[VoIP] Cnet tandem stacking

Steph Kerman stfkerman at jps.net
Thu Mar 22 11:50:39 CST 2007


	



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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
>>
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>>
>>
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>>
>>     
>
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