith]

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.
>
>
> --------------------------------------------------------------------------
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>
> 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
>
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