From lee at spenadel.com  Wed Aug  1 06:23:36 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Wed, 1 Aug 2007 07:23:36 -0400
Subject: [VoIP] Correct Way To Connect X100P To Step By Step Exchange?
In-Reply-To: <023301c7d3f4$133d8bb0$5f4665da@compaq>
References: <023301c7d3f4$133d8bb0$5f4665da@compaq>
Message-ID: <01af01c7d42e$67b2e570$3718b050$@com>

The timing problem could be that the X100P is pulsing out too quickly for
the switch.  In my situation, the X100P began pulsing out before the Line
Finder had a chance to ascend to the proper level.  Therefore, only the last
digit or two were being decoded properly.  I corrected this by placing
several wait (w) options in my dial string.  An example would be:

exten => _${OFFICECODE}863X,4,Dial(ZAP/g2/P/www${EXTEN:4},30)

The www provides my switch enough time to ascend to the correct level and be
ready for the pulsing of digits.  

I don't know enough about the zaptel drivers to help you there, but there
are some very knowledgeable people in this group.

Lee


-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Brent Jones
Sent: Wednesday, August 01, 2007 12:26 AM
To: Voice Over IP Tandem for Analog Switches
Subject: [VoIP] Correct Way To Connect X100P To Step By Step Exchange?

Hi. I have got an Asterisk box up and running. Now I want to be able
to hook up my X100P to a Step by Step Pabx. 

I'll be connecting the FXO to an extension on the PABX.  
It is a step by step type that uses uniselectors to make the connections
between extensions.

But at first I am trying to get it working on a model Step by Step unit that
uses 
subcribers uniselectors connected to a group then final selector.

I have being trying to
connect to the  model step by step without success. I have loaded the
pulse dial patches needed. But what happens is when I dial, the
subscribers line uniselector seems to hunt and step to the pulses. 
Some of the last pulses seem to get through to the group selector and
you see it moving verticaly.

I have played around with the timings in zaptel.h but it doesn't seem
to improve things.

Can someone please tell me what sort of settings they found to work
for the timings. Also can someone please tell what settings they use
for zaptel.conf and zapta.conf files.  Even a diagram of how you wired
the FXO card to your PABX.

Also are X100P cards polarity sensitive with relationship to the line
voltages ie
does it matter which pin gets A or B leg .
Do I need to put a 'w' in the dialing string before the first digit to let
the subscriber
uniselector hunt.

Does the FXO pulse dialing patch return the loop back to the exchange during
dialing
between digits?

Any ideas welcome.

Thanks
Brent Jones
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/


From ikj1234i at yahoo.com  Wed Aug  1 07:36:49 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Wed, 1 Aug 2007 05:36:49 -0700 (PDT)
Subject: [VoIP] Correct Way To Connect X100P To Step By Step Exchange?
In-Reply-To: <023301c7d3f4$133d8bb0$5f4665da@compaq>
Message-ID: <354777.39044.qm@web51610.mail.re2.yahoo.com>


--- Brent Jones <gijones at clear.net.nz> wrote:

> pulse dial patches needed. But what happens is when
> I dial, the
> subscribers line uniselector seems to hunt and step
> to the pulses. 
> Some of the last pulses seem to get through to the
> group selector and
> you see it moving verticaly.

Sounds offhand as though Lee's suggestion to add some
dead time (via the 'w') command might help.  I do
vaguely seem to recall, however, that John N. and
others had reported a while back that 'w' wasn't
working anymore or at all or something - has 'w' been
fixed or what? 

> 
> I have played around with the timings in zaptel.h
> but it doesn't seem
> to improve things.

Can you post specifically what values you've used
and/or changed?

> 
> Can someone please tell me what sort of settings
> they found to work
> for the timings. Also can someone please tell what
> settings they use
> for zaptel.conf and zapta.conf files.  Even a
> diagram of how you wired
> the FXO card to your PABX.
> 
> Also are X100P cards polarity sensitive with
> relationship to the line voltages ie
> does it matter which pin gets A or B leg .

As far as I know, absolutely not.  It shouldn't make
any difference...

> Do I need to put a 'w' in the dialing string before
> the first digit to let the subscriber
> uniselector hunt.
> 
> Does the FXO pulse dialing patch return the loop
> back to the exchange during dialing
> between digits?

Hmm, not sure what you mean by this.  There may be a
slight difference in terminology ;-)  The X100P does
implement what I call a 'spike filter' or transient
filter; immediately upon receipt of any pulse (or even
an inductuve transient, as PDW has found) it will go
into a guarded state in which its audio ckts are cut
out of the loop and not reconnected for approx.
150-250 msec.  I haven't been able to determine if
this is detectable by way of any momentary change in
loop current; however, I wouldn't expect this to be of
any consequence for a ckt that uses only relays...

> 
> Any ideas welcome.
> 
> Thanks
> Brent Jones
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 



       
____________________________________________________________________________________
Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.
http://smallbusiness.yahoo.com/webhosting 

From jnovack at stromberg-carlson.org  Wed Aug  1 07:41:57 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Wed, 01 Aug 2007 08:41:57 -0400
Subject: [VoIP] Correct Way To Connect X100P To Step By Step Exchange?
In-Reply-To: <354777.39044.qm@web51610.mail.re2.yahoo.com>
References: <354777.39044.qm@web51610.mail.re2.yahoo.com>
Message-ID: <46B07F95.4010709@stromberg-carlson.org>

Later versions of zaptel seemed to have fixed that.
Russ Price implies that his patches fix it also, but I believe that 
zaptel after 1.2.12 or so fixed it as well.

JN


ikjtel wrote:
> --- Brent Jones <gijones at clear.net.nz> wrote:
>
>   
>> pulse dial patches needed. But what happens is when
>> I dial, the
>> subscribers line uniselector seems to hunt and step
>> to the pulses. 
>> Some of the last pulses seem to get through to the
>> group selector and
>> you see it moving verticaly.
>>     
>
> Sounds offhand as though Lee's suggestion to add some
> dead time (via the 'w') command might help.  I do
> vaguely seem to recall, however, that John N. and
> others had reported a while back that 'w' wasn't
> working anymore or at all or something - has 'w' been
> fixed or what? 
>
>   
>> I have played around with the timings in zaptel.h
>> but it doesn't seem
>> to improve things.
>>     
>
> Can you post specifically what values you've used
> and/or changed?
>
>   
>> Can someone please tell me what sort of settings
>> they found to work
>> for the timings. Also can someone please tell what
>> settings they use
>> for zaptel.conf and zapta.conf files.  Even a
>> diagram of how you wired
>> the FXO card to your PABX.
>>
>> Also are X100P cards polarity sensitive with
>> relationship to the line voltages ie
>> does it matter which pin gets A or B leg .
>>     
>
> As far as I know, absolutely not.  It shouldn't make
> any difference...
>
>   
>> Do I need to put a 'w' in the dialing string before
>> the first digit to let the subscriber
>> uniselector hunt.
>>
>> Does the FXO pulse dialing patch return the loop
>> back to the exchange during dialing
>> between digits?
>>     
>
> Hmm, not sure what you mean by this.  There may be a
> slight difference in terminology ;-)  The X100P does
> implement what I call a 'spike filter' or transient
> filter; immediately upon receipt of any pulse (or even
> an inductuve transient, as PDW has found) it will go
> into a guarded state in which its audio ckts are cut
> out of the loop and not reconnected for approx.
> 150-250 msec.  I haven't been able to determine if
> this is detectable by way of any momentary change in
> loop current; however, I wouldn't expect this to be of
> any consequence for a ckt that uses only relays...
>
>   
>> Any ideas welcome.
>>
>> Thanks
>> Brent Jones
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>     
>
>
>
>        
> ____________________________________________________________________________________
> Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.
> http://smallbusiness.yahoo.com/webhosting 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From john_reads_cnet_via_archives at covert.org  Wed Aug  1 11:19:43 2007
From: john_reads_cnet_via_archives at covert.org (John R. Covert)
Date: Wed, 1 Aug 2007 12:19:43 -0400 (EDT)
Subject: [VoIP] Diagrams for 500/2500/554/2575 Telephones
Message-ID: <20070801163254.31FCF56346@ns1.vyger.net>

Steph's request for the ITT Dials reminded me that I've been
needing to put these diagrams online.  Feel free to copy,
repost, and redistribute with or without attribution.

http://www6.covert.org/TelephoneDiagrams.pdf

Telephone Circuit Diagrams 
1. Western Electric 500 CW & 500 DW Tel Sets.
   U.S. Army ca. 1965 
2. Western Electric 2500D Tel Set
   Dial from Bell System Tech Journal, merged w/(1) above 
3. Stromberg-Carlson 2500DP/554TDB Telephone Wiring Diagram
   Stromberg-Carlson ca. 1982.  (two pages) 
4. ITT 2575 Type Modular Telephone Circuit
   ITT Corporation ca. 1982

/john

From jnovack at stromberg-carlson.org  Wed Aug  1 12:19:34 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Wed, 01 Aug 2007 13:19:34 -0400
Subject: [VoIP] Diagrams for 500/2500/554/2575 Telephones
In-Reply-To: <20070801163254.31FCF56346@ns1.vyger.net>
References: <20070801163254.31FCF56346@ns1.vyger.net>
Message-ID: <46B0C0A6.2050901@stromberg-carlson.org>

Don't forget the TCI technical library as a source both for set drawings 
and a central repository for drawings that one might have to submit.

John Novack


John R. Covert wrote:
> Steph's request for the ITT Dials reminded me that I've been
> needing to put these diagrams online.  Feel free to copy,
> repost, and redistribute with or without attribution.
>
> http://www6.covert.org/TelephoneDiagrams.pdf
>
> Telephone Circuit Diagrams 
> 1. Western Electric 500 CW & 500 DW Tel Sets.
>    U.S. Army ca. 1965 
> 2. Western Electric 2500D Tel Set
>    Dial from Bell System Tech Journal, merged w/(1) above 
> 3. Stromberg-Carlson 2500DP/554TDB Telephone Wiring Diagram
>    Stromberg-Carlson ca. 1982.  (two pages) 
> 4. ITT 2575 Type Modular Telephone Circuit
>    ITT Corporation ca. 1982
>
> /john
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From john_reads_cnet_via_archives at covert.org  Wed Aug  1 13:46:17 2007
From: john_reads_cnet_via_archives at covert.org (John R. Covert)
Date: Wed, 1 Aug 2007 14:46:17 -0400 (EDT)
Subject: [VoIP] Porting phone number, what will happen with the POTS line?
Message-ID: <20070801185013.1D28056A0A@ns1.vyger.net>

To make a long story short, I had a port away from Verizon
completed on March 9th, and we still have battery on the line
as of today, 1 August.

Back in Jan/Feb/Mar I ported two of my five Verizon numbers to
Vonage.  These had been the two numbers on one of my two ISDN
lines; I wanted to cut back to a single ISDN line and a single
POTS line, and I wanted the numbers from the disconnected ISDN
line to come into the Asterisk via my Vonage softphone.

You can't port ISDN numbers.  So I had to disconnect the ISDN
line, and immediately after the ISDN line was disconnected,
have Verizon install a new POTS line with one of the two ISDN
numbers.  (I decided to do this serially, and had only 30 days
to at least get both numbers moved to POTS, otherwise they would
go back into the pool, no longer reserved for me.)

Came the day of the POTS install, and nothing happened.  Verizon
informed me when I inquired that there had been no need for
anyone to come out to the house, that they had tested, and the
line was active in the spare NI on the side of the house.  Of
course, it wasn't.  A day later, they sent someone out and
figured out where the cross-connect was missing.  A couple
of days later, I figured the order was complete.

I initiated the port from the Vonage side on January 23, 2007.
They sent a request to the carrier on January 24, 2007.  Verizon
immediately replied "cannot be ported."  It turned out that
the order wasn't really completely closed.  On January 29, 2007,
Verizon finally approved the transfer (in the meantime I had to
beat on Vonage to get them to even SEND the request a second
time) and scheduled the port for February 7, 2007.

Keep in mind that when porting to Vonage, you will always
already have temporary service on another number until the
port goes into effect.  At the moment of the actual port,
that temporary number becomes a virtual number.  It can
either be kept permanently as a virtual number (for about
$5+fees per month) or go back into the pool in a couple of
days.

On the day of the port the first thing that happened was that
calls from within Vonage to the ported number started being
delivered to the Vonage line, and outgoing calls from that
line had the ported caller id.  Around 9am, calls from outside
Vonage also started going to Vonage.  But it took well into the
afternoon before all calls moved over, i.e. before the LNP database
was fully consistent for all accesses.  Outgoing calls were still
possible from the Verizon line for most of the day, but then the
line dropped to battery only.

Then I ordered another POTS install for the second number.  Even
though they had to do nothing but issue the order, I was given a
due date of February 19th.  Shortly after midnight on the morning
of 19 Feb, the POTS line was in and working.  But that wasn't good
enough for Verizon.  Oh, no.

All of the utilities except gas enter my house at the same point,
right next to the kitchen door.  There is utility power, a drop
from the cable company, a six-pair copper cable from Verizon,
and Verizon Business FiOS 20Mb/5Mb service all right next to each
other. The six pair cable terminates in a WeCo 116D3B-6 screw-box.
It was installed in the days before external NIs, and there were
two three-pair cables run from it to a little six pair terminal
block mounted in the basement.  Over the years, things changed.
At one point, I had all six pairs in use.  The POTS line from
the Acton C.O., a two-wire SOP (station off premises) from DEC's
No 1. ESS centrex in Maynard, a four-wire SOP from DEC's PBX in
Acton Nagog Woods, and, for a brief period after the installation
of a new PBX at DEC in Marlboro, for the purposes of testing, a
second 4-wire SOP.  Eventually some of this service was removed
and replaced with an ISDN lines and a second POTS line; then a
second ISDN line was ordered and the POTS number moved into one
of the SPIDs of the ISDN line.  And at some point two small
3"x3 1/2" J-112 NIs were installed.

Anyway, about 9am I happened to walk into the kitchen, and
found TWO techs outside the house in the process of removing
both NIs, disconnecting the fairly complicated cross-connects
going into and out of the basement, and about to screw some
sort of HUGE new two-line NID on the side of the house.

I managed to get them to stop, explaining that the line that
had just been installed did not require any premises work;
the existing NIs were just fine, and besides, they shouldn't
spend a bunch of money, since the line was going to be pulled
right back out within a week and ported to Vonage.  There is
only a $13.50 C.O. work installation fee in Massachusetts and
no charge for outside plant work, and I really didn't want to
waste their time and money.

The next port was ordered on Feb 22nd after Verizon assured me
that the order was complete, but it apparently wasn't complete
enough, because they bounced it again on the 23rd and finally
accepted it on the 28th, with a due date of March 9th.

The transfer process was pretty much the same, except that
for about two days after the Verizon POTS line was disconnected,
there were still occasions where calls from various places attempted
to go to Verizon and got a "the number you have called is not in
service" announcement.

And there is still battery but no dialtone on the circuit as
of right now.

/john

From lee at spenadel.com  Wed Aug  1 21:04:32 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Wed, 1 Aug 2007 22:04:32 -0400
Subject: [VoIP] Porting phone number,
	what will happen with the POTS line?
In-Reply-To: <20070801185013.1D28056A0A@ns1.vyger.net>
References: <20070801185013.1D28056A0A@ns1.vyger.net>
Message-ID: <027301c7d4a9$788b5ff0$69a21fd0$@com>

ISDN is still out there?  It's like a Hitchcock movie.......

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
John R. Covert
Sent: Wednesday, August 01, 2007 2:46 PM
To: CNET
Subject: [VoIP] Porting phone number, what will happen with the POTS line?

To make a long story short, I had a port away from Verizon
completed on March 9th, and we still have battery on the line
as of today, 1 August.

Back in Jan/Feb/Mar I ported two of my five Verizon numbers to
Vonage.  These had been the two numbers on one of my two ISDN
lines; I wanted to cut back to a single ISDN line and a single
POTS line, and I wanted the numbers from the disconnected ISDN
line to come into the Asterisk via my Vonage softphone.

You can't port ISDN numbers.  So I had to disconnect the ISDN
line, and immediately after the ISDN line was disconnected,
have Verizon install a new POTS line with one of the two ISDN
numbers.  (I decided to do this serially, and had only 30 days
to at least get both numbers moved to POTS, otherwise they would
go back into the pool, no longer reserved for me.)

Came the day of the POTS install, and nothing happened.  Verizon
informed me when I inquired that there had been no need for
anyone to come out to the house, that they had tested, and the
line was active in the spare NI on the side of the house.  Of
course, it wasn't.  A day later, they sent someone out and
figured out where the cross-connect was missing.  A couple
of days later, I figured the order was complete.

I initiated the port from the Vonage side on January 23, 2007.
They sent a request to the carrier on January 24, 2007.  Verizon
immediately replied "cannot be ported."  It turned out that
the order wasn't really completely closed.  On January 29, 2007,
Verizon finally approved the transfer (in the meantime I had to
beat on Vonage to get them to even SEND the request a second
time) and scheduled the port for February 7, 2007.

Keep in mind that when porting to Vonage, you will always
already have temporary service on another number until the
port goes into effect.  At the moment of the actual port,
that temporary number becomes a virtual number.  It can
either be kept permanently as a virtual number (for about
$5+fees per month) or go back into the pool in a couple of
days.

On the day of the port the first thing that happened was that
calls from within Vonage to the ported number started being
delivered to the Vonage line, and outgoing calls from that
line had the ported caller id.  Around 9am, calls from outside
Vonage also started going to Vonage.  But it took well into the
afternoon before all calls moved over, i.e. before the LNP database
was fully consistent for all accesses.  Outgoing calls were still
possible from the Verizon line for most of the day, but then the
line dropped to battery only.

Then I ordered another POTS install for the second number.  Even
though they had to do nothing but issue the order, I was given a
due date of February 19th.  Shortly after midnight on the morning
of 19 Feb, the POTS line was in and working.  But that wasn't good
enough for Verizon.  Oh, no.

All of the utilities except gas enter my house at the same point,
right next to the kitchen door.  There is utility power, a drop
from the cable company, a six-pair copper cable from Verizon,
and Verizon Business FiOS 20Mb/5Mb service all right next to each
other. The six pair cable terminates in a WeCo 116D3B-6 screw-box.
It was installed in the days before external NIs, and there were
two three-pair cables run from it to a little six pair terminal
block mounted in the basement.  Over the years, things changed.
At one point, I had all six pairs in use.  The POTS line from
the Acton C.O., a two-wire SOP (station off premises) from DEC's
No 1. ESS centrex in Maynard, a four-wire SOP from DEC's PBX in
Acton Nagog Woods, and, for a brief period after the installation
of a new PBX at DEC in Marlboro, for the purposes of testing, a
second 4-wire SOP.  Eventually some of this service was removed
and replaced with an ISDN lines and a second POTS line; then a
second ISDN line was ordered and the POTS number moved into one
of the SPIDs of the ISDN line.  And at some point two small
3"x3 1/2" J-112 NIs were installed.

Anyway, about 9am I happened to walk into the kitchen, and
found TWO techs outside the house in the process of removing
both NIs, disconnecting the fairly complicated cross-connects
going into and out of the basement, and about to screw some
sort of HUGE new two-line NID on the side of the house.

I managed to get them to stop, explaining that the line that
had just been installed did not require any premises work;
the existing NIs were just fine, and besides, they shouldn't
spend a bunch of money, since the line was going to be pulled
right back out within a week and ported to Vonage.  There is
only a $13.50 C.O. work installation fee in Massachusetts and
no charge for outside plant work, and I really didn't want to
waste their time and money.

The next port was ordered on Feb 22nd after Verizon assured me
that the order was complete, but it apparently wasn't complete
enough, because they bounced it again on the 23rd and finally
accepted it on the 28th, with a due date of March 9th.

The transfer process was pretty much the same, except that
for about two days after the Verizon POTS line was disconnected,
there were still occasions where calls from various places attempted
to go to Verizon and got a "the number you have called is not in
service" announcement.

And there is still battery but no dialtone on the circuit as
of right now.

/john
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/


From rdekema at gmail.com  Wed Aug  1 21:37:17 2007
From: rdekema at gmail.com (Rusty Dekema)
Date: Wed, 1 Aug 2007 22:37:17 -0400
Subject: [VoIP] Porting phone number,
	what will happen with the POTS line?
In-Reply-To: <027301c7d4a9$788b5ff0$69a21fd0$@com>
References: <20070801185013.1D28056A0A@ns1.vyger.net>
	<027301c7d4a9$788b5ff0$69a21fd0$@com>
Message-ID: <68171c120708011937r1bdfc78eja0998f21aa890056@mail.gmail.com>

On 8/1/07, Lee Spenadel <lee at spenadel.com> wrote:
> ISDN is still out there?  It's like a Hitchcock movie.......

It's still out there, but at least in southeast Michigan, AT&T seems
to be trying to discourage its use (for BRI, anyway) by increasing the
price. In 1999, I had an ISDN BRI from Ameritech. It cost
approximately $80/mo for unlimited local calls, and then $45/mo for
the 800 calls per month plan that I switched to once I bought a better
router that didn't hang up and redial thousands of times per month.

Earlier this year, I called AT&T (mostly on a whim, but had the price
still been ~$40/mo, I might have ordered one to screw around with for
a few months) and asked about pricing on residential BRI, and they
said they could roll one out for $90/mo and no included calls at all.
I declined to place the order.

Rusty

From duncan.b.smith at gmail.com  Wed Aug  1 22:59:08 2007
From: duncan.b.smith at gmail.com (Duncan Smith)
Date: Wed, 1 Aug 2007 20:59:08 -0700
Subject: [VoIP] Correct Way To Connect X100P To Step By Step Exchange?
In-Reply-To: <354777.39044.qm@web51610.mail.re2.yahoo.com>
References: <023301c7d3f4$133d8bb0$5f4665da@compaq>
	<354777.39044.qm@web51610.mail.re2.yahoo.com>
Message-ID: <20070802035908.GO29193@5.7.5.5.6.6.6.6.0.2.1.e164.arpa>

On Wed, Aug 01, 2007 at 05:36:49AM -0700, ikjtel wrote:
> immediately upon receipt of any pulse (or even
> an inductuve transient, as PDW has found) it will go
> into a guarded state in which its audio ckts are cut
> out of the loop and not reconnected for approx.
> 150-250 msec.

My CO, a 5ESS, occasionally seems to do that.  Sometimes when I
pulse-dial after a call has been connected, the audio is cut off for a
moment.  When it comes back, I can hear a click, probably from the
far-end echo.

I tried it against a foreign office's milliwatt and silent
termination, but I couldn't hear it; thus it's most likely far-end
echo.

I then dialled my voicemail and said "four", dialed four, and
continued to say four at the same volume, stopping three times after I
heard the click.  You can hear it for yourself at
<http://students.washington.edu/f/host/four-pulses.wav>.  Apparently
the CO is interpreting the pulses as audio, adjusting my transmitted
volume to be reasonable.

I probably shouldn't have heard the click on my voicemail, though,
because there ought to be no echo as it is (presumably) fully digital.

-- 
Duncan Smith  --------\    http://students.washington.edu/f/    /---
  ()  ascii ribbon     \--- Signed/encrypted mail preferred ---/
  /\    campaign    [ against html mail ]  [ support open formats ]

From ikj1234i at yahoo.com  Thu Aug  2 07:58:49 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Thu, 2 Aug 2007 05:58:49 -0700 (PDT)
Subject: [VoIP] Correct Way To Connect X100P To Step By Step Exchange?
In-Reply-To: <354777.39044.qm@web51610.mail.re2.yahoo.com>
Message-ID: <586105.96999.qm@web51604.mail.re2.yahoo.com>

> --- Brent Jones <gijones at clear.net.nz> wrote
> >
> > Also are X100P cards polarity sensitive with
> > relationship to the line voltages ie
> > does it matter which pin gets A or B leg .

In fact, it appears that it's not even possible for
the PC software (zaptel and/or asterisk) to detect at
which polarity the line is connected.

I ran some X100P card tests back in 2005 and posted
the results, complete with graphs - see
  http://www.lightlink.com/mhp/x100p/2.html

One of the tests involved repeated battery
reversals...

Max


       
____________________________________________________________________________________
Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. 
http://answers.yahoo.com/dir/?link=list&sid=396545433

From ratguy at insightbb.com  Sun Aug  5 09:52:50 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Sun, 5 Aug 2007 10:52:50 -0400
Subject: [VoIP] Old answering machine simulation
Message-ID: <000501c7d770$4bc725c0$6700a8c0@bluegrasspals.com>

Hi,

     So, I got bored out of my mind this morning, and decided to put
together a simulation of an answering machine from the old days. This new
innovative voicemail system has all the features you would expect. Time and
date stamping? Not! Ability to skip, delete, go back? Not! Ability to mark
messages urgent, forward them, reply? Not! Privacy and security? Not! No,
folks, this is a simulation of an old tape-based answering machine. Anyone
can call +1 622-0014, wait for the tone, and leave a message. Messages are
supposed to be limited to three minutes, but I haven't checked this. Then,
to hear all the messages that have been left, just dial +1 622-0015. There's
no skipping around or anything like that. Not really all that useful, but
still, a fun little toy to play around with. If anyone wants to know exactly
how I did it, just let me know. Too bad I couldn't simulate tape hiss, wow
and flutter, and all those other wonderful analog effects!
Jayson

From dfroula at sbcglobal.net  Sun Aug  5 10:22:48 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Sun, 5 Aug 2007 08:22:48 -0700 (PDT)
Subject: [VoIP] Old answering machine simulation
In-Reply-To: <000501c7d770$4bc725c0$6700a8c0@bluegrasspals.com>
Message-ID: <485522.25302.qm@web83213.mail.mud.yahoo.com>

Jayson, I'd like know how you set this up. I'd like to
set up a simple record and playback system where
people dialing into my ProjectMF trunks could leave
and play back comments without having to traverse the
Asterisk voicemail system.

Don

--- Jayson Smith <ratguy at insightbb.com> wrote:

> Hi,
> 
>      So, I got bored out of my mind this morning,
> and decided to put
> together a simulation of an answering machine from
> the old days. This new
> innovative voicemail system has all the features you
> would expect. Time and
> date stamping? Not! Ability to skip, delete, go
> back? Not! Ability to mark
> messages urgent, forward them, reply? Not! Privacy
> and security? Not! No,
> folks, this is a simulation of an old tape-based
> answering machine. Anyone
> can call +1 622-0014, wait for the tone, and leave a
> message. Messages are
> supposed to be limited to three minutes, but I
> haven't checked this. Then,
> to hear all the messages that have been left, just
> dial +1 622-0015. There's
> no skipping around or anything like that. Not really
> all that useful, but
> still, a fun little toy to play around with. If
> anyone wants to know exactly
> how I did it, just let me know. Too bad I couldn't
> simulate tape hiss, wow
> and flutter, and all those other wonderful analog
> effects!
> Jayson
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From ratguy at insightbb.com  Sun Aug  5 10:27:59 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Sun, 5 Aug 2007 11:27:59 -0400
Subject: [VoIP] Asterisk and Payphones
References: <46AE7B9B.30204@rudholm.com>
	<024701c7d30b$5b937580$750a0a0a@hickory.vyger.net>
	<46AEC579.5060407@rudholm.com>
	<005501c7d36e$61c17480$25445d80$@com>
Message-ID: <002c01c7d775$3487a9c0$6700a8c0@bluegrasspals.com>

Hi,
Actually, I'd originally come up with a different approach for handling coin
collect and return. The approach I'd thought of, as well as others, I think,
was using the bluebox/greenbox tones to signal a box which sat between
Asterisk and a coin phone. The problem with this approach is that if the
coin phone hangs up, Asterisk is never going to get to send the appropriate
tone down the line, so the coin(s) will just sit there in the holding
chamber or whatever the technical term is, until another call is made.
     Would this circuit work with an ATA, or just with an FXS card from the
Asterisk box, or either way? The way I'm seeing this, the computer has a
connection into a box. The box has two phone connections. One goes to a
source of phone service, be it an FXS port, an ATA, or whatever. The other
goes into a coin phone.
     Assuming this project gets completed, or at least in a workable state,
I might have to think about seriously looking for coin phones. I've always
been afraid, if I actually bought one on eBay, that the seller would ask all
sorts of questions like, "What do you want this for? You want to have a
payphone in your home? Don't you know that under federal law number
1234-56789-x3-qrj49, paragraph 29, subparagraph 4C, subsection 21J, it is
illegal to have a working payphone in your home, with the minimum penalty
being fifty years sitting in one room and listening to the same telephone
recording again and again and again?" Ok that got a bit crazy, but until
now, the only type of payphone which would have even been useful to me as a
working payphone would be a COCOT. I assume those phones you see on eBay,
converted for home use, no coins needed, any coins deposited go straight
into the coin box, have had all their coin guts ripped out so as not to be
convertible into a working payphone again?
     On another subject, does anybody have a full set of ACTS recordings, or
were you figuring on having Allison record some things, or what?
Jayson

----- Original Message ----- 
From: "Lee Spenadel" <lee at spenadel.com>
To: "'Voice Over IP Tandem for Analog Switches'" <voip at ckts.info>
Sent: Tuesday, July 31, 2007 8:29 AM
Subject: Re: [VoIP] Asterisk and Payphones


> Mark,
>
> My BSP for the 1D/2D Coin Telephone Set shows:
>
> Operate Current: 41 mA
> Non-Operate Current 30 mA
> Operate Time of 450  +/- 50 ms
>
> Lee
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Mark Rudholm
> Sent: Tuesday, July 31, 2007 1:16 AM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Asterisk and Payphones
>
> Well, I'm glad to hear there is significant interest in this.
>
> For the moment, I'm working on coin-relay control.  My first
> draft of the schematic is available here:
> http://rudholm.com/coin-control.pdf (use "File, Rotate" to
> fix the orientation in Acrobat Reader).
>
> My idea is just to take two TTL lines from the computer, either
> from the parallel port or from a USB to parallel adapter
> (like the Elexol USBMOD4), one for return vs. collect and
> the other as a strobe to trigger the relay.  If I use the
> parallel port, I'll need to add a +5VDC power supply to my
> interface box to power a TTL buffer since parallel ports
> can't source or sink enough current to safely drive those
> solid-state relays.  The Elexol interfaces can source enough
> current to operate them however, and also since USB ports
> have a nice 500mA +5VDC supply line, I could throw in a
> buffer as well.
>
> I'll build the circuit described in that schematic on a
> proto board as soon as my Mouser order for the parts arrives.
> Once I get it working right and get the relay timing right,
> then I'll start thinking about the TTL control lines.
>
> Speaking of which, if someone could look up in their Coin
> Crafts or Coin Services Manual Volume I what the coin relay
> timing is, I need that information.  I'm guessing it's
> about 500ms but would like verification.
>
> Once the hardware is done, it shouldn't be too hard to
> do a simple return or collect from Asterisk based on whether
> the call supervised or not.
>
> Mark Spencer pointed out to me that DTMF detection is
> actually not done in zaptel but in a simple goertzel analysis
> in dsp.c, which makes sense for DTMF decoding.  He suggested
> I just add a new column and row for 1700 and 2200Hz coin
> tone.  I figure I'll start hacking the code once I get the
> hardware working.  I would ultimately like to be able to
> fully emulate ACTS.
>
> I would eventually like to add the Stuck Coin, Coin
> Presence, and initial rate deposit tests to the system.
>
> Support for coin first phones would be nice, but I need
> to get my hands on the Coin Services Manual to even start
> thinking about how I'd support all these functions.
>
> Three-slot support would be interesting.  I understand the
> coin bells are tuned so it shouldn't be too hard to
> add the two tones to the goertzel analysis (one for the
> nickel/dime bell, and one for the quarter gong).
>
> My ultimate goal would be to publish a patch set for the
> relevant files in asterisk and a schematic and Mouser
> part number list.  The circuit is too simple to really
> justify custom printed circuit board artwork (it someone
> wanted to do that, that'd be fine with me, though).
>
> -Mark
>
> Greg Blakely wrote:
> > I don't know if you'd be able to include a 3-slot patch in the project,
> but
> > my payphone is a pre-pay 3-slot Automatic Electric.  It has the "ding"
> bell
> > for nickels and dimes, but using a "gong" for a quarter.
> _______________________________________________
> 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/
>

From stfkerman at jps.net  Sun Aug  5 10:38:39 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Sun, 05 Aug 2007 11:38:39 -0400
Subject: [VoIP] Asterisk and Payphones
In-Reply-To: <002c01c7d775$3487a9c0$6700a8c0@bluegrasspals.com>
References: <46AE7B9B.30204@rudholm.com>	<024701c7d30b$5b937580$750a0a0a@hickory.vyger.net>	<46AEC579.5060407@rudholm.com>	<005501c7d36e$61c17480$25445d80$@com>
	<002c01c7d775$3487a9c0$6700a8c0@bluegrasspals.com>
Message-ID: <46B5EEFF.3080903@jps.net>

Jayson,

The problem of the phone's hanging up preventing coin disposal is easily 
solved.  The coin control adapter you envision can have control over 
supervision towards the FXS or ATA.  If so, even if the telephone goes 
on hook the coin control adapter can keep the FXS or ATA off hook until 
it receives the coin disposal tone and executes it.

Steph

Jayson Smith wrote:
> Hi,
> Actually, I'd originally come up with a different approach for handling coin
> collect and return. The approach I'd thought of, as well as others, I think,
> was using the bluebox/greenbox tones to signal a box which sat between
> Asterisk and a coin phone. The problem with this approach is that if the
> coin phone hangs up, Asterisk is never going to get to send the appropriate
> tone down the line, so the coin(s) will just sit there in the holding
> chamber or whatever the technical term is, until another call is made.
>      Would this circuit work with an ATA, or just with an FXS card from the
> Asterisk box, or either way? The way I'm seeing this, the computer has a
> connection into a box. The box has two phone connections. One goes to a
> source of phone service, be it an FXS port, an ATA, or whatever. The other
> goes into a coin phone.
>      Assuming this project gets completed, or at least in a workable state,
> I might have to think about seriously looking for coin phones. I've always
> been afraid, if I actually bought one on eBay, that the seller would ask all
> sorts of questions like, "What do you want this for? You want to have a
> payphone in your home? Don't you know that under federal law number
> 1234-56789-x3-qrj49, paragraph 29, subparagraph 4C, subsection 21J, it is
> illegal to have a working payphone in your home, with the minimum penalty
> being fifty years sitting in one room and listening to the same telephone
> recording again and again and again?" Ok that got a bit crazy, but until
> now, the only type of payphone which would have even been useful to me as a
> working payphone would be a COCOT. I assume those phones you see on eBay,
> converted for home use, no coins needed, any coins deposited go straight
> into the coin box, have had all their coin guts ripped out so as not to be
> convertible into a working payphone again?
>      On another subject, does anybody have a full set of ACTS recordings, or
> were you figuring on having Allison record some things, or what?
> Jayson
>
> ----- Original Message ----- 
> From: "Lee Spenadel" <lee at spenadel.com>
> To: "'Voice Over IP Tandem for Analog Switches'" <voip at ckts.info>
> Sent: Tuesday, July 31, 2007 8:29 AM
> Subject: Re: [VoIP] Asterisk and Payphones
>
>
>   
>> Mark,
>>
>> My BSP for the 1D/2D Coin Telephone Set shows:
>>
>> Operate Current: 41 mA
>> Non-Operate Current 30 mA
>> Operate Time of 450  +/- 50 ms
>>
>> Lee
>>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
>> Mark Rudholm
>> Sent: Tuesday, July 31, 2007 1:16 AM
>> To: Voice Over IP Tandem for Analog Switches
>> Subject: Re: [VoIP] Asterisk and Payphones
>>
>> Well, I'm glad to hear there is significant interest in this.
>>
>> For the moment, I'm working on coin-relay control.  My first
>> draft of the schematic is available here:
>> http://rudholm.com/coin-control.pdf (use "File, Rotate" to
>> fix the orientation in Acrobat Reader).
>>
>> My idea is just to take two TTL lines from the computer, either
>> from the parallel port or from a USB to parallel adapter
>> (like the Elexol USBMOD4), one for return vs. collect and
>> the other as a strobe to trigger the relay.  If I use the
>> parallel port, I'll need to add a +5VDC power supply to my
>> interface box to power a TTL buffer since parallel ports
>> can't source or sink enough current to safely drive those
>> solid-state relays.  The Elexol interfaces can source enough
>> current to operate them however, and also since USB ports
>> have a nice 500mA +5VDC supply line, I could throw in a
>> buffer as well.
>>
>> I'll build the circuit described in that schematic on a
>> proto board as soon as my Mouser order for the parts arrives.
>> Once I get it working right and get the relay timing right,
>> then I'll start thinking about the TTL control lines.
>>
>> Speaking of which, if someone could look up in their Coin
>> Crafts or Coin Services Manual Volume I what the coin relay
>> timing is, I need that information.  I'm guessing it's
>> about 500ms but would like verification.
>>
>> Once the hardware is done, it shouldn't be too hard to
>> do a simple return or collect from Asterisk based on whether
>> the call supervised or not.
>>
>> Mark Spencer pointed out to me that DTMF detection is
>> actually not done in zaptel but in a simple goertzel analysis
>> in dsp.c, which makes sense for DTMF decoding.  He suggested
>> I just add a new column and row for 1700 and 2200Hz coin
>> tone.  I figure I'll start hacking the code once I get the
>> hardware working.  I would ultimately like to be able to
>> fully emulate ACTS.
>>
>> I would eventually like to add the Stuck Coin, Coin
>> Presence, and initial rate deposit tests to the system.
>>
>> Support for coin first phones would be nice, but I need
>> to get my hands on the Coin Services Manual to even start
>> thinking about how I'd support all these functions.
>>
>> Three-slot support would be interesting.  I understand the
>> coin bells are tuned so it shouldn't be too hard to
>> add the two tones to the goertzel analysis (one for the
>> nickel/dime bell, and one for the quarter gong).
>>
>> My ultimate goal would be to publish a patch set for the
>> relevant files in asterisk and a schematic and Mouser
>> part number list.  The circuit is too simple to really
>> justify custom printed circuit board artwork (it someone
>> wanted to do that, that'd be fine with me, though).
>>
>> -Mark
>>
>> Greg Blakely wrote:
>>     
>>> I don't know if you'd be able to include a 3-slot patch in the project,
>>>       
>> but
>>     
>>> my payphone is a pre-pay 3-slot Automatic Electric.  It has the "ding"
>>>       
>> bell
>>     
>>> for nickels and dimes, but using a "gong" for a quarter.
>>>       
>> _______________________________________________
>> 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/
>
>   

From ratguy at insightbb.com  Sun Aug  5 10:51:13 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Sun, 5 Aug 2007 11:51:13 -0400
Subject: [VoIP] Asterisk and Payphones
References: <46AE7B9B.30204@rudholm.com>
	<024701c7d30b$5b937580$750a0a0a@hickory.vyger.net>
	<46AEC579.5060407@rudholm.com>
	<005501c7d36e$61c17480$25445d80$@com>
	<002c01c7d775$3487a9c0$6700a8c0@bluegrasspals.com>
	<46B5EEFF.3080903@jps.net>
Message-ID: <000501c7d778$738e4720$6700a8c0@bluegrasspals.com>

Hi,
That's true. But, without Asterisk knowing the coin phone has been hung up,
how would it know to send the coin disposal tone? Seems to me, my little box
would sit around waiting forever, for a coin disposal tone which would never
come, since Asterisk thinks the phone is still off the hook.
Jayson

----- Original Message ----- 
From: "Steph Kerman" <stfkerman at jps.net>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Sunday, August 05, 2007 11:38 AM
Subject: Re: [VoIP] Asterisk and Payphones


> Jayson,
>
> The problem of the phone's hanging up preventing coin disposal is easily
> solved.  The coin control adapter you envision can have control over
> supervision towards the FXS or ATA.  If so, even if the telephone goes
> on hook the coin control adapter can keep the FXS or ATA off hook until
> it receives the coin disposal tone and executes it.
>
> Steph
>
> Jayson Smith wrote:
> > Hi,
> > Actually, I'd originally come up with a different approach for handling
coin
> > collect and return. The approach I'd thought of, as well as others, I
think,
> > was using the bluebox/greenbox tones to signal a box which sat between
> > Asterisk and a coin phone. The problem with this approach is that if the
> > coin phone hangs up, Asterisk is never going to get to send the
appropriate
> > tone down the line, so the coin(s) will just sit there in the holding
> > chamber or whatever the technical term is, until another call is made.
> >      Would this circuit work with an ATA, or just with an FXS card from
the
> > Asterisk box, or either way? The way I'm seeing this, the computer has a
> > connection into a box. The box has two phone connections. One goes to a
> > source of phone service, be it an FXS port, an ATA, or whatever. The
other
> > goes into a coin phone.
> >      Assuming this project gets completed, or at least in a workable
state,
> > I might have to think about seriously looking for coin phones. I've
always
> > been afraid, if I actually bought one on eBay, that the seller would ask
all
> > sorts of questions like, "What do you want this for? You want to have a
> > payphone in your home? Don't you know that under federal law number
> > 1234-56789-x3-qrj49, paragraph 29, subparagraph 4C, subsection 21J, it
is
> > illegal to have a working payphone in your home, with the minimum
penalty
> > being fifty years sitting in one room and listening to the same
telephone
> > recording again and again and again?" Ok that got a bit crazy, but until
> > now, the only type of payphone which would have even been useful to me
as a
> > working payphone would be a COCOT. I assume those phones you see on
eBay,
> > converted for home use, no coins needed, any coins deposited go straight
> > into the coin box, have had all their coin guts ripped out so as not to
be
> > convertible into a working payphone again?
> >      On another subject, does anybody have a full set of ACTS
recordings, or
> > were you figuring on having Allison record some things, or what?
> > Jayson
> >
> > ----- Original Message ----- 
> > From: "Lee Spenadel" <lee at spenadel.com>
> > To: "'Voice Over IP Tandem for Analog Switches'" <voip at ckts.info>
> > Sent: Tuesday, July 31, 2007 8:29 AM
> > Subject: Re: [VoIP] Asterisk and Payphones
> >
> >
> >
> >> Mark,
> >>
> >> My BSP for the 1D/2D Coin Telephone Set shows:
> >>
> >> Operate Current: 41 mA
> >> Non-Operate Current 30 mA
> >> Operate Time of 450  +/- 50 ms
> >>
> >> Lee
> >>
> >> -----Original Message-----
> >> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf
Of
> >> Mark Rudholm
> >> Sent: Tuesday, July 31, 2007 1:16 AM
> >> To: Voice Over IP Tandem for Analog Switches
> >> Subject: Re: [VoIP] Asterisk and Payphones
> >>
> >> Well, I'm glad to hear there is significant interest in this.
> >>
> >> For the moment, I'm working on coin-relay control.  My first
> >> draft of the schematic is available here:
> >> http://rudholm.com/coin-control.pdf (use "File, Rotate" to
> >> fix the orientation in Acrobat Reader).
> >>
> >> My idea is just to take two TTL lines from the computer, either
> >> from the parallel port or from a USB to parallel adapter
> >> (like the Elexol USBMOD4), one for return vs. collect and
> >> the other as a strobe to trigger the relay.  If I use the
> >> parallel port, I'll need to add a +5VDC power supply to my
> >> interface box to power a TTL buffer since parallel ports
> >> can't source or sink enough current to safely drive those
> >> solid-state relays.  The Elexol interfaces can source enough
> >> current to operate them however, and also since USB ports
> >> have a nice 500mA +5VDC supply line, I could throw in a
> >> buffer as well.
> >>
> >> I'll build the circuit described in that schematic on a
> >> proto board as soon as my Mouser order for the parts arrives.
> >> Once I get it working right and get the relay timing right,
> >> then I'll start thinking about the TTL control lines.
> >>
> >> Speaking of which, if someone could look up in their Coin
> >> Crafts or Coin Services Manual Volume I what the coin relay
> >> timing is, I need that information.  I'm guessing it's
> >> about 500ms but would like verification.
> >>
> >> Once the hardware is done, it shouldn't be too hard to
> >> do a simple return or collect from Asterisk based on whether
> >> the call supervised or not.
> >>
> >> Mark Spencer pointed out to me that DTMF detection is
> >> actually not done in zaptel but in a simple goertzel analysis
> >> in dsp.c, which makes sense for DTMF decoding.  He suggested
> >> I just add a new column and row for 1700 and 2200Hz coin
> >> tone.  I figure I'll start hacking the code once I get the
> >> hardware working.  I would ultimately like to be able to
> >> fully emulate ACTS.
> >>
> >> I would eventually like to add the Stuck Coin, Coin
> >> Presence, and initial rate deposit tests to the system.
> >>
> >> Support for coin first phones would be nice, but I need
> >> to get my hands on the Coin Services Manual to even start
> >> thinking about how I'd support all these functions.
> >>
> >> Three-slot support would be interesting.  I understand the
> >> coin bells are tuned so it shouldn't be too hard to
> >> add the two tones to the goertzel analysis (one for the
> >> nickel/dime bell, and one for the quarter gong).
> >>
> >> My ultimate goal would be to publish a patch set for the
> >> relevant files in asterisk and a schematic and Mouser
> >> part number list.  The circuit is too simple to really
> >> justify custom printed circuit board artwork (it someone
> >> wanted to do that, that'd be fine with me, though).
> >>
> >> -Mark
> >>
> >> Greg Blakely wrote:
> >>
> >>> I don't know if you'd be able to include a 3-slot patch in the
project,
> >>>
> >> but
> >>
> >>> my payphone is a pre-pay 3-slot Automatic Electric.  It has the "ding"
> >>>
> >> bell
> >>
> >>> for nickels and dimes, but using a "gong" for a quarter.
> >>>
> >> _______________________________________________
> >> 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/
> >
> >
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/

From stfkerman at jps.net  Sun Aug  5 11:40:43 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Sun, 05 Aug 2007 12:40:43 -0400
Subject: [VoIP] Asterisk and Payphones
In-Reply-To: <000501c7d778$738e4720$6700a8c0@bluegrasspals.com>
References: <46AE7B9B.30204@rudholm.com>	<024701c7d30b$5b937580$750a0a0a@hickory.vyger.net>	<46AEC579.5060407@rudholm.com>	<005501c7d36e$61c17480$25445d80$@com>	<002c01c7d775$3487a9c0$6700a8c0@bluegrasspals.com>	<46B5EEFF.3080903@jps.net>
	<000501c7d778$738e4720$6700a8c0@bluegrasspals.com>
Message-ID: <46B5FD8B.1090909@jps.net>

The adapter could send a 4th column DTMF digit or special digit sequence 
using * and/or #.  Or it could send a special timed wink or dial pulse 
digit.  Lots of possibilities.  Since * knows that this is the 
disconnect at end of the call, even dial pulse digits could be used as 
control signals without possible misinterpretation.

Steph

Jayson Smith wrote:
> Hi,
> That's true. But, without Asterisk knowing the coin phone has been hung up,
> how would it know to send the coin disposal tone? Seems to me, my little box
> would sit around waiting forever, for a coin disposal tone which would never
> come, since Asterisk thinks the phone is still off the hook.
> Jayson
>
> ----- Original Message ----- 
> From: "Steph Kerman" <stfkerman at jps.net>
> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> Sent: Sunday, August 05, 2007 11:38 AM
> Subject: Re: [VoIP] Asterisk and Payphones
>
>
>   
>> Jayson,
>>
>> The problem of the phone's hanging up preventing coin disposal is easily
>> solved.  The coin control adapter you envision can have control over
>> supervision towards the FXS or ATA.  If so, even if the telephone goes
>> on hook the coin control adapter can keep the FXS or ATA off hook until
>> it receives the coin disposal tone and executes it.
>>
>> Steph
>>
>> Jayson Smith wrote:
>>     
>>> Hi,
>>> Actually, I'd originally come up with a different approach for handling
>>>       
> coin
>   
>>> collect and return. The approach I'd thought of, as well as others, I
>>>       
> think,
>   
>>> was using the bluebox/greenbox tones to signal a box which sat between
>>> Asterisk and a coin phone. The problem with this approach is that if the
>>> coin phone hangs up, Asterisk is never going to get to send the
>>>       
> appropriate
>   
>>> tone down the line, so the coin(s) will just sit there in the holding
>>> chamber or whatever the technical term is, until another call is made.
>>>      Would this circuit work with an ATA, or just with an FXS card from
>>>       
> the
>   
>>> Asterisk box, or either way? The way I'm seeing this, the computer has a
>>> connection into a box. The box has two phone connections. One goes to a
>>> source of phone service, be it an FXS port, an ATA, or whatever. The
>>>       
> other
>   
>>> goes into a coin phone.
>>>      Assuming this project gets completed, or at least in a workable
>>>       
> state,
>   
>>> I might have to think about seriously looking for coin phones. I've
>>>       
> always
>   
>>> been afraid, if I actually bought one on eBay, that the seller would ask
>>>       
> all
>   
>>> sorts of questions like, "What do you want this for? You want to have a
>>> payphone in your home? Don't you know that under federal law number
>>> 1234-56789-x3-qrj49, paragraph 29, subparagraph 4C, subsection 21J, it
>>>       
> is
>   
>>> illegal to have a working payphone in your home, with the minimum
>>>       
> penalty
>   
>>> being fifty years sitting in one room and listening to the same
>>>       
> telephone
>   
>>> recording again and again and again?" Ok that got a bit crazy, but until
>>> now, the only type of payphone which would have even been useful to me
>>>       
> as a
>   
>>> working payphone would be a COCOT. I assume those phones you see on
>>>       
> eBay,
>   
>>> converted for home use, no coins needed, any coins deposited go straight
>>> into the coin box, have had all their coin guts ripped out so as not to
>>>       
> be
>   
>>> convertible into a working payphone again?
>>>      On another subject, does anybody have a full set of ACTS
>>>       
> recordings, or
>   
>>> were you figuring on having Allison record some things, or what?
>>> Jayson
>>>
>>> ----- Original Message ----- 
>>> From: "Lee Spenadel" <lee at spenadel.com>
>>> To: "'Voice Over IP Tandem for Analog Switches'" <voip at ckts.info>
>>> Sent: Tuesday, July 31, 2007 8:29 AM
>>> Subject: Re: [VoIP] Asterisk and Payphones
>>>
>>>
>>>
>>>       
>>>> Mark,
>>>>
>>>> My BSP for the 1D/2D Coin Telephone Set shows:
>>>>
>>>> Operate Current: 41 mA
>>>> Non-Operate Current 30 mA
>>>> Operate Time of 450  +/- 50 ms
>>>>
>>>> Lee
>>>>
>>>> -----Original Message-----
>>>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf
>>>>         
> Of
>   
>>>> Mark Rudholm
>>>> Sent: Tuesday, July 31, 2007 1:16 AM
>>>> To: Voice Over IP Tandem for Analog Switches
>>>> Subject: Re: [VoIP] Asterisk and Payphones
>>>>
>>>> Well, I'm glad to hear there is significant interest in this.
>>>>
>>>> For the moment, I'm working on coin-relay control.  My first
>>>> draft of the schematic is available here:
>>>> http://rudholm.com/coin-control.pdf (use "File, Rotate" to
>>>> fix the orientation in Acrobat Reader).
>>>>
>>>> My idea is just to take two TTL lines from the computer, either
>>>> from the parallel port or from a USB to parallel adapter
>>>> (like the Elexol USBMOD4), one for return vs. collect and
>>>> the other as a strobe to trigger the relay.  If I use the
>>>> parallel port, I'll need to add a +5VDC power supply to my
>>>> interface box to power a TTL buffer since parallel ports
>>>> can't source or sink enough current to safely drive those
>>>> solid-state relays.  The Elexol interfaces can source enough
>>>> current to operate them however, and also since USB ports
>>>> have a nice 500mA +5VDC supply line, I could throw in a
>>>> buffer as well.
>>>>
>>>> I'll build the circuit described in that schematic on a
>>>> proto board as soon as my Mouser order for the parts arrives.
>>>> Once I get it working right and get the relay timing right,
>>>> then I'll start thinking about the TTL control lines.
>>>>
>>>> Speaking of which, if someone could look up in their Coin
>>>> Crafts or Coin Services Manual Volume I what the coin relay
>>>> timing is, I need that information.  I'm guessing it's
>>>> about 500ms but would like verification.
>>>>
>>>> Once the hardware is done, it shouldn't be too hard to
>>>> do a simple return or collect from Asterisk based on whether
>>>> the call supervised or not.
>>>>
>>>> Mark Spencer pointed out to me that DTMF detection is
>>>> actually not done in zaptel but in a simple goertzel analysis
>>>> in dsp.c, which makes sense for DTMF decoding.  He suggested
>>>> I just add a new column and row for 1700 and 2200Hz coin
>>>> tone.  I figure I'll start hacking the code once I get the
>>>> hardware working.  I would ultimately like to be able to
>>>> fully emulate ACTS.
>>>>
>>>> I would eventually like to add the Stuck Coin, Coin
>>>> Presence, and initial rate deposit tests to the system.
>>>>
>>>> Support for coin first phones would be nice, but I need
>>>> to get my hands on the Coin Services Manual to even start
>>>> thinking about how I'd support all these functions.
>>>>
>>>> Three-slot support would be interesting.  I understand the
>>>> coin bells are tuned so it shouldn't be too hard to
>>>> add the two tones to the goertzel analysis (one for the
>>>> nickel/dime bell, and one for the quarter gong).
>>>>
>>>> My ultimate goal would be to publish a patch set for the
>>>> relevant files in asterisk and a schematic and Mouser
>>>> part number list.  The circuit is too simple to really
>>>> justify custom printed circuit board artwork (it someone
>>>> wanted to do that, that'd be fine with me, though).
>>>>
>>>> -Mark
>>>>
>>>> Greg Blakely wrote:
>>>>
>>>>         
>>>>> I don't know if you'd be able to include a 3-slot patch in the
>>>>>           
> project,
>   
>>>> but
>>>>
>>>>         
>>>>> my payphone is a pre-pay 3-slot Automatic Electric.  It has the "ding"
>>>>>
>>>>>           
>>>> bell
>>>>
>>>>         
>>>>> for nickels and dimes, but using a "gong" for a quarter.
>>>>>
>>>>>           
>>>> _______________________________________________
>>>> 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/
>>>
>>>
>>>       
>> _______________________________________________
>> 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/
>
>   

From ratguy at insightbb.com  Mon Aug  6 07:16:07 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Mon, 6 Aug 2007 08:16:07 -0400
Subject: [VoIP] Fw: [C/O X-Bar] North CX-30/Free Trailer
Message-ID: <000a01c7d823$915a07c0$6700a8c0@bluegrasspals.com>

Hi,
Thought someone over here might have some interest in this. Everyone on this
list who is looking for such is probably already on Centraloffice, but just
thought I'd pass the word.
Jayson.


----- Original Message ----- 
From: "Steve Cichorsky" <steve at telephonepioneer.net>
To: "Centraloffice 3Post" <centraloffice at yahoogroups.com>
Sent: Sunday, August 05, 2007 8:37 PM
Subject: [C/O X-Bar] North CX-30/Free Trailer


> I will soon be making arrangements to pick up a North CX relay switch with
> associated rack with the power/supervisory, coin and operator trunks,
> located in Utah.
>
>
>
> This is the switch that I have previously discussed on the list.
>
>
>
> The only hitch is that it is housed in an old single-wide house trailer
> which the telephone company wants towed off along with the equipment
inside
> of it.
>
>
>
>  Since I will be driving a Ford F250 diesel pickup, I don?t relish the
idea
> of towing a large trailer all the way back to California.
>
>
>
>  I would like to transfer everything into my smaller enclosed cargo
trailer
> and give the other trailer to anyone who wants it, presumably someone
living
> in Nevada or Utah. If there is any interest, please contact me.
>
>
>
> Thanks-
>
>
>
> Steve Cichorsky
>
> steve at telephonepioneer.net
>
>
>
>
> Yahoo! Groups Links
>
> <*> To visit your group on the web, go to:
>     http://groups.yahoo.com/group/centraloffice/
>
> <*> Your email settings:
>     Individual Email | Traditional
>
> <*> To change settings online go to:
>     http://groups.yahoo.com/group/centraloffice/join
>     (Yahoo! ID required)
>
> <*> To change settings via email:
>     mailto:centraloffice-digest at yahoogroups.com
>     mailto:centraloffice-fullfeatured at yahoogroups.com
>
> <*> To unsubscribe from this group, send an email to:
>     centraloffice-unsubscribe at yahoogroups.com
>
> <*> Your use of Yahoo! Groups is subject to:
>     http://docs.yahoo.com/info/terms/
>
>

From kb0tdf at yahoo.com  Wed Aug  8 21:49:18 2007
From: kb0tdf at yahoo.com (Greg Blakely)
Date: Wed, 8 Aug 2007 21:49:18 -0500
Subject: [VoIP] Test
Message-ID: <000001c7da2f$e2a990b0$750a0a0a@hickory.vyger.net>

Okay, so I broke the list again....  What else is new, right?  I'm about 1/2
a second away from putting the sucker up on Yahoo.
 
I sent this to voip at lists.ckts.info rather than plain old ckts.info.
 
Let's see if I have any success.

From jnovack at stromberg-carlson.org  Thu Aug  9 08:28:06 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Thu, 09 Aug 2007 09:28:06 -0400
Subject: [VoIP] Test
In-Reply-To: <000001c7da2f$e2a990b0$750a0a0a@hickory.vyger.net>
References: <000001c7da2f$e2a990b0$750a0a0a@hickory.vyger.net>
Message-ID: <46BB1666.408@stromberg-carlson.org>

You might do better with Google lists.
Yahoo is really overloaded.

Really most of the hosting stuff these days can be done for reasonable 
amounts by companies with data centers.
It really doesn't make good economic sense to host something on a 
basement anymore.

John Novack


Greg Blakely wrote:
> Okay, so I broke the list again....  What else is new, right?  I'm about 1/2
> a second away from putting the sucker up on Yahoo.
>  
> I sent this to voip at lists.ckts.info rather than plain old ckts.info.
>  
> Let's see if I have any success.
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From ratguy at insightbb.com  Thu Aug  9 13:24:28 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Thu, 9 Aug 2007 14:24:28 -0400
Subject: [VoIP] We've lost a pioneer
Message-ID: <000501c7dab2$85ff8da0$6700a8c0@bluegrasspals.com>

Joe Engressia, A.K.A. Joe the Whistler, A.K.A. Joybubbles passed
away in his apartment approximately two days ago. He was found by
management personnel after a phone friend grew concerned when he
hadn't heard from him in several days. His family in Florida was
notified. He was fifty-eight years old.
     As most of you know, his ordeals with the phone company over
the blue box issue were noted in the October 1971 Esquire Magazine
article
"Secrets of the Little Blue Box." Recent interests included
humorous telephone postings called Phlogs on voicemail services.
He wrote Telephiles for Playback Magazine. His interests also
included children's literature and paraphernalia..
     Although I never knew Joe personally, I did talk to him on the
Wawina, Minnesota conference line which was set up on June 15,
2006, when Wawina cut over from N2 to T1 for its toll trunks. He
also left many messages on a message board which was set up in the
weeks prior to the cutover.
     Please post this message to any other lists where you feel it
would be appropriate, or send it to individuals. I am not in close
contact with any of the old phone phreak crowd, but I'm sure some
of you are, and can get the news out.
Jayson

From dfroula at sbcglobal.net  Thu Aug  9 13:50:26 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Thu, 9 Aug 2007 11:50:26 -0700 (PDT)
Subject: [VoIP] We've lost a pioneer
In-Reply-To: <000501c7dab2$85ff8da0$6700a8c0@bluegrasspals.com>
Message-ID: <388679.11256.qm@web83213.mail.mud.yahoo.com>

I'm very sorry to hear this. I spoke to Joe just a few
months ago. I was watching a Discovery Channel video
clip of him a few days ago, "dialing" a phone number
into an old 2600 trunk by whistling pulses of 2600. He
was an interesting guy.

Jayson where did you spot the news?

Don

--- Jayson Smith <ratguy at insightbb.com> wrote:

> Joe Engressia, A.K.A. Joe the Whistler, A.K.A.
> Joybubbles passed
> away in his apartment approximately two days ago. He
> was found by
> management personnel after a phone friend grew
> concerned when he
> hadn't heard from him in several days. His family in
> Florida was
> notified. He was fifty-eight years old.
>      As most of you know, his ordeals with the phone
> company over
> the blue box issue were noted in the October 1971
> Esquire Magazine
> article
> "Secrets of the Little Blue Box." Recent interests
> included
> humorous telephone postings called Phlogs on
> voicemail services.
> He wrote Telephiles for Playback Magazine. His
> interests also
> included children's literature and paraphernalia..
>      Although I never knew Joe personally, I did
> talk to him on the
> Wawina, Minnesota conference line which was set up
> on June 15,
> 2006, when Wawina cut over from N2 to T1 for its
> toll trunks. He
> also left many messages on a message board which was
> set up in the
> weeks prior to the cutover.
>      Please post this message to any other lists
> where you feel it
> would be appropriate, or send it to individuals. I
> am not in close
> contact with any of the old phone phreak crowd, but
> I'm sure some
> of you are, and can get the news out.
> Jayson
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From dfroula at sbcglobal.net  Thu Aug  9 14:18:25 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Thu, 9 Aug 2007 12:18:25 -0700 (PDT)
Subject: [VoIP] We've lost a pioneer
In-Reply-To: <000501c7dab2$85ff8da0$6700a8c0@bluegrasspals.com>
Message-ID: <337509.38794.qm@web83207.mail.mud.yahoo.com>

Here's a link to a Discovery Channel video of Joe
whistling away. This is in the days when long distance
trunks with no DC continuity used 2600 pulses not only
for supervision (on hook/off hook), but also to repeat
the dial pulses. Later, 2600 was used only for
supervision and the Blue Box MF digits for dialing.
You can hear the MF of the next tandem link after Joe
pulse dials with 2600.

Joe is at 4 minutes:40 seconds into the show. Highly
recommended viewing. It has many of the Esquire
phreaks. Captain Crunch is even shown making blue box
calls with Steve Wozniak from a pay phone (simulated
for the camera, I'm sure!).

http://www.phreakvids.com/TLC_Hackers_Computer_Outlaws.avi

Don

--- Jayson Smith <ratguy at insightbb.com> wrote:

> Joe Engressia, A.K.A. Joe the Whistler, A.K.A.
> Joybubbles passed
> away in his apartment approximately two days ago. He
> was found by
> management personnel after a phone friend grew
> concerned when he
> hadn't heard from him in several days. His family in
> Florida was
> notified. He was fifty-eight years old.
>      As most of you know, his ordeals with the phone
> company over
> the blue box issue were noted in the October 1971
> Esquire Magazine
> article
> "Secrets of the Little Blue Box." Recent interests
> included
> humorous telephone postings called Phlogs on
> voicemail services.
> He wrote Telephiles for Playback Magazine. His
> interests also
> included children's literature and paraphernalia..
>      Although I never knew Joe personally, I did
> talk to him on the
> Wawina, Minnesota conference line which was set up
> on June 15,
> 2006, when Wawina cut over from N2 to T1 for its
> toll trunks. He
> also left many messages on a message board which was
> set up in the
> weeks prior to the cutover.
>      Please post this message to any other lists
> where you feel it
> would be appropriate, or send it to individuals. I
> am not in close
> contact with any of the old phone phreak crowd, but
> I'm sure some
> of you are, and can get the news out.
> Jayson
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From ratguy at insightbb.com  Thu Aug  9 14:50:44 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Thu, 9 Aug 2007 15:50:44 -0400
Subject: [VoIP] We've lost a pioneer
References: <388679.11256.qm@web83213.mail.mud.yahoo.com>
Message-ID: <000601c7dabe$934eb9c0$6700a8c0@bluegrasspals.com>

Hi,

     As to where I heard the news, I'm blind, and have blind friends who
were associated with him on a voicemail system.
Also, thanks for that video file! I'll be sure and give it a listen!
Jayson

----- Original Message ----- 
From: "Donald Froula" <dfroula at sbcglobal.net>
To: "Voice Over IP Tandem for Analog Switches" <voip at lists.ckts.info>
Sent: Thursday, August 09, 2007 2:50 PM
Subject: Re: [VoIP] We've lost a pioneer


> I'm very sorry to hear this. I spoke to Joe just a few
> months ago. I was watching a Discovery Channel video
> clip of him a few days ago, "dialing" a phone number
> into an old 2600 trunk by whistling pulses of 2600. He
> was an interesting guy.
>
> Jayson where did you spot the news?
>
> Don
>
> --- Jayson Smith <ratguy at insightbb.com> wrote:
>
> > Joe Engressia, A.K.A. Joe the Whistler, A.K.A.
> > Joybubbles passed
> > away in his apartment approximately two days ago. He
> > was found by
> > management personnel after a phone friend grew
> > concerned when he
> > hadn't heard from him in several days. His family in
> > Florida was
> > notified. He was fifty-eight years old.
> >      As most of you know, his ordeals with the phone
> > company over
> > the blue box issue were noted in the October 1971
> > Esquire Magazine
> > article
> > "Secrets of the Little Blue Box." Recent interests
> > included
> > humorous telephone postings called Phlogs on
> > voicemail services.
> > He wrote Telephiles for Playback Magazine. His
> > interests also
> > included children's literature and paraphernalia..
> >      Although I never knew Joe personally, I did
> > talk to him on the
> > Wawina, Minnesota conference line which was set up
> > on June 15,
> > 2006, when Wawina cut over from N2 to T1 for its
> > toll trunks. He
> > also left many messages on a message board which was
> > set up in the
> > weeks prior to the cutover.
> >      Please post this message to any other lists
> > where you feel it
> > would be appropriate, or send it to individuals. I
> > am not in close
> > contact with any of the old phone phreak crowd, but
> > I'm sure some
> > of you are, and can get the news out.
> > Jayson
> > _______________________________________________
> > VoIP mailing list
> > VoIP at lists.ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
>
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>

From david at josephson.com  Thu Aug  9 15:15:29 2007
From: david at josephson.com (David Josephson)
Date: Thu, 09 Aug 2007 13:15:29 -0700
Subject: [VoIP] Test
In-Reply-To: <46BB1666.408@stromberg-carlson.org>
References: <000001c7da2f$e2a990b0$750a0a0a@hickory.vyger.net>
	<46BB1666.408@stromberg-carlson.org>
Message-ID: <46BB75E1.3010905@josephson.com>

Please stay away from Google and Yahoo lists. Besides the delay and 
noise added by these, I'm really not interested in being part of someone 
else's mega business plan. If you need a machine to host a majordomo or 
mailman list, let me know, I have both and they have been running for 
8-10 years, some in a data center and some here in the office. Hosting 
in a data center works fine but if you already have multiple DSL or T1 
lines for VOIP, then the connectivity for a list server is not a problem.

--
David Josephson

John Novack wrote:
> You might do better with Google lists.
> Yahoo is really overloaded.
>
> Really most of the hosting stuff these days can be done for reasonable 
> amounts by companies with data centers.
> It really doesn't make good economic sense to host something on a 
> basement anymore.
>
> John Novack
>
>
> Greg Blakely wrote:
>   
>> Okay, so I broke the list again....  What else is new, right?  I'm about 1/2
>> a second away from putting the sucker up on Yahoo.
>>  
>> I sent this to voip at lists.ckts.info rather than plain old ckts.info.
>>  
>> Let's see if I have any success.


From rdekema at gmail.com  Thu Aug  9 16:52:22 2007
From: rdekema at gmail.com (Rusty Dekema)
Date: Thu, 9 Aug 2007 17:52:22 -0400
Subject: [VoIP] Test
In-Reply-To: <46BB75E1.3010905@josephson.com>
References: <000001c7da2f$e2a990b0$750a0a0a@hickory.vyger.net>
	<46BB1666.408@stromberg-carlson.org> <46BB75E1.3010905@josephson.com>
Message-ID: <68171c120708091452sad1257dieabcc9d18f4e7a47@mail.gmail.com>

On 8/9/07, David Josephson <david at josephson.com> wrote:
> Please stay away from Google and Yahoo lists. Besides the delay and
> noise added by these, I'm really not interested in being part of someone
> else's mega business plan.

Agreed. I really hate Yahoo lists, groups, or whatever they call them.
The last time I had occasion to use one, it was a complete pain in the
rear. I don't have anything specific against Google groups, although I
am sympathetic to the "someone else's megabusiness" issue.

Rusty

From Lucky225 at 2600.com  Thu Aug  9 22:14:33 2007
From: Lucky225 at 2600.com (Lucky 225)
Date: Thu, 9 Aug 2007 20:14:33 -0700
Subject: [VoIP] We've lost a pioneer
In-Reply-To: <000601c7dabe$934eb9c0$6700a8c0@bluegrasspals.com>
References: <388679.11256.qm@web83213.mail.mud.yahoo.com>
	<000601c7dabe$934eb9c0$6700a8c0@bluegrasspals.com>
Message-ID: <f9f99fbe0708092014j3e7ffaf9h5269f55797780f64@mail.gmail.com>

!!!!! I was going to call him last week but decided to hold out til I had
more time and could gather some other phone phreaks up, now I feel really
bad.  I used to talk to him from time to time, he was a great guy, my
condolunces go out to him and his family.  I hope his adoptive mom in Aus.
is alright :(

On 8/9/07, Jayson Smith <ratguy at insightbb.com> wrote:
>
> Hi,
>
>     As to where I heard the news, I'm blind, and have blind friends who
> were associated with him on a voicemail system.
> Also, thanks for that video file! I'll be sure and give it a listen!
> Jayson
>
> ----- Original Message -----
> From: "Donald Froula" <dfroula at sbcglobal.net>
> To: "Voice Over IP Tandem for Analog Switches" <voip at lists.ckts.info>
> Sent: Thursday, August 09, 2007 2:50 PM
> Subject: Re: [VoIP] We've lost a pioneer
>
>
> > I'm very sorry to hear this. I spoke to Joe just a few
> > months ago. I was watching a Discovery Channel video
> > clip of him a few days ago, "dialing" a phone number
> > into an old 2600 trunk by whistling pulses of 2600. He
> > was an interesting guy.
> >
> > Jayson where did you spot the news?
> >
> > Don
> >
> > --- Jayson Smith <ratguy at insightbb.com> wrote:
> >
> > > Joe Engressia, A.K.A. Joe the Whistler, A.K.A.
> > > Joybubbles passed
> > > away in his apartment approximately two days ago. He
> > > was found by
> > > management personnel after a phone friend grew
> > > concerned when he
> > > hadn't heard from him in several days. His family in
> > > Florida was
> > > notified. He was fifty-eight years old.
> > >      As most of you know, his ordeals with the phone
> > > company over
> > > the blue box issue were noted in the October 1971
> > > Esquire Magazine
> > > article
> > > "Secrets of the Little Blue Box." Recent interests
> > > included
> > > humorous telephone postings called Phlogs on
> > > voicemail services.
> > > He wrote Telephiles for Playback Magazine. His
> > > interests also
> > > included children's literature and paraphernalia..
> > >      Although I never knew Joe personally, I did
> > > talk to him on the
> > > Wawina, Minnesota conference line which was set up
> > > on June 15,
> > > 2006, when Wawina cut over from N2 to T1 for its
> > > toll trunks. He
> > > also left many messages on a message board which was
> > > set up in the
> > > weeks prior to the cutover.
> > >      Please post this message to any other lists
> > > where you feel it
> > > would be appropriate, or send it to individuals. I
> > > am not in close
> > > contact with any of the old phone phreak crowd, but
> > > I'm sure some
> > > of you are, and can get the news out.
> > > Jayson
> > > _______________________________________________
> > > VoIP mailing list
> > > VoIP at lists.ckts.info
> > > http://lists.ckts.info/mailman/listinfo/voip
> > > Project Web Page: http://www.ckts.info/
> > >
> >
> > _______________________________________________
> > VoIP mailing list
> > VoIP at lists.ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>

From ratguy at insightbb.com  Thu Aug  9 22:37:12 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Thu, 9 Aug 2007 23:37:12 -0400
Subject: [VoIP] We've lost a pioneer
References: <388679.11256.qm@web83213.mail.mud.yahoo.com>
	<000601c7dabe$934eb9c0$6700a8c0@bluegrasspals.com>
	<f9f99fbe0708092014j3e7ffaf9h5269f55797780f64@mail.gmail.com>
Message-ID: <000401c7daff$bd5aab20$6700a8c0@bluegrasspals.com>

Hi,
I was looking tonight, and I do still have a fairly good recording of the
time Joybubbles spent with us on the Wawina conference just after noon EDT
on June 15. Unfortunately, two things detract from my recording. First, in
the middle of his appearance, the connection to the conference from which I
was making the recording was terminated, so I missed a bit of it. Secondly,
in order to insure that my recording connection was live, I had it routed to
a small speaker. Whenever my handset wasn't muted, you can sometimes hear
the echo from the speaker, especially when I'm talking. This recording does
have some nice N2 background noise, from just about one and a half hours
before the N2 would be gone forever. If Shane still has the conference
recordings, I'm sure he has an even better recording.
Jayson

----- Original Message ----- 
From: "Lucky 225" <Lucky225 at 2600.com>
To: "Voice Over IP Tandem for Analog Switches" <voip at lists.ckts.info>
Sent: Thursday, August 09, 2007 11:14 PM
Subject: Re: [VoIP] We've lost a pioneer


> !!!!! I was going to call him last week but decided to hold out til I had
> more time and could gather some other phone phreaks up, now I feel really
> bad.  I used to talk to him from time to time, he was a great guy, my
> condolunces go out to him and his family.  I hope his adoptive mom in Aus.
> is alright :(
>
> On 8/9/07, Jayson Smith <ratguy at insightbb.com> wrote:
> >
> > Hi,
> >
> >     As to where I heard the news, I'm blind, and have blind friends who
> > were associated with him on a voicemail system.
> > Also, thanks for that video file! I'll be sure and give it a listen!
> > Jayson
> >
> > ----- Original Message -----
> > From: "Donald Froula" <dfroula at sbcglobal.net>
> > To: "Voice Over IP Tandem for Analog Switches" <voip at lists.ckts.info>
> > Sent: Thursday, August 09, 2007 2:50 PM
> > Subject: Re: [VoIP] We've lost a pioneer
> >
> >
> > > I'm very sorry to hear this. I spoke to Joe just a few
> > > months ago. I was watching a Discovery Channel video
> > > clip of him a few days ago, "dialing" a phone number
> > > into an old 2600 trunk by whistling pulses of 2600. He
> > > was an interesting guy.
> > >
> > > Jayson where did you spot the news?
> > >
> > > Don
> > >
> > > --- Jayson Smith <ratguy at insightbb.com> wrote:
> > >
> > > > Joe Engressia, A.K.A. Joe the Whistler, A.K.A.
> > > > Joybubbles passed
> > > > away in his apartment approximately two days ago. He
> > > > was found by
> > > > management personnel after a phone friend grew
> > > > concerned when he
> > > > hadn't heard from him in several days. His family in
> > > > Florida was
> > > > notified. He was fifty-eight years old.
> > > >      As most of you know, his ordeals with the phone
> > > > company over
> > > > the blue box issue were noted in the October 1971
> > > > Esquire Magazine
> > > > article
> > > > "Secrets of the Little Blue Box." Recent interests
> > > > included
> > > > humorous telephone postings called Phlogs on
> > > > voicemail services.
> > > > He wrote Telephiles for Playback Magazine. His
> > > > interests also
> > > > included children's literature and paraphernalia..
> > > >      Although I never knew Joe personally, I did
> > > > talk to him on the
> > > > Wawina, Minnesota conference line which was set up
> > > > on June 15,
> > > > 2006, when Wawina cut over from N2 to T1 for its
> > > > toll trunks. He
> > > > also left many messages on a message board which was
> > > > set up in the
> > > > weeks prior to the cutover.
> > > >      Please post this message to any other lists
> > > > where you feel it
> > > > would be appropriate, or send it to individuals. I
> > > > am not in close
> > > > contact with any of the old phone phreak crowd, but
> > > > I'm sure some
> > > > of you are, and can get the news out.
> > > > Jayson
> > > > _______________________________________________
> > > > VoIP mailing list
> > > > VoIP at lists.ckts.info
> > > > http://lists.ckts.info/mailman/listinfo/voip
> > > > Project Web Page: http://www.ckts.info/
> > > >
> > >
> > > _______________________________________________
> > > VoIP mailing list
> > > VoIP at lists.ckts.info
> > > http://lists.ckts.info/mailman/listinfo/voip
> > > Project Web Page: http://www.ckts.info/
> > >
> > _______________________________________________
> > VoIP mailing list
> > VoIP at lists.ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> >
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/

From stfkerman at jps.net  Thu Aug  9 22:54:01 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Thu, 09 Aug 2007 23:54:01 -0400
Subject: [VoIP] We've lost a pioneer
In-Reply-To: <000501c7dab2$85ff8da0$6700a8c0@bluegrasspals.com>
References: <000501c7dab2$85ff8da0$6700a8c0@bluegrasspals.com>
Message-ID: <46BBE159.1@jps.net>

Is it known whether he was in good and stable health or declining health?

Steph

Jayson Smith wrote:
> Joe Engressia, A.K.A. Joe the Whistler, A.K.A. Joybubbles passed
> away in his apartment approximately two days ago. He was found by
> management personnel after a phone friend grew concerned when he
> hadn't heard from him in several days. His family in Florida was
> notified. He was fifty-eight years old.
>      As most of you know, his ordeals with the phone company over
> the blue box issue were noted in the October 1971 Esquire Magazine
> article
> "Secrets of the Little Blue Box." Recent interests included
> humorous telephone postings called Phlogs on voicemail services.
> He wrote Telephiles for Playback Magazine. His interests also
> included children's literature and paraphernalia..
>      Although I never knew Joe personally, I did talk to him on the
> Wawina, Minnesota conference line which was set up on June 15,
> 2006, when Wawina cut over from N2 to T1 for its toll trunks. He
> also left many messages on a message board which was set up in the
> weeks prior to the cutover.
>      Please post this message to any other lists where you feel it
> would be appropriate, or send it to individuals. I am not in close
> contact with any of the old phone phreak crowd, but I'm sure some
> of you are, and can get the news out.
> Jayson
>
>   

From ratguy at insightbb.com  Thu Aug  9 23:09:57 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Fri, 10 Aug 2007 00:09:57 -0400
Subject: [VoIP] A rather strange request... Old analog cable boxes
Message-ID: <000701c7db04$50450080$6700a8c0@bluegrasspals.com>

Hi,
     We recently got cable TV service, and we're looking for some simple
cable boxes, to avoid having to buy cable-ready TV's just before analog
on-air TV becomes a thing of the past. I've looked on eBay, but I'm not
finding much. What I'm looking for are old analog boxes, maybe even operated
with a dial, a slider, or mechanical pushbuttons. In other words, way too
old to have any sort of activation/authorization stuff. Does anybody have
anything like that they're wanting to get rid of, or know of somewhere I
might find it?
     On another subject, on my recent eBay search, I uncovered something
that sounds just a bit too good to be true. It's a software program by
Orbatel. Supposedly, you pay only once for the software, then after that,
using nothing but a PC and an Internet connection, you can watch and listen
to thousands of satellite TV and radio stations, and not have to pay
anything except the initial software cost. Now I know there's got to be a
catch, but they certainly aren't saying what it is. Anyone know? I mean,
there's either a big catch they don't want you to know about, or this is the
best-kept secret in the consumer TV industry.
Jayson

From rdsmit at hotmail.com  Thu Aug  9 23:38:14 2007
From: rdsmit at hotmail.com (R. Snyder)
Date: Fri, 10 Aug 2007 04:38:14 +0000
Subject: [VoIP] A rather strange request... Old analog cable boxes
In-Reply-To: <000701c7db04$50450080$6700a8c0@bluegrasspals.com>
Message-ID: <BAY122-F17D3C50937F4B73BA47F0EDAE10@phx.gbl>

Jayson,
I've got an old Radio Shack TV block converter you can have if it will work 
for you.  It's just a fixed frequency local oscillator and a mixer that 
moves a whole block of CATV down to broadcast TV frequencies.  I think it 
has an edge-mounted pot on top to tweak the LO frequency for fine tuning.

By the way, you mentioned that you're blind.  I'm developing a mobility aid 
for the blind, and if you'd be willing to give me some feedback on how well 
it works or doesn't work, I could build a prototype for you and send it to 
you.  I've sent a prototype to a friend of mine who is blind, but he hasn't 
gotten out of his apartment in the last few weeks to try it in unfamiliar 
surroundings.  I figure it should take a couple of days to a couple of weeks 
to get used to using it, but that presumes a reasonable amount of time spent 
in unfamiliar surroundings.  Please let me know if you're interested.  Also 
let me know if you'd like the Radio Shack TV block converter.
-Ross

>From: "Jayson Smith" <ratguy at insightbb.com>
>Reply-To: Voice Over IP Tandem for Analog Switches <voip at lists.ckts.info>
>To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
>Subject: [VoIP] A rather strange request... Old analog cable boxes
>Date: Fri, 10 Aug 2007 00:09:57 -0400
>
>Hi,
>      We recently got cable TV service, and we're looking for some simple
>cable boxes, to avoid having to buy cable-ready TV's just before analog
>on-air TV becomes a thing of the past. I've looked on eBay, but I'm not
>finding much. What I'm looking for are old analog boxes, maybe even 
>operated
>with a dial, a slider, or mechanical pushbuttons. In other words, way too
>old to have any sort of activation/authorization stuff. Does anybody have
>anything like that they're wanting to get rid of, or know of somewhere I
>might find it?
>      On another subject, on my recent eBay search, I uncovered something
>that sounds just a bit too good to be true. It's a software program by
>Orbatel. Supposedly, you pay only once for the software, then after that,
>using nothing but a PC and an Internet connection, you can watch and listen
>to thousands of satellite TV and radio stations, and not have to pay
>anything except the initial software cost. Now I know there's got to be a
>catch, but they certainly aren't saying what it is. Anyone know? I mean,
>there's either a big catch they don't want you to know about, or this is 
>the
>best-kept secret in the consumer TV industry.
>Jayson
>_______________________________________________
>VoIP mailing list
>VoIP at lists.ckts.info
>http://lists.ckts.info/mailman/listinfo/voip
>Project Web Page: http://www.ckts.info/

_________________________________________________________________
Tease your brain--play Clink! Win cool prizes! 
http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2


From Lucky225 at 2600.com  Fri Aug 10 00:09:55 2007
From: Lucky225 at 2600.com (Lucky 225)
Date: Thu, 9 Aug 2007 22:09:55 -0700
Subject: [VoIP] We've lost a pioneer
In-Reply-To: <46BBE159.1@jps.net>
References: <000501c7dab2$85ff8da0$6700a8c0@bluegrasspals.com>
	<46BBE159.1@jps.net>
Message-ID: <f9f99fbe0708092209p40773862l820d56174fe69710@mail.gmail.com>

he was sick

On 8/9/07, Steph Kerman <stfkerman at jps.net> wrote:
>
> Is it known whether he was in good and stable health or declining health?
>
> Steph
>
> Jayson Smith wrote:
> > Joe Engressia, A.K.A. Joe the Whistler, A.K.A. Joybubbles passed
> > away in his apartment approximately two days ago. He was found by
> > management personnel after a phone friend grew concerned when he
> > hadn't heard from him in several days. His family in Florida was
> > notified. He was fifty-eight years old.
> >      As most of you know, his ordeals with the phone company over
> > the blue box issue were noted in the October 1971 Esquire Magazine
> > article
> > "Secrets of the Little Blue Box." Recent interests included
> > humorous telephone postings called Phlogs on voicemail services.
> > He wrote Telephiles for Playback Magazine. His interests also
> > included children's literature and paraphernalia..
> >      Although I never knew Joe personally, I did talk to him on the
> > Wawina, Minnesota conference line which was set up on June 15,
> > 2006, when Wawina cut over from N2 to T1 for its toll trunks. He
> > also left many messages on a message board which was set up in the
> > weeks prior to the cutover.
> >      Please post this message to any other lists where you feel it
> > would be appropriate, or send it to individuals. I am not in close
> > contact with any of the old phone phreak crowd, but I'm sure some
> > of you are, and can get the news out.
> > Jayson
> >
> >
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>

From ratguy at insightbb.com  Fri Aug 10 00:21:42 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Fri, 10 Aug 2007 01:21:42 -0400
Subject: [VoIP] A rather strange request... Old analog cable boxes
References: <BAY122-F17D3C50937F4B73BA47F0EDAE10@phx.gbl>
Message-ID: <000601c7db0e$564906c0$6700a8c0@bluegrasspals.com>

Hi,

     I'd like more info about the converter. Is it adjustable what block of
channels it puts where, or is it preset and can't be changed? Does it have
two standard cable connectors for its input and its output, or does it have
an antenna for the output, or...
     About the mobility aid, I don't do that much traveling, but I'm just
curious, what are you developing? I'm always interested in knowing about new
stuff for the blind. We've recently gotten into the Braille+ from Levelstar
and APH, a very nice product.
Jayson

----- Original Message ----- 
From: "R. Snyder" <rdsmit at hotmail.com>
To: <voip at lists.ckts.info>
Sent: Friday, August 10, 2007 12:38 AM
Subject: Re: [VoIP] A rather strange request... Old analog cable boxes


> Jayson,
> I've got an old Radio Shack TV block converter you can have if it will
work
> for you.  It's just a fixed frequency local oscillator and a mixer that
> moves a whole block of CATV down to broadcast TV frequencies.  I think it
> has an edge-mounted pot on top to tweak the LO frequency for fine tuning.
>
> By the way, you mentioned that you're blind.  I'm developing a mobility
aid
> for the blind, and if you'd be willing to give me some feedback on how
well
> it works or doesn't work, I could build a prototype for you and send it to
> you.  I've sent a prototype to a friend of mine who is blind, but he
hasn't
> gotten out of his apartment in the last few weeks to try it in unfamiliar
> surroundings.  I figure it should take a couple of days to a couple of
weeks
> to get used to using it, but that presumes a reasonable amount of time
spent
> in unfamiliar surroundings.  Please let me know if you're interested.
Also
> let me know if you'd like the Radio Shack TV block converter.
> -Ross
>
> >From: "Jayson Smith" <ratguy at insightbb.com>
> >Reply-To: Voice Over IP Tandem for Analog Switches <voip at lists.ckts.info>
> >To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> >Subject: [VoIP] A rather strange request... Old analog cable boxes
> >Date: Fri, 10 Aug 2007 00:09:57 -0400
> >
> >Hi,
> >      We recently got cable TV service, and we're looking for some simple
> >cable boxes, to avoid having to buy cable-ready TV's just before analog
> >on-air TV becomes a thing of the past. I've looked on eBay, but I'm not
> >finding much. What I'm looking for are old analog boxes, maybe even
> >operated
> >with a dial, a slider, or mechanical pushbuttons. In other words, way too
> >old to have any sort of activation/authorization stuff. Does anybody have
> >anything like that they're wanting to get rid of, or know of somewhere I
> >might find it?
> >      On another subject, on my recent eBay search, I uncovered something
> >that sounds just a bit too good to be true. It's a software program by
> >Orbatel. Supposedly, you pay only once for the software, then after that,
> >using nothing but a PC and an Internet connection, you can watch and
listen
> >to thousands of satellite TV and radio stations, and not have to pay
> >anything except the initial software cost. Now I know there's got to be a
> >catch, but they certainly aren't saying what it is. Anyone know? I mean,
> >there's either a big catch they don't want you to know about, or this is
> >the
> >best-kept secret in the consumer TV industry.
> >Jayson
> >_______________________________________________
> >VoIP mailing list
> >VoIP at lists.ckts.info
> >http://lists.ckts.info/mailman/listinfo/voip
> >Project Web Page: http://www.ckts.info/
>
> _________________________________________________________________
> Tease your brain--play Clink! Win cool prizes!
> http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2
>
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>

From Lucky225 at 2600.com  Fri Aug 10 00:32:35 2007
From: Lucky225 at 2600.com (Lucky 225)
Date: Thu, 9 Aug 2007 22:32:35 -0700
Subject: [VoIP] A rather strange request... Old analog cable boxes
In-Reply-To: <000601c7db0e$564906c0$6700a8c0@bluegrasspals.com>
References: <BAY122-F17D3C50937F4B73BA47F0EDAE10@phx.gbl>
	<000601c7db0e$564906c0$6700a8c0@bluegrasspals.com>
Message-ID: <f9f99fbe0708092232h106a167at4a77985fe87cf041@mail.gmail.com>

radioshack sells 'em I think

On 8/9/07, Jayson Smith <ratguy at insightbb.com> wrote:
>
> Hi,
>
>     I'd like more info about the converter. Is it adjustable what block of
> channels it puts where, or is it preset and can't be changed? Does it have
> two standard cable connectors for its input and its output, or does it
> have
> an antenna for the output, or...
>     About the mobility aid, I don't do that much traveling, but I'm just
> curious, what are you developing? I'm always interested in knowing about
> new
> stuff for the blind. We've recently gotten into the Braille+ from
> Levelstar
> and APH, a very nice product.
> Jayson
>
> ----- Original Message -----
> From: "R. Snyder" <rdsmit at hotmail.com>
> To: <voip at lists.ckts.info>
> Sent: Friday, August 10, 2007 12:38 AM
> Subject: Re: [VoIP] A rather strange request... Old analog cable boxes
>
>
> > Jayson,
> > I've got an old Radio Shack TV block converter you can have if it will
> work
> > for you.  It's just a fixed frequency local oscillator and a mixer that
> > moves a whole block of CATV down to broadcast TV frequencies.  I think
> it
> > has an edge-mounted pot on top to tweak the LO frequency for fine
> tuning.
> >
> > By the way, you mentioned that you're blind.  I'm developing a mobility
> aid
> > for the blind, and if you'd be willing to give me some feedback on how
> well
> > it works or doesn't work, I could build a prototype for you and send it
> to
> > you.  I've sent a prototype to a friend of mine who is blind, but he
> hasn't
> > gotten out of his apartment in the last few weeks to try it in
> unfamiliar
> > surroundings.  I figure it should take a couple of days to a couple of
> weeks
> > to get used to using it, but that presumes a reasonable amount of time
> spent
> > in unfamiliar surroundings.  Please let me know if you're interested.
> Also
> > let me know if you'd like the Radio Shack TV block converter.
> > -Ross
> >
> > >From: "Jayson Smith" <ratguy at insightbb.com>
> > >Reply-To: Voice Over IP Tandem for Analog Switches <
> voip at lists.ckts.info>
> > >To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> > >Subject: [VoIP] A rather strange request... Old analog cable boxes
> > >Date: Fri, 10 Aug 2007 00:09:57 -0400
> > >
> > >Hi,
> > >      We recently got cable TV service, and we're looking for some
> simple
> > >cable boxes, to avoid having to buy cable-ready TV's just before analog
> > >on-air TV becomes a thing of the past. I've looked on eBay, but I'm not
> > >finding much. What I'm looking for are old analog boxes, maybe even
> > >operated
> > >with a dial, a slider, or mechanical pushbuttons. In other words, way
> too
> > >old to have any sort of activation/authorization stuff. Does anybody
> have
> > >anything like that they're wanting to get rid of, or know of somewhere
> I
> > >might find it?
> > >      On another subject, on my recent eBay search, I uncovered
> something
> > >that sounds just a bit too good to be true. It's a software program by
> > >Orbatel. Supposedly, you pay only once for the software, then after
> that,
> > >using nothing but a PC and an Internet connection, you can watch and
> listen
> > >to thousands of satellite TV and radio stations, and not have to pay
> > >anything except the initial software cost. Now I know there's got to be
> a
> > >catch, but they certainly aren't saying what it is. Anyone know? I
> mean,
> > >there's either a big catch they don't want you to know about, or this
> is
> > >the
> > >best-kept secret in the consumer TV industry.
> > >Jayson
> > >_______________________________________________
> > >VoIP mailing list
> > >VoIP at lists.ckts.info
> > >http://lists.ckts.info/mailman/listinfo/voip
> > >Project Web Page: http://www.ckts.info/
> >
> > _________________________________________________________________
> > Tease your brain--play Clink! Win cool prizes!
> > http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2
> >
> > _______________________________________________
> > VoIP mailing list
> > VoIP at lists.ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>

From jnovack at stromberg-carlson.org  Fri Aug 10 07:48:13 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Fri, 10 Aug 2007 08:48:13 -0400
Subject: [VoIP] A rather strange request... Old analog cable boxes
In-Reply-To: <f9f99fbe0708092232h106a167at4a77985fe87cf041@mail.gmail.com>
References: <BAY122-F17D3C50937F4B73BA47F0EDAE10@phx.gbl>	<000601c7db0e$564906c0$6700a8c0@bluegrasspals.com>
	<f9f99fbe0708092232h106a167at4a77985fe87cf041@mail.gmail.com>
Message-ID: <46BC5E8D.6020501@stromberg-carlson.org>

The older block converters only convert a few of the now available channels.
Since you haven't related what system or how many channels you have 
available to you, it will be difficult to select a converter.
There are also several cable system assignment methods in use, or that 
have been in use in the recent past.
If you aren't aware of the analog off air frequency assignments, 
channels 2-6 are in the low VHF range, and start at 54 Mhz and run, with 
a 4 Mhz gap between Channels 4 and 5, to the FM radio start at 88 Mhz. 
Off air Channel 7 begins much higher at 174 Mhz, with lots of unrelated 
to TV assignments in between. Cable channels 14 up are assigned in 
between beginning at about 108 Mhz. Early cable systems had problems 
with the FCC and radiation in the aircraft assignments, and some block 
converters didn't cover this range at all.
In addition, there are different carrier assignments depending on the 
system.
You may want to first contact your cable company to determine if they 
have any low cost/free offerings. I know that in many areas Comcast is 
offering one free converter to all subscribers due to moving channel 
assignments to digital.

John Novack


Lucky 225 wrote:
> radioshack sells 'em I think
>
> On 8/9/07, Jayson Smith <ratguy at insightbb.com> wrote:
>   
>> Hi,
>>
>>     I'd like more info about the converter. Is it adjustable what block of
>> channels it puts where, or is it preset and can't be changed? Does it have
>> two standard cable connectors for its input and its output, or does it
>> have
>> an antenna for the output, or...
>>     About the mobility aid, I don't do that much traveling, but I'm just
>> curious, what are you developing? I'm always interested in knowing about
>> new
>> stuff for the blind. We've recently gotten into the Braille+ from
>> Levelstar
>> and APH, a very nice product.
>> Jayson
>>
>> ----- Original Message -----
>> From: "R. Snyder" <rdsmit at hotmail.com>
>> To: <voip at lists.ckts.info>
>> Sent: Friday, August 10, 2007 12:38 AM
>> Subject: Re: [VoIP] A rather strange request... Old analog cable boxes
>>
>>
>>     
>>> Jayson,
>>> I've got an old Radio Shack TV block converter you can have if it will
>>>       
>> work
>>     
>>> for you.  It's just a fixed frequency local oscillator and a mixer that
>>> moves a whole block of CATV down to broadcast TV frequencies.  I think
>>>       
>> it
>>     
>>> has an edge-mounted pot on top to tweak the LO frequency for fine
>>>       
>> tuning.
>>     
>>> By the way, you mentioned that you're blind.  I'm developing a mobility
>>>       
>> aid
>>     
>>> for the blind, and if you'd be willing to give me some feedback on how
>>>       
>> well
>>     
>>> it works or doesn't work, I could build a prototype for you and send it
>>>       
>> to
>>     
>>> you.  I've sent a prototype to a friend of mine who is blind, but he
>>>       
>> hasn't
>>     
>>> gotten out of his apartment in the last few weeks to try it in
>>>       
>> unfamiliar
>>     
>>> surroundings.  I figure it should take a couple of days to a couple of
>>>       
>> weeks
>>     
>>> to get used to using it, but that presumes a reasonable amount of time
>>>       
>> spent
>>     
>>> in unfamiliar surroundings.  Please let me know if you're interested.
>>>       
>> Also
>>     
>>> let me know if you'd like the Radio Shack TV block converter.
>>> -Ross
>>>
>>>       
>>>> From: "Jayson Smith" <ratguy at insightbb.com>
>>>> Reply-To: Voice Over IP Tandem for Analog Switches <
>>>>         
>> voip at lists.ckts.info>
>>     
>>>> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
>>>> Subject: [VoIP] A rather strange request... Old analog cable boxes
>>>> Date: Fri, 10 Aug 2007 00:09:57 -0400
>>>>
>>>> Hi,
>>>>      We recently got cable TV service, and we're looking for some
>>>>         
>> simple
>>     
>>>> cable boxes, to avoid having to buy cable-ready TV's just before analog
>>>> on-air TV becomes a thing of the past. I've looked on eBay, but I'm not
>>>> finding much. What I'm looking for are old analog boxes, maybe even
>>>> operated
>>>> with a dial, a slider, or mechanical pushbuttons. In other words, way
>>>>         
>> too
>>     
>>>> old to have any sort of activation/authorization stuff. Does anybody
>>>>         
>> have
>>     
>>>> anything like that they're wanting to get rid of, or know of somewhere
>>>>         
>> I
>>     
>>>> might find it?
>>>>      On another subject, on my recent eBay search, I uncovered
>>>>         
>> something
>>     
>>>> that sounds just a bit too good to be true. It's a software program by
>>>> Orbatel. Supposedly, you pay only once for the software, then after
>>>>         
>> that,
>>     
>>>> using nothing but a PC and an Internet connection, you can watch and
>>>>         
>> listen
>>     
>>>> to thousands of satellite TV and radio stations, and not have to pay
>>>> anything except the initial software cost. Now I know there's got to be
>>>>         
>> a
>>     
>>>> catch, but they certainly aren't saying what it is. Anyone know? I
>>>>         
>> mean,
>>     
>>>> there's either a big catch they don't want you to know about, or this
>>>>         
>> is
>>     
>>>> the
>>>> best-kept secret in the consumer TV industry.
>>>> Jayson
>>>> _______________________________________________
>>>> VoIP mailing list
>>>> VoIP at lists.ckts.info
>>>> http://lists.ckts.info/mailman/listinfo/voip
>>>> Project Web Page: http://www.ckts.info/
>>>>         
>>> _________________________________________________________________
>>> Tease your brain--play Clink! Win cool prizes!
>>> http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2
>>>
>>> _______________________________________________
>>> VoIP mailing list
>>> VoIP at lists.ckts.info
>>> http://lists.ckts.info/mailman/listinfo/voip
>>> Project Web Page: http://www.ckts.info/
>>>
>>>       
>> _______________________________________________
>> VoIP mailing list
>> VoIP at lists.ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>
>>     
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From kxt at fubegra.net  Sat Aug 11 11:16:29 2007
From: kxt at fubegra.net (Russ Price)
Date: Sat, 11 Aug 2007 11:16:29 -0500
Subject: [VoIP] Patched 1.2.24 tarball available
Message-ID: <46BDE0DD.5070209@fubegra.net>

I've put my patches into place for Asterisk 1.2.24 and have made both 
the patch and the tarball available on <http://cnet.fubegra.net/>.

	Russ

From ian at uax.org.uk  Mon Aug 13 10:15:40 2007
From: ian at uax.org.uk (Ian Jolly)
Date: Mon, 13 Aug 2007 16:15:40 +0100
Subject: [VoIP] Unknown problem ?
Message-ID: <01f101c7ddbc$cf966a80$0a01a8c0@acer1dd0bbc6d0>

Hi all,

Help !!!!!   I keep getting error messages as below (just a view of some of them)

Aug 13 16:25:35 NOTICE[2368]: chan_sip.c:11328 sip_poke_noanswer: Peer 'sipgate2' is now UNREACHABLE!  Last qualify: 60
Aug 13 16:25:35 NOTICE[2368]: chan_sip.c:11328 sip_poke_noanswer: Peer 'sipgate' is now UNREACHABLE!  Last qualify: 72
Aug 13 16:25:45 NOTICE[2368]: chan_sip.c:9689 handle_response_peerpoke: Peer 'sipgate2' is now REACHABLE! (100ms / 2000ms)
Aug 13 16:25:45 NOTICE[2368]: chan_sip.c:9689 handle_response_peerpoke: Peer 'sipgate' is now REACHABLE! (113ms / 2000ms)

Does anyone know exactly what causes these messages?  Various services on my Asterisk - either VoIP providers or telephones hosted off my *box come up with these messages but not all the time . They are always a ten second interval between becoming UNREACHABLE/REACHABLE.  It does not affect the conversation being held to one of those numbers at the time it happens

 Even a couple of IPphones that are on my own network suffer the same problem.  It varies from day to day as to which service/phone that it is happening to.

>From time to time I also get a message/s -
Aug 13 16:31:06 WARNING[2368]: chan_sip.c:9601 handle_response_register: Got 200 OK on REGISTER that isn't a register
Aug 13 16:31:06 WARNING[2368]: chan_sip.c:9601 handle_response_register: Got 200 OK on REGISTER that isn't a register

Again anyone any idea what causes it?  Are they related?

Ian Jolly

From greg at vyger.net  Mon Aug 13 17:36:02 2007
From: greg at vyger.net (Greg Blakely)
Date: Mon, 13 Aug 2007 17:36:02 -0500
Subject: [VoIP] Unknown problem ?
Message-ID: <A691C3685A152E4AAC9608B8D8E11316686400@godzilla.hickory.vyger.net>

If you have either your sip or iax2 clients tagged as anything other
than QUALIFY=NO, then asterisk will check them on a continual basis to
see if they are reachable.  If they're not, then calls won't even be
attempted to them.  

You'll also get notice on the console when reachable/unreachable status
changes. 

> -----Original Message-----
> From: voip-bounces at lists.ckts.info 
> [mailto:voip-bounces at lists.ckts.info] On Behalf Of Ian Jolly
> Sent: Monday, August 13, 2007 10:16 AM
> To: CNET VoIP
> Subject: [VoIP] Unknown problem ?
> 
> Hi all,
> 
> Help !!!!!   I keep getting error messages as below (just a 
> view of some of them)
> 
> Aug 13 16:25:35 NOTICE[2368]: chan_sip.c:11328 
> sip_poke_noanswer: Peer 'sipgate2' is now UNREACHABLE!  Last 
> qualify: 60 Aug 13 16:25:35 NOTICE[2368]: chan_sip.c:11328 
> sip_poke_noanswer: Peer 'sipgate' is now UNREACHABLE!  Last 
> qualify: 72 Aug 13 16:25:45 NOTICE[2368]: chan_sip.c:9689 
> handle_response_peerpoke: Peer 'sipgate2' is now REACHABLE! 
> (100ms / 2000ms) Aug 13 16:25:45 NOTICE[2368]: 
> chan_sip.c:9689 handle_response_peerpoke: Peer 'sipgate' is 
> now REACHABLE! (113ms / 2000ms)
> 
> Does anyone know exactly what causes these messages?  Various 
> services on my Asterisk - either VoIP providers or telephones 
> hosted off my *box come up with these messages but not all 
> the time . They are always a ten second interval between 
> becoming UNREACHABLE/REACHABLE.  It does not affect the 
> conversation being held to one of those numbers at the time it happens
> 
>  Even a couple of IPphones that are on my own network suffer 
> the same problem.  It varies from day to day as to which 
> service/phone that it is happening to.
> 
> >From time to time I also get a message/s -
> Aug 13 16:31:06 WARNING[2368]: chan_sip.c:9601 
> handle_response_register: Got 200 OK on REGISTER that isn't a 
> register Aug 13 16:31:06 WARNING[2368]: chan_sip.c:9601 
> handle_response_register: Got 200 OK on REGISTER that isn't a register
> 
> Again anyone any idea what causes it?  Are they related?
> 
> Ian Jolly
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 

From rdekema at gmail.com  Mon Aug 13 19:55:55 2007
From: rdekema at gmail.com (Rusty Dekema)
Date: Mon, 13 Aug 2007 20:55:55 -0400
Subject: [VoIP] Unknown problem ?
In-Reply-To: <A691C3685A152E4AAC9608B8D8E11316686400@godzilla.hickory.vyger.net>
References: <A691C3685A152E4AAC9608B8D8E11316686400@godzilla.hickory.vyger.net>
Message-ID: <68171c120708131755i8d90fe8t8cacd1c855001fc8@mail.gmail.com>

On 8/13/07, Greg Blakely <greg at vyger.net> wrote:
> If you have either your sip or iax2 clients tagged as anything other
> than QUALIFY=NO, then asterisk will check them on a continual basis to
> see if they are reachable.  If they're not, then calls won't even be
> attempted to them.
>
> You'll also get notice on the console when reachable/unreachable status
> changes.

In my experience running Asterisk 1.2, the qualify feature was
absolutely useless. SIP devices (of all kinds; various softphones,
hardware-based ATAs, etc) would habitually "flap" back and forth from
reachable to unreachable even though they were connected to my
Asterisk server on a completely non-loaded 100mbit local ethernet,
even though the Asterisk server and the PCs running the soft phones
had no other CPU load, and even though I increased the qualify= time
to over 2000ms.

Since these SIP devices otherwise worked perfectly, it was quite clear
that they were not truly "unreachable" for periods approaching
anything like 2 seconds. I don't know what the problem was, and I
don't know whether it still exists in Asterisk 1.4 since, although I
run it, I have not bothered to turn the qualify option back on.

Rusty

From jnovack at stromberg-carlson.org  Tue Aug 14 09:12:18 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Tue, 14 Aug 2007 10:12:18 -0400
Subject: [VoIP] Unknown problem ?
In-Reply-To: <68171c120708131755i8d90fe8t8cacd1c855001fc8@mail.gmail.com>
References: <A691C3685A152E4AAC9608B8D8E11316686400@godzilla.hickory.vyger.net>
	<68171c120708131755i8d90fe8t8cacd1c855001fc8@mail.gmail.com>
Message-ID: <46C1B842.7060503@stromberg-carlson.org>



Rusty Dekema wrote:
> On 8/13/07, Greg Blakely <greg at vyger.net> wrote:
>   
>> If you have either your sip or iax2 clients tagged as anything other than QUALIFY=NO, then asterisk will check them on a continual basis to see if they are reachable.  If they're not, then calls won't even be attempted to them.
>>
>> You'll also get notice on the console when reachable/unreachable status
>> changes.
I don't see these with my install of 1.2, probably because I have 
removed notices from logger.conf

I also don't use qualify, and don't have any issues with SIP devices on 
my LAN.

Perhaps the simple solution is to remove "notice" from the console line 
in logger.conf. Especially if the SIP devices all work.

John Novack


> In my experience running Asterisk 1.2, the qualify feature was
> absolutely useless. SIP devices (of all kinds; various softphones,
> hardware-based ATAs, etc) would habitually "flap" back and forth from
> reachable to unreachable even though they were connected to my
> Asterisk server on a completely non-loaded 100mbit local ethernet,
> even though the Asterisk server and the PCs running the soft phones
> had no other CPU load, and even though I increased the qualify= time
> to over 2000ms.
>
> Since these SIP devices otherwise worked perfectly, it was quite clear
> that they were not truly "unreachable" for periods approaching
> anything like 2 seconds. I don't know what the problem was, and I
> don't know whether it still exists in Asterisk 1.4 since, although I
> run it, I have not bothered to turn the qualify option back on.
>
> Rusty
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From ian at uax.org.uk  Tue Aug 14 10:55:13 2007
From: ian at uax.org.uk (Ian Jolly)
Date: Tue, 14 Aug 2007 16:55:13 +0100
Subject: [VoIP] Unknown problem ?
References: <A691C3685A152E4AAC9608B8D8E11316686400@godzilla.hickory.vyger.net><68171c120708131755i8d90fe8t8cacd1c855001fc8@mail.gmail.com>
	<46C1B842.7060503@stromberg-carlson.org>
Message-ID: <009c01c7de8b$80a95570$0a01a8c0@acer1dd0bbc6d0>

Thanks folks for the comments. Looks as though we just "switch off" the 
qualify statement and the problem goes away :-)

I was just wondering if it was a genuine problem or not.

Ian Jolly
+64 85 32 889
+44 (0)  352      82 26 (via a 1929 GPO Rural Automatic eXchange!) from 
CNET - the Heritage Telephone Network
+44 (0)1352 83 82 26 (via a 1929 GPO Rural Automatic eXchange!) from  Public 
Telephone Network
FWD Telephone No  83 2230

----- Original Message ----- 
From: "John Novack" <jnovack at stromberg-carlson.org>
To: "Voice Over IP Tandem for Analog Switches" <voip at lists.ckts.info>
Sent: Tuesday, August 14, 2007 3:12 PM
Subject: Re: [VoIP] Unknown problem ?


>
>
> Rusty Dekema wrote:
>> On 8/13/07, Greg Blakely <greg at vyger.net> wrote:
>>
>>> If you have either your sip or iax2 clients tagged as anything other 
>>> than QUALIFY=NO, then asterisk will check them on a continual basis to 
>>> see if they are reachable.  If they're not, then calls won't even be 
>>> attempted to them.
>>>
>>> You'll also get notice on the console when reachable/unreachable status
>>> changes.
> I don't see these with my install of 1.2, probably because I have
> removed notices from logger.conf
>
> I also don't use qualify, and don't have any issues with SIP devices on
> my LAN.
>
> Perhaps the simple solution is to remove "notice" from the console line
> in logger.conf. Especially if the SIP devices all work.
>
> John Novack
>
>
>> In my experience running Asterisk 1.2, the qualify feature was
>> absolutely useless. SIP devices (of all kinds; various softphones,
>> hardware-based ATAs, etc) would habitually "flap" back and forth from
>> reachable to unreachable even though they were connected to my
>> Asterisk server on a completely non-loaded 100mbit local ethernet,
>> even though the Asterisk server and the PCs running the soft phones
>> had no other CPU load, and even though I increased the qualify= time
>> to over 2000ms.
>>
>> Since these SIP devices otherwise worked perfectly, it was quite clear
>> that they were not truly "unreachable" for periods approaching
>> anything like 2 seconds. I don't know what the problem was, and I
>> don't know whether it still exists in Asterisk 1.4 since, although I
>> run it, I have not bothered to turn the qualify option back on.
>>
>> Rusty
>> _______________________________________________
>> VoIP mailing list
>> VoIP at lists.ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>
>
> -- 
> Dog is my co-pilot
>
> _______________________________________________
> VoIP mailing list
> VoIP at lists.ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.5.476 / Virus Database: 269.11.17/951 - Release Date: 
> 13/08/2007 10:15
> 


From ratguy at insightbb.com  Tue Aug 14 16:47:51 2007
From: ratguy at insightbb.com (Jayson Smith)
Date: Tue, 14 Aug 2007 17:47:51 -0400
Subject: [VoIP] More information about Joybubbles
Message-ID: <000b01c7debc$c3a16260$6700a8c0@bluegrasspals.com>

Hello,

     Today, I got the number for Joybubbles's mailbox where he posted what
he called phlogs, just little messages about what's been going on in his
life, what he's doing, etc. The last phlog he recorded in this mailbox was
on Saturday, August 4, so he was definitely alive on that date. His voice
sounded normal. He talked about things he was going to do, and was planning
on coming back in a day or two or three to post another phlog, so obviously,
the heart attack or whatever killed him was as much of a surprise to him as
it was to the rest of us.
Jayson

From stfkerman at jps.net  Tue Aug 14 17:04:31 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Tue, 14 Aug 2007 18:04:31 -0400
Subject: [VoIP] More information about Joybubbles
In-Reply-To: <000b01c7debc$c3a16260$6700a8c0@bluegrasspals.com>
References: <000b01c7debc$c3a16260$6700a8c0@bluegrasspals.com>
Message-ID: <46C226EF.3020700@jps.net>

Thanks Jayson.
Steph

Jayson Smith wrote:
> Hello,
>
>      Today, I got the number for Joybubbles's mailbox where he posted what
> he called phlogs, just little messages about what's been going on in his
> life, what he's doing, etc. The last phlog he recorded in this mailbox was
> on Saturday, August 4, so he was definitely alive on that date. His voice
> sounded normal. He talked about things he was going to do, and was planning
> on coming back in a day or two or three to post another phlog, so obviously,
> the heart attack or whatever killed him was as much of a surprise to him as
> it was to the rest of us.
> Jayson
>
>   

From dfroula at sbcglobal.net  Tue Aug 14 22:29:36 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Tue, 14 Aug 2007 20:29:36 -0700 (PDT)
Subject: [VoIP] Joybubbles Interview
In-Reply-To: <46C226EF.3020700@jps.net>
Message-ID: <271780.17839.qm@web83210.mail.mud.yahoo.com>

I added the hour-long interview Emmanuel Goldstein of
2600 magazine did with Joybubbles back in 1991.

1-762-0123 - Part 1
1-762-0124 - Part 2

Also available via blue box at 1-762-2600 or
1-762-2601 with KP+123+ST and KP+124+ST

Don

From dfroula at sbcglobal.net  Tue Aug 14 22:32:13 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Tue, 14 Aug 2007 20:32:13 -0700 (PDT)
Subject: [VoIP] More information about Joybubbles
In-Reply-To: <46C226EF.3020700@jps.net>
Message-ID: <475894.44170.qm@web83214.mail.mud.yahoo.com>

This was posted today on the binrev forums:

Don

=====================================================

Hello Everyone,

This is Steven Gibb a friend of Joybubbles for 25
years and also his personal representative that is
taking care of all his affairs.

First of all thanks for all the great comments and
links. I do have the domain Joybubbles.com and will be
building a site honoring his memory and will need you
help and input doing so. 

I want to clear up a point and then ask for all your
help!

#1. We're not sure what he passed on from as there has
been no official statement from the medical examiner.
However, it is true that he had some extra weight and
was retaining water. I was coaching him regarding his
diet and he was also on a diuretic to take the water
off. If someone does this too long without addressing
the root cause...you can experience congested heart
failure among an assortment of other symptoms. The
landlord also told me that she thought his neck was at
an acute angle depicting a broken neck. The bottom
line is we don't know what really happened and may not
ever know because it wasn't a crime scene and the
medial examiner is basically making an educated guess.
However, I will get as accurate as a diagnoses and
will post in in the website when appropriate.

Now I need your help. For those of you that have been
in touch with Joybubbles since August of 2006, surely
know that he won a spelling bee contest hosted in
Australia. He also had three people (one is a couple)
that were very close to him who live in Sydney and
Melbourne. In fact, one of them supposedly flew over
here about every 6 weeks and visited Joybubbles. I
have these peoples names but am having a very
difficult time locating these people and their contact
information. Some of the other people working in this
mission have told me that these people have
non-published numbers so my question is...how do we
get these numbers or even an address?

If you have any ideas please feel to contact me via
the information below.

You may contact me via e-mail rawfasting at earthlink.net
or 760.917.6200

Warmly,

Steven Gibb 

From hockd at dteenergy.com  Wed Aug 15 04:07:49 2007
From: hockd at dteenergy.com (Dennis D Hock)
Date: Wed, 15 Aug 2007 05:07:49 -0400
Subject: [VoIP] Joybubbles Interview
Message-ID: <OF49B8F187.E1EFDEC3-ON85257338.00322799-85257338.0032279C@dteenergy.com>

Don and Jason,

Jason thankls for posting that additional info on Joy He will be missed.

Don thanks for putting these up on 726.  I have enjoyed listening to all of
themover the past few weekends.  It is appreciated.

Thank you both,

Dennis Hock

-----voip-bounces at lists.ckts.info wrote: -----


To: Voice Over IP Tandem for Analog Switches <voip at lists.ckts.info>
From: Donald Froula <dfroula at sbcglobal.net>
Sent by: voip-bounces at lists.ckts.info
Date: 08/14/2007 11:29PM
Subject: [VoIP] Joybubbles Interview

I added the hour-long interview Emmanuel Goldstein of
2600 magazine did with Joybubbles back in 1991.

1-762-0123 - Part 1
1-762-0124 - Part 2

Also available via blue box at 1-762-2600 or
1-762-2601 with KP+123+ST and KP+124+ST

Don
_______________________________________________
VoIP mailing list
VoIP at lists.ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/

From ka2wft at arrl.net  Wed Aug 15 15:51:42 2007
From: ka2wft at arrl.net (Doug Alderdice)
Date: Wed, 15 Aug 2007 16:51:42 -0400
Subject: [VoIP] Moto VT1005 Settings Update
Message-ID: <5.1.0.14.2.20070815164617.02f9fca8@incoming.verizon.net>

For the benefit of those running the Moto VT1005 ATA, today I discovered a 
setting change that is needed if you are trying to access Asterisk Voice 
Mail via this ATA.  The setting is called "BTIOPT TeleSipOobDtmf" and is 
located in the top dozen or so lines in the sample configuration that some 
of us have been passing around.  In that sample file this setting is set to 
"FALSE."  When set that way, I was not able to access any DTMF choices in 
my * voice mail system.  The ATA was simply passing the tones through, but 
not passing the info digitally to the * box, apparently.  The VT1005 manual 
indicates that the default for this setting is "TRUE."  Don't know how/why 
it's set to FALSE in the sample file.

I changed the "BTIOPT TeleSipOobDtmf" setting to "TRUE" and recompiled and 
sftp'd the new setting file to the ATA and I can now access the voice mail 
options on my *.

FWIW.

YMMV.

Doug.


From lee at spenadel.com  Thu Aug 16 07:17:42 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Thu, 16 Aug 2007 08:17:42 -0400
Subject: [VoIP] PBX Options for Fire Dept.
Message-ID: <010a01c7dfff$729b1890$57d149b0$@com>

Sorry for the cross-post and hence duplicate copies of this email.

 

Some of you know that I have been working with my local Fire Department (I'm
a member) to help spec a new phone system for new Fire Station.  The
architect has brought in a "consulting firm" to generate the specs for the
new system for "free".  However, the consulting firm is also sells PBxs and
installs them (so much for objectivity and independence).  I am concerned
that the spec and RFP will favor this company and be crafted to make it
difficult for others to meet a reasonable spec.  So I've offered to help
review the spec, RFP and other documents for free.

 

The existing Fire Station is co-located with the Police and served by an
Avaya Key System with Centrex trunks.  The building has approximately 35
Stations.  The new Fire Station will be located across the street from the
old station, which will continue to house the Police dept.  There will be
several conduits in the ground between the two buildings for services.
These systems need to be tied together as though they were serving two
departments in one building (I'm sure a T1 or two would suffice here).

 

I expect there to be a need to support 50 - 55  stations with possible
growth to 75 stations. 

 

I was in telecom consulting for seven years back in the late 80's and have
not stayed in touch with product offerings.  This consulting firm has listed
"acceptable manufacturers and models as follows:

 

NEC 2000

Avaya 8700

Nortel Meridian Option 11

Vodavi series

 

I'm looking for opinions from knowledgeable members who can let me know what
other viable options exist.  I want the consulting company to consider these
other systems as acceptible.  This is a small department in a small town and
does not need a PBX sized to 300 lines. 

 

Opinions are welcome and thanks in advance.

 

Lee

 

 

 

 

 

 

 

 

 

 

If you are at the North Pole, what time is it?

 


From jnovack at stromberg-carlson.org  Thu Aug 16 07:33:36 2007
From: jnovack at stromberg-carlson.org (John Nov