From ikj1234i at yahoo.com  Sun Apr  1 10:07:01 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Sun, 1 Apr 2007 08:07:01 -0700 (PDT)
Subject: [VoIP] SxS Call Progress Sounds
In-Reply-To: <00f501c773e0$8610e070$9232a150$@com>
Message-ID: <20070401150701.42195.qmail@web51612.mail.re2.yahoo.com>


This seems to be working fairly nicely now.  Switching
from an X100P-FXO to a channel-bank FXO has *vastly*
improved the experience, because the dial tone is
still perfectly audible throughout the *entire* first
digit, including the onhook portions of the outpulsing
cycle.  There is no missing audio in the entire
sequence, especially now you can hear the "clunk" as
the selector cuts through to the connector!  

Too bad the "2600" SF pulsing stuff isn't included.  I
once lived in a No. 5 Crossbar that had SF trunks to a
number of outlying SxS offices and spent a lot of time
with a homemade 2600 keying unit dialing through
them...  If the SF stuff were present, it would be
possible to dial through the machine remotely, *in
real time*.

Not sure exactly what to attribute the "irregular"
pulsing.  The proper way to isolate this will be to
back out the patches and listen with a butt set to the
outpulsing, see if the removal of the patch makes the
pulsing more regular or not.

John Novack - your XY office returns a fast-busy to
the caller during the outpulsing of the digit into its
first selector.  How difficult would it be to connect
up your channel-bank FXO to it???

Finally, this experience has made me *much* less
inclined to add anything to attenuate the audio during
the digit outpulsing!  It sounds too good as it is! 
;-)

Max

--- Lee Spenadel <lee at spenadel.com> wrote:

> Thanks to Russ's extra eyes, the call progress
> sounds are now available on
> my switch.  Please give a listen.
> 
>  
> 
> Lee
> 
>  
> 
>  
> 
>  
> 
>  
> 
> If your car could travel at the speed of light,
> would your headlights work?
> 
>  
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 



 
____________________________________________________________________________________
8:00? 8:25? 8:40? Find a flick in no time 
with the Yahoo! Search movie showtime shortcut.
http://tools.search.yahoo.com/shortcuts/#news

From jnovack at stromberg-carlson.org  Sun Apr  1 10:30:34 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 01 Apr 2007 11:30:34 -0400
Subject: [VoIP] Patches
In-Reply-To: <007e01c773e3$a5a51b60$0a01a8c0@acer1dd0bbc6d0>
References: <460ED063.2070103@stromberg-carlson.org>
	<007e01c773e3$a5a51b60$0a01a8c0@acer1dd0bbc6d0>
Message-ID: <460FD01A.8030400@stromberg-carlson.org>



Ian Jolly wrote:
> I couldn't agree more with you John!!   We've been putting similar 
> information on the CNET-UK-I website for beginners. But you guys are 
> way ahead of us in understanding both Linux and Asterisk.
>
> I'm sat here with a new PC with serial SATA hard drive that will not 
> accept the old "UK" combination of Fedora Core 4 and Asterisk 1.2.x 
> (fairly early but it is a working combination!).  I've got a Sanogoma 
> A200 with 4FXS/4FXO. The more I read the more confusing it gets :-)  
> There seems to be a complete 'Linux' language with little explanation 
> of it.
>
> Fedora is up to Fedora 7 and as you say, Asterisk is up to 1.2.17 and 
> I believe that Asterisk 1.4.x is said to be out of Beta and a Stable 
> edition now.
>
> In the US, you seem to use Centos.
I use it because it is easy to install and seems to give fewer problems.
There also seems to be some issues with Fedora and updates.
Many users on the asterisk-users list report success with Fedora and 
many others report problems
CentOS is the free version of RHEL, and seems to have a smoother upgrade 
path.

Asterisk 1.4 Seems to require CentOS 4.x so keep that in mind if/when 
you want to go there.

Support for SATA drives also seems to be a problem with some versions, 
but since I am still in  the "Dark Ages" of IDE, I have no insight into 
that problem, other than reported problems on the users list.
>
> Decisions, Decisions, Decisions !!
>
> It is just puzzling which way to go - particularly as I want to use 
> the 'NZ reverse dial' patch that Russ has come up with for my nice 
> 'new' ivory old bakelite NZ telephone with its +64 phone number whose 
> numbering is based on the use of a 'reverse' dial.
>
Given that, so far, there is little need for the extra features in 1.4, 
it can't hurt too much to stick with the 1.2 version for now
I would suggest, from my long ago experience with FC2, that you consider 
moving to CentOS for the next build, and see what works

I have cloned several drives for CNET users in the US who want to get up 
and running without digging into Linux
All are based on CentOS 3.x and Asterisk 1.2
So far, so good.

I even have one CNET user running on RH9 and a K-6-233 machine.
He has one X100 card and one FXS ATA, doesn't mess with the computer and 
is happy not to .

MOST of what we do doesn't have to be bleeding edge
But then, I am still running 2 Win98 machines here, alongside NT4, Win2K 
and XP

Does it work?
Don't F**k with it!

John Novack

> Incidentally, the first 'Oslo dial' telephone I've ever seen on eBay  
> is currently for sale in Canada - eBay item No 220095311266   - bid 
> now to use Russ's patch !! :-)
>
> Ian Jolly
> +64 85 32 864 from CNET
> +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" <voip at ckts.info>
> Sent: Saturday, March 31, 2007 10:19 PM
> Subject: [VoIP] Patches
>
>
>> Back in late Feb, Russ posted some patches to Zaptel and Asterisk for
>> several items, including Oslo and NZ dials and early opening of the
>> audio channel to hear pulsing
>> The current version of Zaptel and Asterisk have  progressed, as of
>> today, at least, to 1.2.16 and 1.2.17
>> I have finally been able to obtain the modified driver for my T1 card
>> and am now current ,
>> I see in the patches references to 1.2.13 for Zaptel, and have to wonder
>> if this will fail on 1.2.16
>> Remember that SOME of us aren't well versed in Linux or applying the
>> patches, so perhaps someone could post a short tutorial on applying
>> patches and any modifications necessary for the later versions.
>> Remember this needs to be "for dummies". Or at least for those of us
>> that don't eat,drink and sleep Linux.
>>
>> John Novack
>
>

From jnovack at stromberg-carlson.org  Sun Apr  1 10:49:34 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 01 Apr 2007 11:49:34 -0400
Subject: [VoIP] SxS Call Progress Sounds
In-Reply-To: <20070401150701.42195.qmail@web51612.mail.re2.yahoo.com>
References: <20070401150701.42195.qmail@web51612.mail.re2.yahoo.com>
Message-ID: <460FD48E.5070001@stromberg-carlson.org>



ikjtel wrote:
> This seems to be working fairly nicely now.  Switching
> from an X100P-FXO to a channel-bank FXO has *vastly*
> improved the experience, because the dial tone is
> still perfectly audible throughout the *entire* first
> digit, including the onhook portions of the outpulsing
> cycle.  There is no missing audio in the entire
> sequence, especially now you can hear the "clunk" as
> the selector cuts through to the connector!  
>
> Too bad the "2600" SF pulsing stuff isn't included.  I
> once lived in a No. 5 Crossbar that had SF trunks to a
> number of outlying SxS offices and spent a lot of time
> with a homemade 2600 keying unit dialing through
> them...  If the SF stuff were present, it would be
> possible to dial through the machine remotely, *in
> real time*.
>   
Perhaps a collaboration could see that back in down the road.
One CNEt person is considering some SF trunking in his switch as well, 
but he needs to finish his CO shed and install his EM switch before 
going on line
With a little luck that will happen before Summer.
> Not sure exactly what to attribute the "irregular"
> pulsing.  The proper way to isolate this will be to
> back out the patches and listen with a butt set to the
> outpulsing, see if the removal of the patch makes the
> pulsing more regular or not.
>   
With Zaptel-1.2.16 there seems to be a problem BEFORE patching.
IF I am able to today, I will look into that

> John Novack - your XY office returns a fast-busy to
> the caller during the outpulsing of the digit into its
> first selector. 
Right now, I have an FXO circuit from my T1 doing ALL dialing into an 
ITEC first selector, then the selector outputs, which are all 
multiplied, connects to the respective connectors.
In effect, I have 3 linegorups with 3 sets of first selectors, digit 
absorbing, and 4 hundreds groups of connectors. The WE linegroup and 
first selectors are pulse only, the ITEC linegroup and first selector 
have built in TT converters, and the S-C linegroup and first selector 
also digit absorb.
Due to a shortage of tuits, ( round ones ) the S-C isn't yet fully wired.
EVENTUALLY, I plan on using the X-Y send section of selectors as 
incoming, both for Asterisk and K3200 trunk carrier
Probably more than you wanted to know about how I chose to wire up my 
switch, but from this you can see that IF you got a fast busy on 25xx, 
that was probably due to the mis pulse dialing.
Right now I can't seem to properly dial 2231 for a RNA every call from 
Asterisk, yet from my WE switch it always completes.

I also get this curious error message
Apr  1 11:44:33 WARNING[870]: chan_zap.c:4074 zt_handle_event: 
Ring/Off-hook in strange state 6 on channel 25

Because of my brand of T1 card, I am forced to use the govarion version 
of Zaptel
They have released a patch for future versions, however.
I really have no way of determining where the problem is, but for sure 
it isn't Russ and Max patches.

John Novack


From ikj1234i at yahoo.com  Sun Apr  1 11:10:27 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Sun, 1 Apr 2007 09:10:27 -0700 (PDT)
Subject: [VoIP] Real-time dialling and tandem stacking
In-Reply-To: <460FD48E.5070001@stromberg-carlson.org>
Message-ID: <20070401161027.67864.qmail@web51612.mail.re2.yahoo.com>

In brief, here's what might be required to move
forward on this...

1.  Establishment of direct interoffice trunk groups
between offices.  From the perspective of the SxS
office, this should be largely identical to how things
would have been done in real offices in the old
days...

2. For the outgoing leg of the trunk, the outgoing SxS
selector levels would terminate on some sort of zaptel
"FXS-like" instance within asterisk.  Asterisk would
be set up to send SF signalling outward on these ports
(essentially DC-to-SF conversion).  It's been a long
time since I looked at this, but if I recall
correctly, this is a standard Zaptel feature and would
require no extra patching.  It's not immediately clear
how the "S" lead signal would be generated back to the
selector, but I'm sure someone has already solved that
problem...

3. The asterisk routing would presumably be set up
such that these calls would be specially tagged
somehow to identify them as direct interoffice trunk
calls rather than ordinary CNET user calls.  When
siezed in the outgoing office, asterisk would
automatically initiate a VOIP connection to the proper
destination in the called office, based on the
selector level that was selected.

4. Incoming from the VOIP world, the direct
interoffice trunks would be mapped to zaptel FXO ports
that would then terminate on incoming selector levels
on the EM machine.  Presumably the EM office might be
arranged not to return dial tone on these trunks.  The
zaptel driver on this leg of the call would require my
"2600/SF" patch in order to reconvert the pulses from
SF to DC for application to the incoming selectors in
the called office.  Multi-hop operation should not be
a problem, since the SF patch is configured to notch
the 2600 and not pass it forward to succeeding legs of
the call...

5. When the caller initially connects to the first SxS
office in the connection, it might be set up to simply
give them access to a first selector which returns
dial tone without any initially-dialled digits (i.e.,
unlike in CNET today).  The caller would connect, hear
the dial tone, and begin dialling using SF
outpulsing...

6. Channel-bank FXO should be used throughout instead
of X100P in order to provide a realistic listening
experience...

Max


 
____________________________________________________________________________________
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From kxt at fubegra.net  Sun Apr  1 11:35:53 2007
From: kxt at fubegra.net (Russ Price)
Date: Sun, 01 Apr 2007 11:35:53 -0500
Subject: [VoIP] SxS Call Progress Sounds
In-Reply-To: <20070401150701.42195.qmail@web51612.mail.re2.yahoo.com>
References: <20070401150701.42195.qmail@web51612.mail.re2.yahoo.com>
Message-ID: <460FDF69.10702@fubegra.net>

ikjtel wrote:
> Not sure exactly what to attribute the "irregular"
> pulsing.  The proper way to isolate this will be to
> back out the patches and listen with a butt set to the
> outpulsing, see if the removal of the patch makes the
> pulsing more regular or not.

Lack of a butt set makes that impossible for me - however, I went back 
to zaptel 1.2.7 (and had to manually patch one file because there were 
significant differences), and still had irregular pulsing.  I have since 
gone back to zaptel 1.2.16.

It may well be that this is something unique to Adtran FXOs.  However, 
I'm not buying another channel bank to investigate this.

Sometimes, the irregular pulsing even goes beyond the rather forgiving 
Panasonic switch's tolerances, and I get an inaccurate result.

Compare 442-2106 (TDM400P) with 442-6106 (Adtran).

	Russ
	CNET: 1-442-7877
	FWD: 699408

From erwin at darcoury.nl  Sun Apr  1 12:28:56 2007
From: erwin at darcoury.nl (Erwin Dokter)
Date: Sun, 1 Apr 2007 19:28:56 +0200
Subject: [VoIP] SIP error message
References: <mailman.3837.1175441438.2853.voip@ckts.info>
Message-ID: <000501c77483$39e75c30$0501a8c0@windekind.demon.nl>

SIP does indeed have a concept of routing, called re-invites. This is
basically how SIP forwarding works. If a number is not reachable on a node,
that node can either tell the calling node to try somewhere else, or forward
the call itself. The limit of re-invites is usually 70 hops.

The most likely cause for your error message is probably a
mis-configuration; either re-invites is set too low, or two nodes keer
forwarding to eachother, resulting in a loop.

-- 
Erwin Dokter


----- Original Message -----
> Date: Sat, 31 Mar 2007 21:00:18 -0400
> From: "Lee Spenadel" <lee at spenadel.com>
> Subject: [VoIP] SIP error message
> To: "'Voice Over IP Tandem for Analog Switches'" <voip at ckts.info>
>
> What conditions would give the SIP error message below?  I've peered and
> registered with sipphone.  In the routing world I understand hop counts
and
> when they can be exceeded.  Is this a similar concept in the SIP world?
>
>     -- Called 17474745000)@proxy01.sipphone.com
>     -- Got SIP response 483 "Too Many Hops" back from 198.65.166.131
>



From jnovack at stromberg-carlson.org  Sun Apr  1 13:25:16 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 01 Apr 2007 14:25:16 -0400
Subject: [VoIP] Patches
In-Reply-To: <460F3B43.50003@fubegra.net>
References: <460ED063.2070103@stromberg-carlson.org>
	<460F3B43.50003@fubegra.net>
Message-ID: <460FF90C.4010703@stromberg-carlson.org>



Russ Price wrote:
> John Novack wrote:
> When all else fails, check my web page at <http://cnet.fubegra.net/>
Got that
Understand that my comments were NOT meant as a critisizm of Russ or 
anyone in particular.
All of us have some difficulty in communicating something we know well 
to those who aren't as far along in their understanding.

> I have found that the patches will apply just fine to Zaptel 1.2.16 
> and Asterisk 1.2.17.
In reading the patch file, one of the few items I understand is a 
reference to asterisk-1.2.16
Should this be edited to asterisk-1.2.17 to apply to 1.2.17??

>
> I have also updated the page, with a more detailed HOWTO, based on the 
> instructions I gave Lee.  Direct link:
>
> <http://cnet.fubegra.net/cnet-howto.html>
>
Thanks for that as well.
> NOTE: Lee and I have found that if you are using a channel bank FXO 
> port to pulse dial into a step switch, all bets are off!  I recently 
> acquired an FXO card for my channel bank,
What channel bank are you using?
> and I've noticed that the pulses sound irregular (and also seem to 
> have an improper break/make ratio).
This seems to be a change in the later zaptel versions.
Mine does seem to misdial too often.
And it is pulse dialing into an ITEC selector, which SHOULD be more 
forgiving.
> My Panasonic switch can handle the ragged pulses, but Lee's stepper is 
> a bit more picky.  Use either an X100P or TDM400P card instead, or be 
> content with using DTMF.
That is too bad, since not everyone has that choice.

Thanks again, Russ and Max

John Novack


From jnovack at stromberg-carlson.org  Sun Apr  1 14:39:20 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 01 Apr 2007 15:39:20 -0400
Subject: [VoIP] Tone duration
Message-ID: <46100A68.4080506@stromberg-carlson.org>


One of you smart Asterisk guys know where in Zaptel the DURATION of a 
specific DTMF pair is controlled?
One of the Cnet guys is having trouble with misdialing in DTMF to his 
step switch, and we'd like to extend the duration of each digit dialed.
The switch doesn't mis dial from a butt set, but Asterisk does.

I will be installing the latest Zaptel and Asterisk first, so this needs 
to be changed in 1.2.16

TIA

John Novack



From lee at spenadel.com  Sun Apr  1 15:48:11 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Sun, 1 Apr 2007 16:48:11 -0400
Subject: [VoIP] SIP error message
In-Reply-To: <000501c77483$39e75c30$0501a8c0@windekind.demon.nl>
References: <mailman.3837.1175441438.2853.voip@ckts.info>
	<000501c77483$39e75c30$0501a8c0@windekind.demon.nl>
Message-ID: <014101c7749f$10946050$31bd20f0$@com>

Here's another interesting error that I got; who thought that a register
might not be a register:

Apr  1 16:24:41 WARNING[2219]: chan_sip.c:9919 handle_response_register: Got
200 OK on REGISTER that isn't a register


Lee

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Erwin Dokter
Sent: Sunday, April 01, 2007 1:29 PM
To: voip at ckts.info
Subject: Re: [VoIP] SIP error message

SIP does indeed have a concept of routing, called re-invites. This is
basically how SIP forwarding works. If a number is not reachable on a node,
that node can either tell the calling node to try somewhere else, or forward
the call itself. The limit of re-invites is usually 70 hops.

The most likely cause for your error message is probably a
mis-configuration; either re-invites is set too low, or two nodes keer
forwarding to eachother, resulting in a loop.

-- 
Erwin Dokter


----- Original Message -----
> Date: Sat, 31 Mar 2007 21:00:18 -0400
> From: "Lee Spenadel" <lee at spenadel.com>
> Subject: [VoIP] SIP error message
> To: "'Voice Over IP Tandem for Analog Switches'" <voip at ckts.info>
>
> What conditions would give the SIP error message below?  I've peered and
> registered with sipphone.  In the routing world I understand hop counts
and
> when they can be exceeded.  Is this a similar concept in the SIP world?
>
>     -- Called 17474745000)@proxy01.sipphone.com
>     -- Got SIP response 483 "Too Many Hops" back from 198.65.166.131
>


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


From david at josephson.com  Sun Apr  1 16:55:46 2007
From: david at josephson.com (David Josephson)
Date: Sun, 01 Apr 2007 14:55:46 -0700
Subject: [VoIP] Tone duration, etc.
In-Reply-To: <46100A68.4080506@stromberg-carlson.org>
References: <46100A68.4080506@stromberg-carlson.org>
Message-ID: <46102A62.9090605@josephson.com>

John Novack wrote:

>One of you smart Asterisk guys know where in Zaptel the DURATION of a 
>specific DTMF pair is controlled?
>  
>
toneduration=n ;milliseconds

in zapata.conf

There is a good description of all the options in zapata.conf at 
http://wiki.sangoma.com/ast-original-zapata

This changes in 1.4 where there is apparently a system-wide tone 
duration config somewhere.

I can't see any reason to move to 1.4 yet, but it does have a Morse Code 
function...

And by the way, thanks for the help on the VT1000's, they are now 
working. I had neglected to set type=friend, and with the changes to the 
pulse duration values they work with each rotary set I've tried. Now I 
see that voipsupply doesn't list them anymore, maybe they were too much 
trouble for people to provision?
--
David Josephson

From voip at rexnet.us  Sun Apr  1 16:53:47 2007
From: voip at rexnet.us (Rexford M. Ennis)
Date: Sun, 1 Apr 2007 17:53:47 -0400
Subject: [VoIP] CAC Access II Channel Bank
Message-ID: <GKEPJHPBPKDGAGBOBILCIECDCEAA.voip@rexnet.us>

Greetings;

Has anyone experience using a CAC Access II Channel Bank with 24 FXS and the
Asterisk 1.2 (Trixbox)? I just got my TE110P T1 card back from repair and I
don't have much experience with T1. John Novack said that one or two of you
are using Adtran 750's. Do they work with dial pulse? I would think so, but?

Rexford M. Ennis
Grindstone Island
Clayton, NY 13624

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From ikj1234i at yahoo.com  Sun Apr  1 17:27:26 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Sun, 1 Apr 2007 15:27:26 -0700 (PDT)
Subject: [VoIP] Patches
In-Reply-To: <460FF90C.4010703@stromberg-carlson.org>
Message-ID: <41804.66341.qm@web51611.mail.re2.yahoo.com>


OK, I for one would like to try to get a better handle
on this (the outpulsing problem[s])...

When dialling into Lee's switch sometimes I receive a
fast-busy which I have assumed was caused by a
mis-dial.

Sometimes I can hear what sounds like a pause or
"burp" in what should be a regular rythm of pulses...

Has anyone been able to isolate whether this is due to
:

* improper make/break ratio
* version of asterisk and/or zaptel
* version of driver
* type of hardware in use (channel bank vs. x100p vs.
tdm400; brand of channel bank), etc.
* whether the "audible pulsing" patches have been
applied or not

The answers to the above might provide a clue 

Max

--- John Novack <jnovack at stromberg-carlson.org> wrote:
[snipped out some parts]
> Mine does seem to misdial too often.
> And it is pulse dialing into an ITEC selector, which
> SHOULD be more 
> forgiving.
> > My Panasonic switch can handle the ragged pulses,
> but Lee's stepper is 
> > a bit more picky.  Use either an X100P or TDM400P
> card instead, or be 
> > content with using DTMF.
> That is too bad, since not everyone has that choice.



 
____________________________________________________________________________________
Now that's room service!  Choose from over 150,000 hotels
in 45,000 destinations on Yahoo! Travel to find your fit.
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From ikj1234i at yahoo.com  Sun Apr  1 17:43:16 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Sun, 1 Apr 2007 15:43:16 -0700 (PDT)
Subject: [VoIP] SxS Call Progress Sounds
In-Reply-To: <460FD48E.5070001@stromberg-carlson.org>
Message-ID: <94610.90405.qm@web51607.mail.re2.yahoo.com>


--- John Novack <jnovack at stromberg-carlson.org> wrote:

> Probably more than you wanted to know about how I
> chose to wire up my 
> switch, but from this you can see that IF you got a
> fast busy on 25xx, 
> that was probably due to the mis pulse dialing.
> Right now I can't seem to properly dial 2231 for a

John

I probably didn't state this clearly in the first
email...  What I was saying was that it's *normal* in
an XY (and yours is no exception) to hear fast-busy
from the first selector during the outpulsing of the
first digit, no matter what digit is dialed.  At the
time of the initial break pulse of the first digit,
the first selector stops sending dial tone and starts
sending fast-busy.  IIRC, if the first selector is a
DA selector, and the first digit(s) absorbed, the busy
tone comes back on each subsequent digit dialled to
the first selector...  Western Electric style 1st
selectors send back continuous dial tone during this
period...

Max


 
____________________________________________________________________________________
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From kxt at fubegra.net  Sun Apr  1 18:24:09 2007
From: kxt at fubegra.net (Russ Price)
Date: Sun, 01 Apr 2007 18:24:09 -0500
Subject: [VoIP] Patches
In-Reply-To: <41804.66341.qm@web51611.mail.re2.yahoo.com>
References: <41804.66341.qm@web51611.mail.re2.yahoo.com>
Message-ID: <46103F19.6010902@fubegra.net>

ikjtel wrote:
> Sometimes I can hear what sounds like a pause or
> "burp" in what should be a regular rythm of pulses...
> 
> Has anyone been able to isolate whether this is due to
> :
> 
> * improper make/break ratio
> * version of asterisk and/or zaptel
> * version of driver
> * type of hardware in use (channel bank vs. x100p vs.
> tdm400; brand of channel bank), etc.
> * whether the "audible pulsing" patches have been
> applied or not

FWIW, my TDM400P's pulsing is audibly better than that from the Adtran 
FXO.  I don't hear any raggedness, and the make/break ratio sounds more 
appropriate on the TDM400P.

This has been true with Zaptel 1.2.16 as well as 1.2.7, both with open 
audio path.  I'm running Asterisk 1.2.17 here, with the open audio path.

The channel bank is an Adtran 750, with a 1175407L2 quad FXO card, 5 
quad FXS cards, and a 1175012L1 BCU, and the T1 card is a Digium TE110P.

I tried vanilla Zaptel 1.2.16 (no patches), and still got occasional 
misdials when pulse dialing into my Panasonic switch.

I also tried both B8ZS and AMI coding, and D4 and ESF framing, and they 
made no difference.

	Russ
	CNET: 1-442-7877
	FWD: 699408

From jnovack at stromberg-carlson.org  Sun Apr  1 20:36:10 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 01 Apr 2007 21:36:10 -0400
Subject: [VoIP] Patches
In-Reply-To: <46103F19.6010902@fubegra.net>
References: <41804.66341.qm@web51611.mail.re2.yahoo.com>
	<46103F19.6010902@fubegra.net>
Message-ID: <46105E0A.3000705@stromberg-carlson.org>



Russ Price wrote:
> ikjtel wrote:
>   
>> Sometimes I can hear what sounds like a pause or
>> "burp" in what should be a regular rythm of pulses...
>>
>> Has anyone been able to isolate whether this is due to
>> :
>>
>> * improper make/break ratio
>> * version of asterisk and/or zaptel
>> * version of driver
>> * type of hardware in use (channel bank vs. x100p vs.
>> tdm400; brand of channel bank), etc.
>> * whether the "audible pulsing" patches have been
>> applied or not
>>     
>
> FWIW, my TDM400P's pulsing is audibly better than that from the Adtran 
> FXO.  I don't hear any raggedness, and the make/break ratio sounds more 
> appropriate on the TDM400P.
>
> This has been true with Zaptel 1.2.16 as well as 1.2.7, both with open 
> audio path.  I'm running Asterisk 1.2.17 here, with the open audio path.
>
> The channel bank is an Adtran 750, with a 1175407L2 quad FXO card, 5 
> quad FXS cards, and a 1175012L1 BCU, and the T1 card is a Digium TE110P.
>
> I tried vanilla Zaptel 1.2.16 (no patches), and still got occasional 
> misdials when pulse dialing into my Panasonic switch.
>
> I also tried both B8ZS and AMI coding, and D4 and ESF framing, and they 
> made no difference.
>
> 	Russ
> 	CNET: 1-442-7877
> 	FWD: 699408
>   
Ok, for you Adtran users:
What, if any difference is there between the configurations for an FXO 
between DPT and FXO loop
Adtran offers both as options, but I am unclear if there is any difference.
Any Channel Bank EXPERTS?

John Novack


From voiptandem at shaneyoung.com  Sun Apr  1 20:59:46 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Sun, 01 Apr 2007 20:59:46 -0500
Subject: [VoIP] Patches
In-Reply-To: <46105E0A.3000705@stromberg-carlson.org>
References: <41804.66341.qm@web51611.mail.re2.yahoo.com>
	<46103F19.6010902@fubegra.net> <46105E0A.3000705@stromberg-carlson.org>
Message-ID: <20070401205946.6qmsgpqidcsww4sg@mail.shaneyoung.com>

Quoting John Novack <jnovack at stromberg-carlson.org>:


> Ok, for you Adtran users:
> What, if any difference is there between the configurations for an FXO
> between DPT and FXO loop
> Adtran offers both as options, but I am unclear if there is any difference.
> Any Channel Bank EXPERTS?


In other channel banks, the DPT shows up as an E&M channel on the DS1  
and returns answer supervision on polarity reversal.

I have DPT's on incoming selectors for "toll" completing.

--Shane--Shane
+1-821-7311 CNET



From ratguy at bellsouth.net  Sun Apr  1 21:53:38 2007
From: ratguy at bellsouth.net (Jayson Smith)
Date: Sun, 1 Apr 2007 22:53:38 -0400
Subject: [VoIP] Real-time dialling and tandem stacking
References: <20070401161027.67864.qmail@web51612.mail.re2.yahoo.com>
Message-ID: <000e01c774d2$1d1f6760$0600a8c0@bluegrasspals.com>

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

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


> In brief, here's what might be required to move
> forward on this...
>
> 1.  Establishment of direct interoffice trunk groups
> between offices.  From the perspective of the SxS
> office, this should be largely identical to how things
> would have been done in real offices in the old
> days...
>
> 2. For the outgoing leg of the trunk, the outgoing SxS
> selector levels would terminate on some sort of zaptel
> "FXS-like" instance within asterisk.  Asterisk would
> be set up to send SF signalling outward on these ports
> (essentially DC-to-SF conversion).  It's been a long
> time since I looked at this, but if I recall
> correctly, this is a standard Zaptel feature and would
> require no extra patching.  It's not immediately clear
> how the "S" lead signal would be generated back to the
> selector, but I'm sure someone has already solved that
> problem...
>
> 3. The asterisk routing would presumably be set up
> such that these calls would be specially tagged
> somehow to identify them as direct interoffice trunk
> calls rather than ordinary CNET user calls.  When
> siezed in the outgoing office, asterisk would
> automatically initiate a VOIP connection to the proper
> destination in the called office, based on the
> selector level that was selected.
>
> 4. Incoming from the VOIP world, the direct
> interoffice trunks would be mapped to zaptel FXO ports
> that would then terminate on incoming selector levels
> on the EM machine.  Presumably the EM office might be
> arranged not to return dial tone on these trunks.  The
> zaptel driver on this leg of the call would require my
> "2600/SF" patch in order to reconvert the pulses from
> SF to DC for application to the incoming selectors in
> the called office.  Multi-hop operation should not be
> a problem, since the SF patch is configured to notch
> the 2600 and not pass it forward to succeeding legs of
> the call...
>
> 5. When the caller initially connects to the first SxS
> office in the connection, it might be set up to simply
> give them access to a first selector which returns
> dial tone without any initially-dialled digits (i.e.,
> unlike in CNET today).  The caller would connect, hear
> the dial tone, and begin dialling using SF
> outpulsing...
>
> 6. Channel-bank FXO should be used throughout instead
> of X100P in order to provide a realistic listening
> experience...
>
> Max
>
>
>
>
____________________________________________________________________________
________
> 8:00? 8:25? 8:40? Find a flick in no time
> with the Yahoo! Search movie showtime shortcut.
> http://tools.search.yahoo.com/shortcuts/#news
> _______________________________________________
> 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  Sun Apr  1 23:31:36 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Sun, 1 Apr 2007 21:31:36 -0700 (PDT)
Subject: [VoIP] Real-time dialling and tandem stacking
In-Reply-To: <000e01c774d2$1d1f6760$0600a8c0@bluegrasspals.com>
Message-ID: <353062.11746.qm@web51605.mail.re2.yahoo.com>


--- Jayson Smith <ratguy at bellsouth.net> wrote:
> [snip]   Now at
> this point, I would need to know when it was safe to
> continue dialing. Then,

There should be a distinctive click when the selector
in the distant office is seized.  Ordinary Joe callers
wouldn't know what to listen for, but I doubt anyone
on this list fits that description ;-)

Max


 
____________________________________________________________________________________
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Get more visitors on your site using Yahoo! Search Marketing.
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From kxt at fubegra.net  Mon Apr  2 22:29:21 2007
From: kxt at fubegra.net (Russ Price)
Date: Mon, 02 Apr 2007 22:29:21 -0500
Subject: [VoIP] More pulse dialing mysteries
Message-ID: <4611CA11.1070402@fubegra.net>

Delving into the code, I noticed that zaptel.h naively uses a 50:50 
pulse make/break ratio (ZT_DEFAULT_PULSEBREAKTIME and 
ZT_DEFAULT_PULSEBREAKTIME are both set to 50 ms).

I tried using the US-standard 61:39 and UK-standard 67:33 ratios, and I 
found that pulsing was still accurate with the TDM400P card - but the 
Adtran FXO card got even worse.

Another thing I tried, which made no difference either way, was to 
configure the channel bank to generate the timing and tell the TE110P to 
use the channel bank as the timing source.

I'm not really a T1 guru, but I'm wondering if the way that T1 sends 
supervisory signals messes with the timing of the pulses?  I kind of 
doubt it, though, since FXS pulse dialing works just fine on the channel 
bank.

	Russ
	CNET: 1-442-7877
	FWD: 699408

From brandon at parsonstx.com  Tue Apr  3 00:53:16 2007
From: brandon at parsonstx.com (Brandon Parsons)
Date: Tue, 3 Apr 2007 00:53:16 -0500
Subject: [VoIP] more info on the VT1005
Message-ID: <8c272bbf0704022253n297305eduf7d80a3013c18f9@mail.gmail.com>

After tinkering some more, I take back some of what I say about the
VT1005.  The settings I provide do 'stick', except for the new encryption
key.  I believe I now have a reliable recipe for hacking a vendor-locked
VT1005 reliably.  One nugget that may be useful to some of you is a
TERMINATE statement for dial plans.

----------------><---------- beginning of recipe
----------------><------------------

(1) if the VT1005 is vendor locked, it's likely that the Telnet port is not
open on either interface, so crack open the device (just a couple of screws
on the bottom), and remove the cover.  Using serial TTL connect to the
jumper on the board located opposite all of the plugs; it's 38.4 Kbps
8N1.  The 4 pins are (1 being nearest the LED, 4 being farthest):  1) +5V --
leave unconnected; 2) Gnd; 3) Tx -- from VT1005; 4) Rx -- to VT1005.  More
info can be found here:   http://www.0xdecafbad.com/node/16

(2) once you're in you can use a remote management function for setting the
sip tftp parameters (including the encryption key).  I've been asked not to
post the config file utility or documents--which contain the name and format
of the command--so just ask if you need them.  This is the point that had
tripped me up before and confused me regarding the stickiness of settings.
After modifying the encryption key to all zeros (no encryption), it will
load your TFTP file (when you reboot the device, and maybe when it times
out).  Those settings from teh TFTP file will stay in NVRAM (not sure if it
the device actually has to register for them to be committed), however I've
been unable to get the config file parameter that changes the encryption key
to work, so the next attempt to fetch the TFTP file will use the factory
programmed encryption key again.  The format for the config file parameter
'TeleSipRc4EncryptionKey' isn't specified, so I'm not sure if it's just
plain hex syntax, double quoted string of hex digits with leading 0x (or
without leading 0x), etc.  All I know is that it's 256 bits (64 hex
nibbles).

(3) using the config utility and sample config files (referenced in other
posts and in step (2)), create the TFTP binary, and load it.  Occasionally
it tries to reload the tftp file, but since I'm unable to get the encryption
permanently turned off it fails.  Doesn't matter because the settings are
stored in NVRAM, and if you enable it in the config file you'll have telnet
access on the WAN or LAN interface (or both) where you can manually disable
the encryption for another reload if necessary.

(4) Dial plan notes.  Instead of using a plain old DIGITMAP entry with a
static number of digits, you can specify variable length via a couple of
methods.  Standard DIGITMAP entry:
BTIOPT TeleSipDialPlanEntry[0]             = "DIGITMAP:1XXXXXXXXXX"  #
standard NA dial plan

You could also specify something like:
BTIOPT TeleSipDialPlanEntry[0]             = "DIGITMAP:011[2-9]X.T"  #
international calling NA style (T means timeout)

and an easy method is to use a dial terminator.  This is what I use to dial
any sipbroker based number from my sipphone account:
BTIOPT TeleSipDialPlanEntry[0]             = "TERMINATE:*XXXX.:#"
   broken down as follows:
    *XXX  --  specifier to access any of the sipbroker networks
    X.      -- 1-N digit extension at any of the sipbroker networks
    :       -- just a delimeter
    #       -- terminating character (either * or # )

so to hear the monkeys at the unfix network
(http://unfix.org/projects/voip/) I dial  *512 103#
or *011188888# for the sipbroker welcome call

----------------><---------- end of recipe
----------------><------------------

So that's all I know.  Many thanks to David Josephson.  When I'm done
tinkering with the device in the next couple of weeks, and get an SPA3102,
I'll send it to David.  If he doesn't want it, first person to ask can have
it, free of charge.  Still have to manually clear the encryption key each
time to load a tftp file, but telnet ports are now open on both LAN and WAN,
so it's not too much trouble.  If I can get the info on how to set
permanently (or semi-permanently) set the key to zero, it'll be that much
better.  I don't have any pulse-dial equipment, so this box is mostly a test
vehicle for different SIP services.  I really want that little box with 1
FXS and 1 FXO port.  It would be awesome to find a box with 1 FXO and 2 FXS
for $120 or less that's not more than 3 times the size of the VT1005.

brandon

From john_reads_cnet_via_archives at covert.org  Tue Apr  3 06:56:36 2007
From: john_reads_cnet_via_archives at covert.org (John R. Covert)
Date: Tue, 3 Apr 2007 07:56:36 -0400 (EDT)
Subject: [VoIP] Patches
In-Reply-To: <460F3B43.50003@fubegra.net>
References: <460ED063.2070103@stromberg-carlson.org>
Message-ID: <20070403120843.AACC156305@ns1.vyger.net>

These patches, especially for the NZ/Oslo dial, should be submitted
so that they become part of the regular distribution and don't have
to be continually applied.  As long as they are under control of
options for the config files and work reliably, they should be
accepted.  If you submit them, please let me know the tracking
number, and I'll put in my oar in favor of them.

/john

From john_reads_cnet_via_archives at covert.org  Tue Apr  3 07:08:44 2007
From: john_reads_cnet_via_archives at covert.org (John R. Covert)
Date: Tue, 3 Apr 2007 08:08:44 -0400 (EDT)
Subject: [VoIP] Legacy Ringback from Recordings in Asterisk
Message-ID: <20070403123018.10D115634B@ns1.vyger.net>

It has been asked before if it's possible in Asterisk to use a
recording for ringback.  The answer is YES.  IMHO, don't bother
unless you've got a good ulaw recording, but anyway, here's how.

In musiconhold.conf:

[5xbring]
mode=files
directory=/var/lib/asterisk/moh-5xbring

Put a single .ulaw file in that directory.  Note that I had to
take an mp3 original; I need to replace this with something that
was never compressed.  I trimmed some silence from the front so
that we get ring pretty quickly (so that people will hear it if
I answer on the first actual ring before the "tone plant" cycles
in with this recording).  I made it an exact number of cycles
long so that if it wraps around on no answer, it doesn't sound
weird.

sox 5xbring.mp3 -r 8000 -tul 5xbring.ulaw trim 2.7 50

Then the dial statement in extensions.conf gets the music-on-hold
modifier with the class created above, thus:

Dial(${RINGALL},120,m(5xbring)t)

I use it whenever ringing my "reach-me" numbers, whether ringing
everything in the PBX, the default for generic incoming calls, or
when forwarding to my cellphones and remote stations when I'm
travelling.  You'd hear it if you call me on CNET 1-263-9900.

/john

P.S.: I think we lost some submissions from last Friday, because I sent
this in once before and there are some gaps in the archive, as
though we all went on hols [getting ready for my UK/Germany trip
next week].

From john_reads_cnet_via_archives at covert.org  Tue Apr  3 07:31:45 2007
From: john_reads_cnet_via_archives at covert.org (John R. Covert)
Date: Tue, 3 Apr 2007 08:31:45 -0400 (EDT)
Subject: [VoIP] Real-time dialling and tandem stacking
In-Reply-To: <20070401161027.67864.qmail@web51612.mail.re2.yahoo.com>
References: <460FD48E.5070001@stromberg-carlson.org>
Message-ID: <20070403125228.B34765634B@ns1.vyger.net>

>Establishment of direct interoffice trunk groups between offices

Globally, we don't have to do to anything special to create a direct
inbound trunk group in any office.  We already have all that we need
in the current infrastructure.

ENUM as currently configured will respond just as well to a "selector
level" as to a full seven digit number.

Anyone can publish selector levels in the directory, just as they
currently publish seven digit numbers, at least at the 1-NXX-X level,
and for the 1-NXX level, just use a throw-away letter.

On the incoming side, you just configure your Asterisk to handle
the number of digits you're sent (taking the possibility of a
throw-away letter into account if you accept four digits inbound
rather than one, two, or three).

On the outbound side, you code the trunk to operate on seizure,
which is a standard feature in Asterisk, and based on the interface
that was selected, do the ENUM lookup of the partial number, and
connect.  The remaining digits can be sent inband.  You better have
a really good connection to your ISP and a good ISP if you expect
inband MF or SF to work.

For a switch, an outbound trunk group would seem to go to just a
single other CNET switch.  But the inbound trunk group would be
reachable by anyone who can properly signal into it.

/john

From john_reads_cnet_via_archives at covert.org  Tue Apr  3 07:59:49 2007
From: john_reads_cnet_via_archives at covert.org (John R. Covert)
Date: Tue, 3 Apr 2007 08:59:49 -0400 (EDT)
Subject: [VoIP] Update your directory entries, please!
Message-ID: <20070403131059.894E45634B@ns1.vyger.net>

If you've got something running on CNET, please update your
entries.  Every once in a while, I happen to notice a call that
comes through my switch and get curious about either its caller
id or what it's dialling via my gateways.

I'm not going to publicly "out" anyone, but I have noticed that
there is at least one working office code with some interesting
step-by-step office sounds (maybe just a recording, maybe not)
that isn't even on the list, and doesn't yet follow our convention
of putting an informational recording on -0001.  Peculiarly, in
the one office I'm thinking about, a call to that number is
_accepted_ but then somehow returns CHANUNAVAIL rather than
CONGESTION, which I find intriguing.

/john

From ikj1234i at yahoo.com  Tue Apr  3 10:06:17 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Tue, 3 Apr 2007 08:06:17 -0700 (PDT)
Subject: [VoIP] More pulse dialing mysteries
In-Reply-To: <4611CA11.1070402@fubegra.net>
Message-ID: <448330.18118.qm@web51607.mail.re2.yahoo.com>


Russ

This is interesting...  I am not surprised that
changing the T1 clocking source had no effect. 
Changing the m/b ratio probably just signifies that
things were already kind of marginal to begin with...

You might try enabling the 'printk' which I've pasted
below.  It might also be helpful to add a printout of
the "jiffies" value so as to get an idea of the
passage of time and specifically the points in time
when the rbs bits are being changed...

My expectation however would be that you'll see *both*
sets of timings (i.e., the working case and the
problem case) be correct at this point within the
code...  Offhand and without anything to back it, I
suspect the problem may be further down in the wct1xxp
drivers...

Unfortunately I don't have a channel bank FXO here.  I
was trying to figure if it could be simulated using a
channel bank FXS...

Max

[existing code from zaptel.c]
#ifdef CONFIG_ZAPATA_DEBUG
                                printk("Setting bits
to %d for channel %s state %d in %d signalling\n",
outs[x][txsig + 1], chan->name, txsig, chan->sig);
#endif  

--- Russ Price <kxt at fubegra.net> wrote:

> Delving into the code, I noticed that zaptel.h
> naively uses a 50:50 
> pulse make/break ratio (ZT_DEFAULT_PULSEBREAKTIME
> and 
> ZT_DEFAULT_PULSEBREAKTIME are both set to 50 ms).
> 
> I tried using the US-standard 61:39 and UK-standard
> 67:33 ratios, and I 
> found that pulsing was still accurate with the
> TDM400P card - but the 
> Adtran FXO card got even worse.
> 
> Another thing I tried, which made no difference
> either way, was to 
> configure the channel bank to generate the timing
> and tell the TE110P to 
> use the channel bank as the timing source.
> 
> I'm not really a T1 guru, but I'm wondering if the
> way that T1 sends 
> supervisory signals messes with the timing of the
> pulses?  I kind of 
> doubt it, though, since FXS pulse dialing works just
> fine on the channel 
> bank.
> 
> 	Russ
> 	CNET: 1-442-7877
> 	FWD: 699408
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 



 
____________________________________________________________________________________
Don't get soaked.  Take a quick peek at the forecast
with the Yahoo! Search weather shortcut.
http://tools.search.yahoo.com/shortcuts/#loc_weather

From john_reads_cnet_via_archives at covert.org  Tue Apr  3 10:30:08 2007
From: john_reads_cnet_via_archives at covert.org (John R. Covert)
Date: Tue, 3 Apr 2007 11:30:08 -0400 (EDT)
Subject: [VoIP] The Museum of Communications, Seattle
Message-ID: <20070403155140.78AE156346@ns1.vyger.net>

I just got off the phone with Don Ostrand, Curator of the Seattle
museum, one of the great museums of switching equipment, with an
actual working panel office and a large 5XB.

The purpose of my call was to arrange a special evening visit during
the Asterisk Boot Camp I expect to teach in Seattle 11-15 June.

But I also talked to him about getting the museum connected to
CNET.  He sounded interested, and I sent him to the ckts website
to have a look-see.  There is already a DSL connection into the
museum building, so it's just a matter of getting them set up
with an Asterisk box and some sort of channel bank.  Their switches
only operate during museum hours on Tuesdays, but they'd still be
a wonderful addition to the network.

/john

From ikj1234i at yahoo.com  Tue Apr  3 11:20:03 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Tue, 3 Apr 2007 09:20:03 -0700 (PDT)
Subject: [VoIP] patches
In-Reply-To: <20070403125228.B34765634B@ns1.vyger.net>
Message-ID: <20070403162003.15834.qmail@web51601.mail.re2.yahoo.com>


--- "John R. Covert"
<john_reads_cnet_via_archives at covert.org> wrote:
> [snip]
> These patches, especially for the NZ/Oslo dial,
> should be submitted so that they become part of
> the regular distribution and don't have to be
> continually applied.

Agreed.  Digium.com states that they will condescend
to accept patches only from those who sign a
waiver/disclaimer in order to be allowed to submit
patches, otherwise they'll reject them.

I ain't gonna sign it.  Anyone who wishes should feel
free to do so.  For the ones I've created, if you must
take credit for their creation, so be it.

Max



 
____________________________________________________________________________________
Need Mail bonding?
Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users.
http://answers.yahoo.com/dir/?link=list&sid=396546091

From voip at rexnet.us  Tue Apr  3 12:08:04 2007
From: voip at rexnet.us (Rexford M. Ennis)
Date: Tue, 3 Apr 2007 13:08:04 -0400
Subject: [VoIP] CAC Channel Bank
Message-ID: <GKEPJHPBPKDGAGBOBILCOEDHCEAA.voip@rexnet.us>

Greetings;

I have been reading about some of you using channel banks. A friend gave me
a CAC Access II channel bank with manuals. I also acquired a Digium TE110P
card. I know just enough about T1 to be dangerous. Have any of you had
experience with this combination? The CAC has two cards and they are both
FXS. I hope to end up with some stations I can use with some of my antique
phones.

I am running a Trixbox with a 2.5 gig AMD processor and 1 gig of ram. It
seems to work well. It is not connected to the outside world yet as I have a
satellite internet connection. I am working with a friend to install a
Ethernet bridge so that I can get off the satellite.

Thanks for any help.

Rexford M. Ennis
Grindstone Island
Clayton, NY 13624




From jnovack at stromberg-carlson.org  Tue Apr  3 12:17:14 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Tue, 03 Apr 2007 13:17:14 -0400
Subject: [VoIP] CAC Channel Bank
In-Reply-To: <GKEPJHPBPKDGAGBOBILCOEDHCEAA.voip@rexnet.us>
References: <GKEPJHPBPKDGAGBOBILCOEDHCEAA.voip@rexnet.us>
Message-ID: <46128C1A.4010002@stromberg-carlson.org>

Except for Trixbox, you should have no trouble configuring and having a 
24 station PBX.
Before you get too deep into configuration, you may want to consider a 
new configuration, without  Trixbox.
I worked a little with the early A at H, and found that, for CNET and 
learning Asterisk, it was counter productive.
Others on the list may have a different opinion.

John Novack


Rexford M. Ennis wrote:
> Greetings;
>
> I have been reading about some of you using channel banks. A friend gave me
> a CAC Access II channel bank with manuals. I also acquired a Digium TE110P
> card. I know just enough about T1 to be dangerous. Have any of you had
> experience with this combination? The CAC has two cards and they are both
> FXS. I hope to end up with some stations I can use with some of my antique
> phones.
>
> I am running a Trixbox with a 2.5 gig AMD processor and 1 gig of ram. It
> seems to work well. It is not connected to the outside world yet as I have a
> satellite internet connection. I am working with a friend to install a
> Ethernet bridge so that I can get off the satellite.
>
> Thanks for any help.
>
> Rexford M. Ennis
> Grindstone Island
> Clayton, NY 13624
>
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

From ratguy at bellsouth.net  Tue Apr  3 12:46:44 2007
From: ratguy at bellsouth.net (Jayson Smith)
Date: Tue, 3 Apr 2007 13:46:44 -0400
Subject: [VoIP] DTMF question
Message-ID: <000f01c77618$0b9734e0$0600a8c0@bluegrasspals.com>

Hi all,
Got a puzzling one for you guys. I recently got three Sipura SPA-2000 units,
and am using one, and like it. Just for the heck of it, I set
dtmfmode=inband for both lines in sip.conf, and have both lines set for
inband DTMF in the SPA. Still, calling one line from the other via my
Asterisk system, when I hit a touchtone on either phone, the other gets a
tiny blip of the original tone, then, once I let up on the button, a
regenerated, fixed-duration touchtone is heard. In other words, as far as I
can tell, inband DTMF isn't really working. My codec choice is Ulaw, so that
shouldn't be a problem. Any thoughts?
Thanks.


From jnovack at stromberg-carlson.org  Tue Apr  3 13:25:19 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Tue, 03 Apr 2007 14:25:19 -0400
Subject: [VoIP] DTMF question
In-Reply-To: <000f01c77618$0b9734e0$0600a8c0@bluegrasspals.com>
References: <000f01c77618$0b9734e0$0600a8c0@bluegrasspals.com>
Message-ID: <46129C0F.6030608@stromberg-carlson.org>

I use:
dtmfmode=rfc2833
for my SIP devices, and have no problem
I do believe that is the suggested mode for use with Asterisk, and may 
solve your DTMF problem

John Novack


Jayson Smith wrote:
> Hi all,
> Got a puzzling one for you guys. I recently got three Sipura SPA-2000 units,
> and am using one, and like it. Just for the heck of it, I set
> dtmfmode=inband for both lines in sip.conf, and have both lines set for
> inband DTMF in the SPA. Still, calling one line from the other via my
> Asterisk system, when I hit a touchtone on either phone, the other gets a
> tiny blip of the original tone, then, once I let up on the button, a
> regenerated, fixed-duration touchtone is heard. In other words, as far as I
> can tell, inband DTMF isn't really working. My codec choice is Ulaw, so that
> shouldn't be a problem. Any thoughts?
> Thanks.
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

From david at josephson.com  Tue Apr  3 14:56:57 2007
From: david at josephson.com (David Josephson)
Date: Tue, 03 Apr 2007 12:56:57 -0700
Subject: [VoIP] CAC Channel Bank
In-Reply-To: <46128C1A.4010002@stromberg-carlson.org>
References: <GKEPJHPBPKDGAGBOBILCOEDHCEAA.voip@rexnet.us>
	<46128C1A.4010002@stromberg-carlson.org>
Message-ID: <4612B189.4040306@josephson.com>

John Novack wrote:
> Except for Trixbox, you should have no trouble configuring and having a 
> 24 station PBX.
> Before you get too deep into configuration, you may want to consider a 
> new configuration, without  Trixbox.
> I worked a little with the early A at H, and found that, for CNET and 
> learning Asterisk, it was counter productive.
> Others on the list may have a different opinion.
>   
I found that playing with AAH, and Trixbox, was useful to see the sort 
of things that could be done. Once I saw some of the possibilities, it 
was time to start pruning all the cruft that Trixbox had added. It does 
allow you to download one iso image and its patches, follow the cookbook 
and have a running installation that has been compiled on your hardware. 
Then your forklift the whole /etc/asterisk directory and start from scratch.


--
David Josephson

From ratguy at bellsouth.net  Tue Apr  3 17:46:33 2007
From: ratguy at bellsouth.net (Jayson Smith)
Date: Tue, 3 Apr 2007 18:46:33 -0400
Subject: [VoIP] DTMF question
References: <000f01c77618$0b9734e0$0600a8c0@bluegrasspals.com>
	<46129C0F.6030608@stromberg-carlson.org>
Message-ID: <000a01c77641$ed9894a0$0600a8c0@bluegrasspals.com>

Hi,
This isn't what I want. I think it'd be cool to use actual inband DTMF. Such
that, when I hit a button, the other end of the call can hear the tone for
as long as I keep it held down, and any imperfections, etc. That's what's
inband DTMF is supposed to be, right? Where no digital translation happens,
and the two ends of the call just send DTMF directly to each other?
Thanks.

----- Original Message ----- 
From: "John Novack" <jnovack at stromberg-carlson.org>
To: "Jayson Smith" <ratguy at bellsouth.net>; "Voice Over IP Tandem for Analog
Switches" <voip at ckts.info>
Sent: Tuesday, April 03, 2007 2:25 PM
Subject: Re: [VoIP] DTMF question


> I use:
> dtmfmode=rfc2833
> for my SIP devices, and have no problem
> I do believe that is the suggested mode for use with Asterisk, and may
> solve your DTMF problem
>
> John Novack
>
>
> Jayson Smith wrote:
> > Hi all,
> > Got a puzzling one for you guys. I recently got three Sipura SPA-2000
units,
> > and am using one, and like it. Just for the heck of it, I set
> > dtmfmode=inband for both lines in sip.conf, and have both lines set for
> > inband DTMF in the SPA. Still, calling one line from the other via my
> > Asterisk system, when I hit a touchtone on either phone, the other gets
a
> > tiny blip of the original tone, then, once I let up on the button, a
> > regenerated, fixed-duration touchtone is heard. In other words, as far
as I
> > can tell, inband DTMF isn't really working. My codec choice is Ulaw, so
that
> > shouldn't be a problem. Any thoughts?
> > Thanks.
> >
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> >
>


From jnovack at stromberg-carlson.org  Tue Apr  3 19:07:14 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Tue, 03 Apr 2007 20:07:14 -0400
Subject: [VoIP] DTMF question
In-Reply-To: <000a01c77641$ed9894a0$0600a8c0@bluegrasspals.com>
References: <000f01c77618$0b9734e0$0600a8c0@bluegrasspals.com>	<46129C0F.6030608@stromberg-carlson.org>
	<000a01c77641$ed9894a0$0600a8c0@bluegrasspals.com>
Message-ID: <4612EC32.1030604@stromberg-carlson.org>

Others may know more, but I believe Asterisk doesn't play well with 
inband DTMF
For that you need a real switch!

John Novack


Jayson Smith wrote:
> Hi,
> This isn't what I want. I think it'd be cool to use actual inband DTMF. Such
> that, when I hit a button, the other end of the call can hear the tone for
> as long as I keep it held down, and any imperfections, etc. That's what's
> inband DTMF is supposed to be, right? Where no digital translation happens,
> and the two ends of the call just send DTMF directly to each other?
> Thanks.
>
> ----- Original Message ----- 
> From: "John Novack" <jnovack at stromberg-carlson.org>
> To: "Jayson Smith" <ratguy at bellsouth.net>; "Voice Over IP Tandem for Analog
> Switches" <voip at ckts.info>
> Sent: Tuesday, April 03, 2007 2:25 PM
> Subject: Re: [VoIP] DTMF question
>
>
>   
>> I use:
>> dtmfmode=rfc2833
>> for my SIP devices, and have no problem
>> I do believe that is the suggested mode for use with Asterisk, and may
>> solve your DTMF problem
>>
>> John Novack
>>
>>
>> Jayson Smith wrote:
>>     
>>> Hi all,
>>> Got a puzzling one for you guys. I recently got three Sipura SPA-2000
>>>       
> units,
>   
>>> and am using one, and like it. Just for the heck of it, I set
>>> dtmfmode=inband for both lines in sip.conf, and have both lines set for
>>> inband DTMF in the SPA. Still, calling one line from the other via my
>>> Asterisk system, when I hit a touchtone on either phone, the other gets
>>>       
> a
>   
>>> tiny blip of the original tone, then, once I let up on the button, a
>>> regenerated, fixed-duration touchtone is heard. In other words, as far
>>>       
> as I
>   
>>> can tell, inband DTMF isn't really working. My codec choice is Ulaw, so
>>>       
> that
>   
>>> shouldn't be a problem. Any thoughts?
>>> Thanks.
>>>
>>> _______________________________________________
>>> 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 voiptandem at shaneyoung.com  Tue Apr  3 19:12:31 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Tue, 03 Apr 2007 19:12:31 -0500
Subject: [VoIP] DTMF question
In-Reply-To: <4612EC32.1030604@stromberg-carlson.org>
References: <000f01c77618$0b9734e0$0600a8c0@bluegrasspals.com>
	<46129C0F.6030608@stromberg-carlson.org>
	<000a01c77641$ed9894a0$0600a8c0@bluegrasspals.com>
	<4612EC32.1030604@stromberg-carlson.org>
Message-ID: <20070403191231.37ekx4cojk0kogw0@mail.shaneyoung.com>

You should be able to do DTMF all the way through, end to end, inband  
if everything is set correctly on your SIP devices.

IAX, on the other hand, always handles DTMF out of band.

If Asterisk sees that one side is in-band (the Sipura) and the other  
side (someone else's switch) is out of band, it should send the DTMF  
out of band.



Quoting John Novack <jnovack at stromberg-carlson.org>:

> Others may know more, but I believe Asterisk doesn't play well with
> inband DTMF
> For that you need a real switch!
>
> John Novack
>
>
> Jayson Smith wrote:
>> Hi,
>> This isn't what I want. I think it'd be cool to use actual inband DTMF. Such
>> that, when I hit a button, the other end of the call can hear the tone for
>> as long as I keep it held down, and any imperfections, etc. That's what's
>> inband DTMF is supposed to be, right? Where no digital translation happens,
>> and the two ends of the call just send DTMF directly to each other?
>> Thanks.
>>
>> ----- Original Message -----
>> From: "John Novack" <jnovack at stromberg-carlson.org>
>> To: "Jayson Smith" <ratguy at bellsouth.net>; "Voice Over IP Tandem for Analog
>> Switches" <voip at ckts.info>
>> Sent: Tuesday, April 03, 2007 2:25 PM
>> Subject: Re: [VoIP] DTMF question
>>
>>
>>
>>> I use:
>>> dtmfmode=rfc2833
>>> for my SIP devices, and have no problem
>>> I do believe that is the suggested mode for use with Asterisk, and may
>>> solve your DTMF problem
>>>
>>> John Novack
>>>
>>>
>>> Jayson Smith wrote:
>>>
>>>> Hi all,
>>>> Got a puzzling one for you guys. I recently got three Sipura SPA-2000
>>>>
>> units,
>>
>>>> and am using one, and like it. Just for the heck of it, I set
>>>> dtmfmode=inband for both lines in sip.conf, and have both lines set for
>>>> inband DTMF in the SPA. Still, calling one line from the other via my
>>>> Asterisk system, when I hit a touchtone on either phone, the other gets
>>>>
>> a
>>
>>>> tiny blip of the original tone, then, once I let up on the button, a
>>>> regenerated, fixed-duration touchtone is heard. In other words, as far
>>>>
>> as I
>>
>>>> can tell, inband DTMF isn't really working. My codec choice is Ulaw, so
>>>>
>> that
>>
>>>> shouldn't be a problem. Any thoughts?
>>>> Thanks.
>>>>
>>>> _______________________________________________
>>>> 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/
>

--Shane
+1-821-7311 CNET



From lee at spenadel.com  Thu Apr  5 20:18:25 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Thu, 5 Apr 2007 21:18:25 -0400
Subject: [VoIP] Jane Barbe
Message-ID: <015b01c777e9$79ecdd00$6dc69700$@com>

Is there somewhere that I can download Jane's standard "canned" recordings,
including the individual numbers?

 

TIA

Lee

 

 

 

 

If your car could travel at the speed of light, would your headlights work?

 


From andrew.e.green at gmail.com  Thu Apr  5 20:33:33 2007
From: andrew.e.green at gmail.com (Andrew Green)
Date: Thu, 5 Apr 2007 23:03:33 -0230
Subject: [VoIP] Jane Barbe
In-Reply-To: <015b01c777e9$79ecdd00$6dc69700$@com>
References: <015b01c777e9$79ecdd00$6dc69700$@com>
Message-ID: <c30e256a0704051833o2979b141g9fe9bea01724681@mail.gmail.com>

Try here http://www.dmine.com/phworld/sounds/misc/ you could try
downloading the "Jane Barbe Collection" and using your favourite audio
editing program (ie: Audacity, free http://audacity.sf.net/) to chop
them up.

-Andrew

On 4/5/07, Lee Spenadel <lee at spenadel.com> wrote:
> Is there somewhere that I can download Jane's standard "canned" recordings,
> including the individual numbers?
>
>
>
> TIA
>
> Lee
>
>
>
>
>
>
>
>
>
> If your car could travel at the speed of light, would your headlights work?
>
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>

From lee at spenadel.com  Thu Apr  5 20:41:43 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Thu, 5 Apr 2007 21:41:43 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <c30e256a0704051833o2979b141g9fe9bea01724681@mail.gmail.com>
References: <015b01c777e9$79ecdd00$6dc69700$@com>
	<c30e256a0704051833o2979b141g9fe9bea01724681@mail.gmail.com>
Message-ID: <016b01c777ec$bba52ce0$32ef86a0$@com>

Thanks Andrew,

I already had that recording - I wasn't sure if someone had already chopped
up the various recordings and posted them somewhere.

Lee

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Andrew Green
Sent: Thursday, April 05, 2007 9:34 PM
To: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Jane Barbe

Try here http://www.dmine.com/phworld/sounds/misc/ you could try
downloading the "Jane Barbe Collection" and using your favourite audio
editing program (ie: Audacity, free http://audacity.sf.net/) to chop
them up.

-Andrew

On 4/5/07, Lee Spenadel <lee at spenadel.com> wrote:
> Is there somewhere that I can download Jane's standard "canned"
recordings,
> including the individual numbers?
>
>
>
> TIA
>
> Lee
>
>
>
>
>
>
>
>
>
> If your car could travel at the speed of light, would your headlights
work?
>
>
>
> _______________________________________________
> 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 bbj at innismir.net  Thu Apr  5 21:22:13 2007
From: bbj at innismir.net (Ben Jackson)
Date: Thu, 5 Apr 2007 22:22:13 -0400
Subject: [VoIP] Jane Barbe
Message-ID: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>

I have it... Somewhere... All the neutral and falling inflections and most of the AIS phrases... Let me look around...

--
Ben Jackson - bbj at innismir.net - http://www.innismir.net 
>From my Cell Phone

-----Original Message-----
From: "Lee Spenadel" <lee at spenadel.com>
To: "'Voice Over IP Tandem for Analog Switches'" <voip at ckts.info>
Sent: 4/5/07 9:41 PM
Subject: Re: [VoIP] Jane Barbe

Thanks Andrew,

I already had that recording - I wasn't sure if someone had already chopped
up the various recordings and posted them somewhere.

Lee

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Andrew Green
Sent: Thursday, April 05, 2007 9:34 PM
To: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Jane Barbe

Try here http://www.dmine.com/phworld/sounds/misc/ you could try
downloading the "Jane Barbe Collection" and using your favourite audio
editing program (ie: Audacity, free http://audacity.sf.net/) to chop
them up.

-Andrew

On 4/5/07, Lee Spenadel <lee at spenadel.com> wrote:
> Is there somewhere that I can download Jane's standard "canned"
recordings,
> including the individual numbers?
>
>
>
> TIA
>
> Lee
>
>
>
>
>
>
>
>
>
> If your car could travel at the speed of light, would your headlights
work?
>
>
>
> _______________________________________________
> 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 bellsouth.net  Thu Apr  5 21:38:49 2007
From: ratguy at bellsouth.net (Jayson Smith)
Date: Thu, 5 Apr 2007 22:38:49 -0400
Subject: [VoIP] Jane Barbe
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>
Message-ID: <000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>

Hi,
I don't know of anybody with a working AIS machine. Now if only we could
find one, we could get crisp clear recordings of everything on the machine.
I know of one guy who has a time and temperature machine with Jane Barbe's
voice on it. Unfortunately, IIRC the only way to record everything would be
to keep calling it over and over again while adjusting the temperature
reading to get all the temperatures, and also call it at least sixty times
to get every minute, and twelve times to get every hour.
Jayson

----- Original Message ----- 
From: "Ben Jackson" <bbj at innismir.net>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>; "'Voice
Over IP Tandem for Analog Switches'" <voip at ckts.info>
Sent: Thursday, April 05, 2007 10:22 PM
Subject: Re: [VoIP] Jane Barbe


> I have it... Somewhere... All the neutral and falling inflections and most
of the AIS phrases... Let me look around...
>
> --
> Ben Jackson - bbj at innismir.net - http://www.innismir.net
> >From my Cell Phone
>
> -----Original Message-----
> From: "Lee Spenadel" <lee at spenadel.com>
> To: "'Voice Over IP Tandem for Analog Switches'" <voip at ckts.info>
> Sent: 4/5/07 9:41 PM
> Subject: Re: [VoIP] Jane Barbe
>
> Thanks Andrew,
>
> I already had that recording - I wasn't sure if someone had already
chopped
> up the various recordings and posted them somewhere.
>
> Lee
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Andrew Green
> Sent: Thursday, April 05, 2007 9:34 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Jane Barbe
>
> Try here http://www.dmine.com/phworld/sounds/misc/ you could try
> downloading the "Jane Barbe Collection" and using your favourite audio
> editing program (ie: Audacity, free http://audacity.sf.net/) to chop
> them up.
>
> -Andrew
>
> On 4/5/07, Lee Spenadel <lee at spenadel.com> wrote:
> > Is there somewhere that I can download Jane's standard "canned"
> recordings,
> > including the individual numbers?
> >
> >
> >
> > TIA
> >
> > Lee
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > If your car could travel at the speed of light, would your headlights
> work?
> >
> >
> >
> > _______________________________________________
> > 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 bbj at innismir.net  Thu Apr  5 22:14:32 2007
From: bbj at innismir.net (Ben Jackson)
Date: Thu, 05 Apr 2007 23:14:32 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>
Message-ID: <4615BB18.9090806@innismir.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Ben Jackson wrote:
> I have it... Somewhere... All the neutral and falling inflections and most of the AIS phrases... Let me look around...
> 


I knew it was on my website somewhere.

There are bits missing, mainly because I've never been able to find
recordings of certain parts. But it is mostly complete.

http://www.innismir.net/etc/jane.zip

Share and Enjoy :)

- --
Ben Jackson - N1WBV - New Bedford, MA
bbj <at> innismir.net - http://www.innismir.net/
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From lee at spenadel.com  Thu Apr  5 22:29:03 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Thu, 5 Apr 2007 23:29:03 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <4615BB18.9090806@innismir.net>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>
	<4615BB18.9090806@innismir.net>
Message-ID: <017b01c777fb$b9d6d7b0$2d848710$@com>

Many thanks
Lee

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Ben Jackson
Sent: Thursday, April 05, 2007 11:15 PM
To: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Jane Barbe

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Ben Jackson wrote:
> I have it... Somewhere... All the neutral and falling inflections and most
of the AIS phrases... Let me look around...
> 


I knew it was on my website somewhere.

There are bits missing, mainly because I've never been able to find
recordings of certain parts. But it is mostly complete.

http://www.innismir.net/etc/jane.zip

Share and Enjoy :)

- --
Ben Jackson - N1WBV - New Bedford, MA
bbj <at> innismir.net - http://www.innismir.net/
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.5 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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=S2f9
-----END PGP SIGNATURE-----

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


From david at josephson.com  Fri Apr  6 00:54:00 2007
From: david at josephson.com (David Josephson)
Date: Thu, 05 Apr 2007 22:54:00 -0700
Subject: [VoIP] Jane Barbe
In-Reply-To: <4615BB18.9090806@innismir.net>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>
	<4615BB18.9090806@innismir.net>
Message-ID: <4615E078.6030200@josephson.com>

Ben Jackson wrote:

>I knew it was on my website somewhere.
>
>There are bits missing, mainly because I've never been able to find
>recordings of certain parts. But it is mostly complete.
>
>http://www.innismir.net/etc/jane.zip
>  
>
Complete enough. A couple weeks ago I built those recordings into an 
ANAC simulator at 1-555-1122.

--
David Josephson

From stfkerman at jps.net  Thu Apr  5 23:57:46 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Fri, 06 Apr 2007 00:57:46 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>
	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>
Message-ID: <4615D34A.5090409@jps.net>

Which AIS message tracks are not currently available in digital form?

Steph

Jayson Smith wrote:
> Hi,
> I don't know of anybody with a working AIS machine. Now if only we could
> find one, we could get crisp clear recordings of everything on the machine.
>
>   

From stfkerman at jps.net  Thu Apr  5 23:59:29 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Fri, 06 Apr 2007 00:59:29 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <4615E078.6030200@josephson.com>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<4615BB18.9090806@innismir.net>
	<4615E078.6030200@josephson.com>
Message-ID: <4615D3B1.3040103@jps.net>

Your message time stamps appear to be incorrect.  You are 1 hour ahead 
of actual time as well as the vast majority of messages I am receiving 
from others.

SK

David Josephson wrote:
> Ben Jackson wrote:
>
>   
>> I knew it was on my website somewhere.
>>
>> There are bits missing, mainly because I've never been able to find
>> recordings of certain parts. But it is mostly complete.
>>
>> http://www.innismir.net/etc/jane.zip
>>  
>>
>>     
> Complete enough. A couple weeks ago I built those recordings into an 
> ANAC simulator at 1-555-1122.
>
> --
> David Josephson
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

From ratguy at bellsouth.net  Fri Apr  6 00:03:34 2007
From: ratguy at bellsouth.net (Jayson Smith)
Date: Fri, 6 Apr 2007 01:03:34 -0400
Subject: [VoIP] Jane Barbe
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>
	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>
	<4615D34A.5090409@jps.net>
Message-ID: <000901c77808$ed76b600$0600a8c0@bluegrasspals.com>

Hi,
Actually, I don't know. Judging from some things, I once thought that Ben's
recordings originated from Evan Doorbell tapes, and he certainly did tape a
lot of Jane Barbe AIS systems. There are some dynamic differences with some
of the digits, etc. You'd have to ask Ben what's totally missing. Do you
have a Jane Barbe AIS?
Jayson

----- Original Message ----- 
From: "Steph Kerman" <stfkerman at jps.net>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Cc: "Jayson Smith" <ratguy at bellsouth.net>
Sent: Friday, April 06, 2007 12:57 AM
Subject: Re: [VoIP] Jane Barbe


> Which AIS message tracks are not currently available in digital form?
>
> Steph
>
> Jayson Smith wrote:
> > Hi,
> > I don't know of anybody with a working AIS machine. Now if only we could
> > find one, we could get crisp clear recordings of everything on the
machine.
> >
> >
>


From stfkerman at jps.net  Fri Apr  6 00:06:15 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Fri, 06 Apr 2007 01:06:15 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <000901c77808$ed76b600$0600a8c0@bluegrasspals.com>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>
	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>
Message-ID: <4615D547.5000501@jps.net>

I have the complete AIS CSRAF (Common Systems Recorded Announcement 
Frame) rack including the playback amplifier electronics for all tracks 
and duplicate machines.  It is not assembled and powered.

SK

Jayson Smith wrote:
> Hi,
> Actually, I don't know. Judging from some things, I once thought that Ben's
> recordings originated from Evan Doorbell tapes, and he certainly did tape a
> lot of Jane Barbe AIS systems. There are some dynamic differences with some
> of the digits, etc. You'd have to ask Ben what's totally missing. Do you
> have a Jane Barbe AIS?
> Jayson
>
> ----- Original Message ----- 
> From: "Steph Kerman" <stfkerman at jps.net>
> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> Cc: "Jayson Smith" <ratguy at bellsouth.net>
> Sent: Friday, April 06, 2007 12:57 AM
> Subject: Re: [VoIP] Jane Barbe
>
>
>   
>> Which AIS message tracks are not currently available in digital form?
>>
>> Steph
>>
>> Jayson Smith wrote:
>>     
>>> Hi,
>>> I don't know of anybody with a working AIS machine. Now if only we could
>>> find one, we could get crisp clear recordings of everything on the
>>>       
> machine.
>   
>>>       
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

From ratguy at bellsouth.net  Fri Apr  6 00:58:38 2007
From: ratguy at bellsouth.net (Jayson Smith)
Date: Fri, 6 Apr 2007 01:58:38 -0400
Subject: [VoIP] Jane Barbe
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>
	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>
	<4615D547.5000501@jps.net>
Message-ID: <000401c77810$9eed2520$0600a8c0@bluegrasspals.com>

Hi,
I just downloaded Ben's zip file, and can say for certain that 2d, 3d, 4d
and 8d are missing. The d files seem to be the ones that are used at the end
of a spoken number. There are differences in the quality on many of these
recordings. This is probably because of the recordings being taken from
different sources. It would be interesting to know if Evan Doorbell has a
complete set of numbers and phrases spoken by these machines.
On a somewhat related note, how do you get Asterisk to say digits with a
different set of numbers than the defaults, and how do you tell it to do
like the AIS does, E.G. six, two, two. two, zero, two, zero. instead of just
six two two two zero two zero
Jayson

----- Original Message ----- 
From: "Steph Kerman" <stfkerman at jps.net>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Cc: "Jayson Smith" <ratguy at bellsouth.net>
Sent: Friday, April 06, 2007 1:06 AM
Subject: Re: [VoIP] Jane Barbe


> I have the complete AIS CSRAF (Common Systems Recorded Announcement
> Frame) rack including the playback amplifier electronics for all tracks
> and duplicate machines.  It is not assembled and powered.
>
> SK
>
> Jayson Smith wrote:
> > Hi,
> > Actually, I don't know. Judging from some things, I once thought that
Ben's
> > recordings originated from Evan Doorbell tapes, and he certainly did
tape a
> > lot of Jane Barbe AIS systems. There are some dynamic differences with
some
> > of the digits, etc. You'd have to ask Ben what's totally missing. Do you
> > have a Jane Barbe AIS?
> > Jayson
> >
> > ----- Original Message ----- 
> > From: "Steph Kerman" <stfkerman at jps.net>
> > To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> > Cc: "Jayson Smith" <ratguy at bellsouth.net>
> > Sent: Friday, April 06, 2007 12:57 AM
> > Subject: Re: [VoIP] Jane Barbe
> >
> >
> >
> >> Which AIS message tracks are not currently available in digital form?
> >>
> >> Steph
> >>
> >> Jayson Smith wrote:
> >>
> >>> Hi,
> >>> I don't know of anybody with a working AIS machine. Now if only we
could
> >>> find one, we could get crisp clear recordings of everything on the
> >>>
> > machine.
> >
> >>>
> >
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> >
>


From kxt at fubegra.net  Fri Apr  6 07:59:22 2007
From: kxt at fubegra.net (Russ Price)
Date: Fri, 06 Apr 2007 07:59:22 -0500
Subject: [VoIP] Oslo phone
Message-ID: <4616442A.3070008@fubegra.net>

I recently found an Oslo phone on eBay (one that was in a bit better 
shape than the one in Ian's post).  The good news: when I use it on the 
channel bank with the dial pulse mapping option, it dials correctly. The 
bad news: When I ring the phone on the channel bank, it will ring a 
random number of times, and then suddenly Zaptel will think that the 
phone was answered, just for it to immediately go back on-hook.

If I connect it to my TDM400P, this problem doesn't occur.  However, the 
dial is slightly too fast (shades of my North Electric set), and it 
doesn't dial accurately on the TDM400P, pulse mapping or no.

The ring problem also doesn't occur when the phone is directly connected 
to the CO, or even, surprisingly enough, the Panasonic switch.

When I open the phone up, I don't see any obviously bad insulation, but 
the wiring is all bundled up, and most of the internal wiring is 
*soldered* to the terminal strip.  I can't disconnect the ringer without 
unsoldering or cutting wires.

My prime suspect is the capacitor, but still it rings just fine without 
glitching on anything except the channel bank, and it doesn't matter 
which port I use on the channel bank, either.

At this point, my options are: regulate the dial speed and use it on the 
TDM400P, use it as outgoing-only on the channel bank, or try to figure 
out what's going wrong.

	Russ
	CNET: 1-442-7877
	FWD: 699408

From bbj at innismir.net  Fri Apr  6 09:27:40 2007
From: bbj at innismir.net (Ben Jackson)
Date: Fri, 06 Apr 2007 10:27:40 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <000901c77808$ed76b600$0600a8c0@bluegrasspals.com>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>
	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>
	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>
Message-ID: <461658DC.4020001@innismir.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Jayson Smith wrote:
> Hi,
> Actually, I don't know. Judging from some things, I once thought that Ben's
> recordings originated from Evan Doorbell tapes, and he certainly did tape a
> lot of Jane Barbe AIS systems. There are some dynamic differences with some
> of the digits, etc. You'd have to ask Ben what's totally missing. Do you
> have a Jane Barbe AIS?
> Jayson

I was going through the list of intercepts I sent to the list and kept
saying to myself "I thought I have that"

I looked in Asterisk's sound directory... And sure enough...


http://www.innismir.net/moreintercept.zip


That's far more complete then the one I sent previously...

- --
Ben Jackson - N1WBV - New Bedford, MA
bbj <at> innismir.net - http://www.innismir.net/
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.5 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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4oq4hM35KNhEX2QjZO+Dqifauvsxk23BTZAx5CAXth/dRa2B3zSQ6BzZEvQBeZkF
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Eqifn53R0kHHzbZxtgj33Bcp1187ljHFBk7Rlg2LnShZCHgA2eCpSiOq6czoqlKc
132c/x27HTfeFwIKLbKYlfemHbYm1DOd7cv3BGVjKm/N7JpeS7m7EA==
=5Go6
-----END PGP SIGNATURE-----


From stfkerman at jps.net  Fri Apr  6 09:31:28 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Fri, 06 Apr 2007 10:31:28 -0400
Subject: [VoIP] Oslo phone
In-Reply-To: <4616442A.3070008@fubegra.net>
References: <4616442A.3070008@fubegra.net>
Message-ID: <461659C0.5000200@jps.net>

Perhaps it just has a low impedance ringer that causes false trip 
because of inadequate design of the channel card. 

If there is leakage you will be able to see that by placing a DMM set to 
DC mA in series during ringing.  The DMM should register zero during the 
silent intervals.  In theory it should register zero during the ringing 
intervals too but some DMMs do not behave well below 60 Hz so it may 
thrash around, which would be a false indication.  If it registers zero 
during ringing that would be a reliable indication that there is no 
leakage problem.

Most likely if it does have too low an impedance for the channel bank 
you can put a resistor equal to the DC resistance of the ringer in 
series with it and it will still ring adequately without tripping 
ringing.  What is the resistance of the ringer coil and the series 
capacitance value? 

What model Oslo phone is it?  There are modern ones and very antique 
ones such as were used on the first WECo Rotary offices.  These have the 
7001 type dial with 0 and 1 holes at the bottom and a number plate which 
rotates with the finger wheel.

Steph

Russ Price wrote:
> I recently found an Oslo phone on eBay (one that was in a bit better 
> shape than the one in Ian's post).  The good news: when I use it on the 
> channel bank with the dial pulse mapping option, it dials correctly. The 
> bad news: When I ring the phone on the channel bank, it will ring a 
> random number of times, and then suddenly Zaptel will think that the 
> phone was answered, just for it to immediately go back on-hook.
>
> If I connect it to my TDM400P, this problem doesn't occur.  However, the 
> dial is slightly too fast (shades of my North Electric set), and it 
> doesn't dial accurately on the TDM400P, pulse mapping or no.
>
> The ring problem also doesn't occur when the phone is directly connected 
> to the CO, or even, surprisingly enough, the Panasonic switch.
>
> When I open the phone up, I don't see any obviously bad insulation, but 
> the wiring is all bundled up, and most of the internal wiring is 
> *soldered* to the terminal strip.  I can't disconnect the ringer without 
> unsoldering or cutting wires.
>
> My prime suspect is the capacitor, but still it rings just fine without 
> glitching on anything except the channel bank, and it doesn't matter 
> which port I use on the channel bank, either.
>
> At this point, my options are: regulate the dial speed and use it on the 
> TDM400P, use it as outgoing-only on the channel bank, or try to figure 
> out what's going wrong.
>
>   

From lee at spenadel.com  Fri Apr  6 09:53:01 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Fri, 6 Apr 2007 10:53:01 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <461658DC.4020001@innismir.net>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>
	<461658DC.4020001@innismir.net>
Message-ID: <01bd01c7785b$46d99490$d48cbdb0$@com>

Link is bad

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Ben Jackson
Sent: Friday, April 06, 2007 10:28 AM
To: Jayson Smith; Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Jane Barbe

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Jayson Smith wrote:
> Hi,
> Actually, I don't know. Judging from some things, I once thought that
Ben's
> recordings originated from Evan Doorbell tapes, and he certainly did tape
a
> lot of Jane Barbe AIS systems. There are some dynamic differences with
some
> of the digits, etc. You'd have to ask Ben what's totally missing. Do you
> have a Jane Barbe AIS?
> Jayson

I was going through the list of intercepts I sent to the list and kept
saying to myself "I thought I have that"

I looked in Asterisk's sound directory... And sure enough...


http://www.innismir.net/moreintercept.zip


That's far more complete then the one I sent previously...

- --
Ben Jackson - N1WBV - New Bedford, MA
bbj <at> innismir.net - http://www.innismir.net/
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.5 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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132c/x27HTfeFwIKLbKYlfemHbYm1DOd7cv3BGVjKm/N7JpeS7m7EA==
=5Go6
-----END PGP SIGNATURE-----

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


From bbj at innismir.net  Fri Apr  6 09:57:06 2007
From: bbj at innismir.net (Ben Jackson)
Date: Fri, 06 Apr 2007 10:57:06 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <01bd01c7785b$46d99490$d48cbdb0$@com>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>
	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>
	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>
	<461658DC.4020001@innismir.net> <01bd01c7785b$46d99490$d48cbdb0$@com>
Message-ID: <46165FC2.8000101@innismir.net>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Lee Spenadel wrote:
> Link is bad
> 

Whoops. Don't post before Coffee, kids

http://www.innismir.net/etc/moreintercept.zip


- --
Ben Jackson - N1WBV - New Bedford, MA
bbj <at> innismir.net - http://www.innismir.net/
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From david at josephson.com  Fri Apr  6 14:38:42 2007
From: david at josephson.com (David Josephson)
Date: Fri, 06 Apr 2007 12:38:42 -0700
Subject: [VoIP] Jane Barbe
In-Reply-To: <4615D547.5000501@jps.net>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>
	<4615D547.5000501@jps.net>
Message-ID: <4616A1C2.2060501@josephson.com>

Steph Kerman wrote:
> I have the complete AIS CSRAF (Common Systems Recorded Announcement 
> Frame) rack including the playback amplifier electronics for all tracks 
> and duplicate machines.  It is not assembled and powered.
>   
Is that in the warehouse I know about? We have a large assortment of 
analog magnetic tape repro equipment here. If we can get the transports 
(tape, drum or belt) running, we can probably extract the sounds 
recorded on those tracks with good fidelity. That would be an 
interesting project.

And, I have no doubt that there is random +/- 1 hour time stamp error in 
many systems, due to the 2007 daylight time punt. You probably got a 
message composed on my home computer which is still an hour ahead, 
having been manually bumped and then automatically advanced by Mr. 
Gates' wizards.

--
David
> SK
>
> Jayson Smith wrote:
>   
>> Hi,
>> Actually, I don't know. Judging from some things, I once thought that Ben's
>> recordings originated from Evan Doorbell tapes, and he certainly did tape a
>> lot of Jane Barbe AIS systems. There are some dynamic differences with some
>> of the digits, etc. You'd have to ask Ben what's totally missing. Do you
>> have a Jane Barbe AIS?
>> Jayson
>>
>> ----- Original Message ----- 
>> From: "Steph Kerman" <stfkerman at jps.net>
>> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
>> Cc: "Jayson Smith" <ratguy at bellsouth.net>
>> Sent: Friday, April 06, 2007 12:57 AM
>> Subject: Re: [VoIP] Jane Barbe
>>
>>
>>   
>>     
>>> Which AIS message tracks are not currently available in digital form?
>>>
>>> Steph
>>>
>>> Jayson Smith wrote:
>>>     
>>>       
>>>> Hi,
>>>> I don't know of anybody with a working AIS machine. Now if only we could
>>>> find one, we could get crisp clear recordings of everything on the
>>>>       
>>>>         
>> machine.
>>   
>>     
>>>>       
>>>>         
>> _______________________________________________
>> 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  Fri Apr  6 14:47:52 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Fri, 06 Apr 2007 15:47:52 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <4616A1C2.2060501@josephson.com>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>	<4615D547.5000501@jps.net>
	<4616A1C2.2060501@josephson.com>
Message-ID: <4616A3E8.1010803@jps.net>



David Josephson wrote:
> Is that in the warehouse I know about?
No.
> We have a large assortment of analog magnetic tape repro equipment 
> here. If we can get the transports (tape, drum or belt) running, we 
> can probably extract the sounds recorded on those tracks with good 
> fidelity. That would be an interesting project.
Thanks for offering.  I'm sure there are enough channels of the original 
electronics that work just fine to dub all 96 tracks serially.  If not, 
I have appropriate preamps that I could substitute.  If there *are* 
complications they might be related to control of the drive mechanism, 
probably not something you are particularly prepared to deal with any 
better than I am.
> And, I have no doubt that there is random +/- 1 hour time stamp error 
> in many systems, due to the 2007 daylight time punt. You probably got a
> message composed on my home computer which is still an hour ahead, 
> having been manually bumped and then automatically advanced by Mr. 
> Gates' wizards.
One needs to turn off auto-adjustment by unchecking the box and if 
correcting manually one needed to bump the time zone and then correct 
the time setting.

Steph
>
> Steph Kerman wrote:
>> I have the complete AIS CSRAF (Common Systems Recorded Announcement 
>> Frame) rack including the playback amplifier electronics for all tracks 
>> and duplicate machines.  It is not assembled and powered.
>>   
>>     
>
>   
>> SK
>>
>> Jayson Smith wrote:
>>   
>>     
>>> Hi,
>>> Actually, I don't know. Judging from some things, I once thought that Ben's
>>> recordings originated from Evan Doorbell tapes, and he certainly did tape a
>>> lot of Jane Barbe AIS systems. There are some dynamic differences with some
>>> of the digits, etc. You'd have to ask Ben what's totally missing. Do you
>>> have a Jane Barbe AIS?
>>> Jayson
>>>
>>> ----- Original Message ----- 
>>> From: "Steph Kerman" <stfkerman at jps.net>
>>> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
>>> Cc: "Jayson Smith" <ratguy at bellsouth.net>
>>> Sent: Friday, April 06, 2007 12:57 AM
>>> Subject: Re: [VoIP] Jane Barbe 
>>>       
>>>> Which AIS message tracks are not currently available in digital form?
>>>>
>>>> Steph
>>>>
>>>> Jayson Smith wrote:  
>>>>         
>>>>> Hi,
>>>>> I don't know of anybody with a working AIS machine. Now if only we could
>>>>> find one, we could get crisp clear recordings of everything on the machine.     
>>>>>         
>>>>>           
>>>   
>>>     
>>>       
>>>
>   

From david at josephson.com  Fri Apr  6 14:51:08 2007
From: david at josephson.com (David Josephson)
Date: Fri, 06 Apr 2007 12:51:08 -0700
Subject: [VoIP] Jane Barbe
In-Reply-To: <4616A3E8.1010803@jps.net>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>	<4615D547.5000501@jps.net>	<4616A1C2.2060501@josephson.com>
	<4616A3E8.1010803@jps.net>
Message-ID: <4616A4AC.9020402@josephson.com>

Steph Kerman wrote:
> David Josephson wrote:
>   
>> Is that in the warehouse I know about?
>>     
> No.
>   
>> We have a large assortment of analog magnetic tape repro equipment 
>> here. If we can get the transports (tape, drum or belt) running, we 
>> can probably extract the sounds recorded on those tracks with good 
>> fidelity. That would be an interesting project.
>>     
> Thanks for offering.  I'm sure there are enough channels of the original 
> electronics that work just fine to dub all 96 tracks serially.  If not, 
> I have appropriate preamps that I could substitute.  If there *are* 
> complications they might be related to control of the drive mechanism, 
> probably not something you are particularly prepared to deal with any 
> better than I am.
>   
Fine. When can you, I or we do this?

From stfkerman at jps.net  Fri Apr  6 14:57:03 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Fri, 06 Apr 2007 15:57:03 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <4616A4AC.9020402@josephson.com>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>	<4615D547.5000501@jps.net>	<4616A1C2.2060501@josephson.com>	<4616A3E8.1010803@jps.net>
	<4616A4AC.9020402@josephson.com>
Message-ID: <4616A60F.1050503@jps.net>

I haven't the faintest idea.  Under the present circumstances it enjoys 
an especially low level of priority.

SK

David Josephson wrote:
> Fine. When can you, I or we do this?
>
> Steph Kerman wrote:
>   
>> David Josephson wrote:
>>   
>>     
>>> Is that in the warehouse I know about?
>>>     
>>>       
>> No.
>>   
>>     
>>> We have a large assortment of analog magnetic tape repro equipment 
>>> here. If we can get the transports (tape, drum or belt) running, we 
>>> can probably extract the sounds recorded on those tracks with good 
>>> fidelity. That would be an interesting project.
>>>     
>>>       
>> Thanks for offering.  I'm sure there are enough channels of the original 
>> electronics that work just fine to dub all 96 tracks serially.  If not, 
>> I have appropriate preamps that I could substitute.  If there *are* 
>> complications they might be related to control of the drive mechanism, 
>> probably not something you are particularly prepared to deal with any 
>> better than I am.
>>   
>>     
>
>   

From lee at spenadel.com  Fri Apr  6 15:15:44 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Fri, 6 Apr 2007 16:15:44 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <4616A60F.1050503@jps.net>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>	<4615D547.5000501@jps.net>	<4616A1C2.2060501@josephson.com>	<4616A3E8.1010803@jps.net>	<4616A4AC.9020402@josephson.com>
	<4616A60F.1050503@jps.net>
Message-ID: <01e901c77888$5b46cb00$11d46100$@com>

It sure would be nice for us to enjoy clean crisp recordings of Jane.  What
would be a possible time line on this?

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Steph Kerman
Sent: Friday, April 06, 2007 3:57 PM
To: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Jane Barbe

I haven't the faintest idea.  Under the present circumstances it enjoys 
an especially low level of priority.

SK

David Josephson wrote:
> Fine. When can you, I or we do this?
>
> Steph Kerman wrote:
>   
>> David Josephson wrote:
>>   
>>     
>>> Is that in the warehouse I know about?
>>>     
>>>       
>> No.
>>   
>>     
>>> We have a large assortment of analog magnetic tape repro equipment 
>>> here. If we can get the transports (tape, drum or belt) running, we 
>>> can probably extract the sounds recorded on those tracks with good 
>>> fidelity. That would be an interesting project.
>>>     
>>>       
>> Thanks for offering.  I'm sure there are enough channels of the original 
>> electronics that work just fine to dub all 96 tracks serially.  If not, 
>> I have appropriate preamps that I could substitute.  If there *are* 
>> complications they might be related to control of the drive mechanism, 
>> probably not something you are particularly prepared to deal with any 
>> better than I am.
>>   
>>     
>
>   
_______________________________________________
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  Fri Apr  6 15:35:25 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Fri, 06 Apr 2007 16:35:25 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <01e901c77888$5b46cb00$11d46100$@com>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>	<4615D547.5000501@jps.net>	<4616A1C2.2060501@josephson.com>	<4616A3E8.1010803@jps.net>	<4616A4AC.9020402@josephson.com>	<4616A60F.1050503@jps.net>
	<01e901c77888$5b46cb00$11d46100$@com>
Message-ID: <4616AF0D.8040902@jps.net>

I've been waiting patiently for a good opportunity to use this phrase:  
What part of "I haven't the faintest idea.  Under the present 
circumstances it enjoys an especially low level of priority." don't you 
understand?  ;-)

SK

Lee Spenadel wrote:
> It sure would be nice for us to enjoy clean crisp recordings of Jane.  What
> would be a possible time line on this?
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Steph Kerman
> Sent: Friday, April 06, 2007 3:57 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Jane Barbe
>
> I haven't the faintest idea.  Under the present circumstances it enjoys 
> an especially low level of priority.
>
> SK
>
> David Josephson wrote:
>   
>> Fine. When can you, I or we do this?
>>
>> Steph Kerman wrote:
>>   
>>     
>>> David Josephson wrote:
>>>   
>>>     
>>>       
>>>> Is that in the warehouse I know about?
>>>>     
>>>>       
>>>>         
>>> No.
>>>   
>>>     
>>>       
>>>> We have a large assortment of analog magnetic tape repro equipment 
>>>> here. If we can get the transports (tape, drum or belt) running, we 
>>>> can probably extract the sounds recorded on those tracks with good 
>>>> fidelity. That would be an interesting project.
>>>>     
>>>>       
>>>>         
>>> Thanks for offering.  I'm sure there are enough channels of the original 
>>> electronics that work just fine to dub all 96 tracks serially.  If not, 
>>> I have appropriate preamps that I could substitute.  If there *are* 
>>> complications they might be related to control of the drive mechanism, 
>>> probably not something you are particularly prepared to deal with any 
>>> better than I am.
>>>   
>>>     
>>>       
>>   
>>     
> _______________________________________________
> 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 lee at spenadel.com  Fri Apr  6 15:45:59 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Fri, 6 Apr 2007 16:45:59 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <4616AF0D.8040902@jps.net>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>	<4615D547.5000501@jps.net>	<4616A1C2.2060501@josephson.com>	<4616A3E8.1010803@jps.net>	<4616A4AC.9020402@josephson.com>	<4616A60F.1050503@jps.net>	<01e901c77888$5b46cb00$11d46100$@com>
	<4616AF0D.8040902@jps.net>
Message-ID: <01eb01c7788c$95b59b50$c120d1f0$@com>

There you go demonstrating your usefulness again.  

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Steph Kerman
Sent: Friday, April 06, 2007 4:35 PM
To: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Jane Barbe

I've been waiting patiently for a good opportunity to use this phrase:  
What part of "I haven't the faintest idea.  Under the present 
circumstances it enjoys an especially low level of priority." don't you 
understand?  ;-)

SK

Lee Spenadel wrote:
> It sure would be nice for us to enjoy clean crisp recordings of Jane.
What
> would be a possible time line on this?
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Steph Kerman
> Sent: Friday, April 06, 2007 3:57 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Jane Barbe
>
> I haven't the faintest idea.  Under the present circumstances it enjoys 
> an especially low level of priority.
>
> SK
>
> David Josephson wrote:
>   
>> Fine. When can you, I or we do this?
>>
>> Steph Kerman wrote:
>>   
>>     
>>> David Josephson wrote:
>>>   
>>>     
>>>       
>>>> Is that in the warehouse I know about?
>>>>     
>>>>       
>>>>         
>>> No.
>>>   
>>>     
>>>       
>>>> We have a large assortment of analog magnetic tape repro equipment 
>>>> here. If we can get the transports (tape, drum or belt) running, we 
>>>> can probably extract the sounds recorded on those tracks with good 
>>>> fidelity. That would be an interesting project.
>>>>     
>>>>       
>>>>         
>>> Thanks for offering.  I'm sure there are enough channels of the original

>>> electronics that work just fine to dub all 96 tracks serially.  If not, 
>>> I have appropriate preamps that I could substitute.  If there *are* 
>>> complications they might be related to control of the drive mechanism, 
>>> probably not something you are particularly prepared to deal with any 
>>> better than I am.
>>>   
>>>     
>>>       
>>   
>>     
> _______________________________________________
> 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 jnovack at stromberg-carlson.org  Fri Apr  6 15:46:44 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Fri, 06 Apr 2007 16:46:44 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <4616AF0D.8040902@jps.net>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>	<4615D547.5000501@jps.net>	<4616A1C2.2060501@josephson.com>	<4616A3E8.1010803@jps.net>	<4616A4AC.9020402@josephson.com>	<4616A60F.1050503@jps.net>	<01e901c77888$5b46cb00$11d46100$@com>
	<4616AF0D.8040902@jps.net>
Message-ID: <4616B1B4.8040300@stromberg-carlson.org>

Perhaps Steph's executor will make these available.
Otherwise,  either get in a time machine or find another source

John Novack



Steph Kerman wrote:
> I've been waiting patiently for a good opportunity to use this phrase:  
> What part of "I haven't the faintest idea.  Under the present 
> circumstances it enjoys an especially low level of priority." don't you 
> understand?  ;-)
>
> SK
>
> Lee Spenadel wrote:
>   
>> It sure would be nice for us to enjoy clean crisp recordings of Jane.  What
>> would be a possible time line on this?
>>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
>> Steph Kerman
>> Sent: Friday, April 06, 2007 3:57 PM
>> To: Voice Over IP Tandem for Analog Switches
>> Subject: Re: [VoIP] Jane Barbe
>>
>> I haven't the faintest idea.  Under the present circumstances it enjoys 
>> an especially low level of priority.
>>
>> SK
>>
>> David Josephson wrote:
>>   
>>     
>>> Fine. When can you, I or we do this?
>>>
>>> Steph Kerman wrote:
>>>   
>>>     
>>>       
>>>> David Josephson wrote:
>>>>   
>>>>     
>>>>       
>>>>         
>>>>> Is that in the warehouse I know about?
>>>>>     
>>>>>       
>>>>>         
>>>>>           
>>>> No.
>>>>   
>>>>     
>>>>       
>>>>         
>>>>> We have a large assortment of analog magnetic tape repro equipment 
>>>>> here. If we can get the transports (tape, drum or belt) running, we 
>>>>> can probably extract the sounds recorded on those tracks with good 
>>>>> fidelity. That would be an interesting project.
>>>>>     
>>>>>       
>>>>>         
>>>>>           
>>>> Thanks for offering.  I'm sure there are enough channels of the original 
>>>> electronics that work just fine to dub all 96 tracks serially.  If not, 
>>>> I have appropriate preamps that I could substitute.  If there *are* 
>>>> complications they might be related to control of the drive mechanism, 
>>>> probably not something you are particularly prepared to deal with any 
>>>> better than I am.
>>>>   
>>>>     
>>>>       
>>>>         
>>>   
>>>     
>>>       
>> _______________________________________________
>> 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  Fri Apr  6 15:53:34 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Fri, 06 Apr 2007 16:53:34 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <01eb01c7788c$95b59b50$c120d1f0$@com>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>	<4615D547.5000501@jps.net>	<4616A1C2.2060501@josephson.com>	<4616A3E8.1010803@jps.net>	<4616A4AC.9020402@josephson.com>	<4616A60F.1050503@jps.net>	<01e901c77888$5b46cb00$11d46100$@com>	<4616AF0D.8040902@jps.net>
	<01eb01c7788c$95b59b50$c120d1f0$@com>
Message-ID: <4616B34E.8030406@jps.net>

You're welcome!  "To whom" would be the question....

SK

Lee Spenadel wrote:
> There you go demonstrating your usefulness again.  
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Steph Kerman
> Sent: Friday, April 06, 2007 4:35 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Jane Barbe
>
> I've been waiting patiently for a good opportunity to use this phrase:  
> What part of "I haven't the faintest idea.  Under the present 
> circumstances it enjoys an especially low level of priority." don't you 
> understand?  ;-)
>
> SK
>
> Lee Spenadel wrote:
>   
>> It sure would be nice for us to enjoy clean crisp recordings of Jane.
>>     
> What
>   
>> would be a possible time line on this?
>>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
>> Steph Kerman
>> Sent: Friday, April 06, 2007 3:57 PM
>> To: Voice Over IP Tandem for Analog Switches
>> Subject: Re: [VoIP] Jane Barbe
>>
>> I haven't the faintest idea.  Under the present circumstances it enjoys 
>> an especially low level of priority.
>>
>> SK
>>
>> David Josephson wrote:
>>   
>>     
>>> Fine. When can you, I or we do this?
>>>
>>> Steph Kerman wrote:
>>>   
>>>     
>>>       
>>>> David Josephson wrote:
>>>>   
>>>>     
>>>>       
>>>>         
>>>>> Is that in the warehouse I know about?
>>>>>     
>>>>>       
>>>>>         
>>>>>           
>>>> No.
>>>>   
>>>>     
>>>>       
>>>>         
>>>>> We have a large assortment of analog magnetic tape repro equipment 
>>>>> here. If we can get the transports (tape, drum or belt) running, we 
>>>>> can probably extract the sounds recorded on those tracks with good 
>>>>> fidelity. That would be an interesting project.
>>>>>     
>>>>>       
>>>>>         
>>>>>           
>>>> Thanks for offering.  I'm sure there are enough channels of the original
>>>>         
>
>   
>>>> electronics that work just fine to dub all 96 tracks serially.  If not, 
>>>> I have appropriate preamps that I could substitute.  If there *are* 
>>>> complications they might be related to control of the drive mechanism, 
>>>> probably not something you are particularly prepared to deal with any 
>>>> better than I am.
>>>>   
>>>>     
>>>>       
>>>>         
>>>   
>>>     
>>>       
>> _______________________________________________
>> 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  Fri Apr  6 16:00:06 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Fri, 06 Apr 2007 17:00:06 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <01eb01c7788c$95b59b50$c120d1f0$@com>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>	<4615D547.5000501@jps.net>	<4616A1C2.2060501@josephson.com>	<4616A3E8.1010803@jps.net>	<4616A4AC.9020402@josephson.com>	<4616A60F.1050503@jps.net>	<01e901c77888$5b46cb00$11d46100$@com>	<4616AF0D.8040902@jps.net>
	<01eb01c7788c$95b59b50$c120d1f0$@com>
Message-ID: <4616B4D6.5020505@jps.net>

On further reflection, that strikes me as a particularly ironic question 
coming from someone like yourself who asks for a lot more help on this 
list than he gives to others.  There are a whole host of others on this 
list from whom the question would seem considerably less bizarre.

SK

Lee Spenadel wrote:
> There you go demonstrating your usefulness again.  
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Steph Kerman
> Sent: Friday, April 06, 2007 4:35 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Jane Barbe
>
> I've been waiting patiently for a good opportunity to use this phrase:  
> What part of "I haven't the faintest idea.  Under the present 
> circumstances it enjoys an especially low level of priority." don't you 
> understand?  ;-)
>
> SK
>
> Lee Spenadel wrote:
>   
>> It sure would be nice for us to enjoy clean crisp recordings of Jane.
>>     
> What
>   
>> would be a possible time line on this?
>>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
>> Steph Kerman
>> Sent: Friday, April 06, 2007 3:57 PM
>> To: Voice Over IP Tandem for Analog Switches
>> Subject: Re: [VoIP] Jane Barbe
>>
>> I haven't the faintest idea.  Under the present circumstances it enjoys 
>> an especially low level of priority.
>>
>> SK
>>
>> David Josephson wrote:
>>   
>>     
>>> Fine. When can you, I or we do this?
>>>
>>> Steph Kerman wrote:
>>>   
>>>     
>>>       
>>>> David Josephson wrote:
>>>>   
>>>>     
>>>>       
>>>>         
>>>>> Is that in the warehouse I know about?
>>>>>     
>>>>>       
>>>>>         
>>>>>           
>>>> No.
>>>>   
>>>>     
>>>>       
>>>>         
>>>>> We have a large assortment of analog magnetic tape repro equipment 
>>>>> here. If we can get the transports (tape, drum or belt) running, we 
>>>>> can probably extract the sounds recorded on those tracks with good 
>>>>> fidelity. That would be an interesting project.
>>>>>     
>>>>>       
>>>>>         
>>>>>           
>>>> Thanks for offering.  I'm sure there are enough channels of the original
>>>>         
>
>   
>>>> electronics that work just fine to dub all 96 tracks serially.  If not, 
>>>> I have appropriate preamps that I could substitute.  If there *are* 
>>>> complications they might be related to control of the drive mechanism, 
>>>> probably not something you are particularly prepared to deal with any 
>>>> better than I am.
>>>>   
>>>>     
>>>>       
>>>>         
>>>   
>>>     
>>>       
>> _______________________________________________
>> 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 lee at spenadel.com  Fri Apr  6 16:16:12 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Fri, 6 Apr 2007 17:16:12 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <4616B4D6.5020505@jps.net>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>	<4615D547.5000501@jps.net>	<4616A1C2.2060501@josephson.com>	<4616A3E8.1010803@jps.net>	<4616A4AC.9020402@josephson.com>	<4616A60F.1050503@jps.net>	<01e901c77888$5b46cb00$11d46100$@com>	<4616AF0D.8040902@jps.net>	<01eb01c7788c$95b59b50$c120d1f0$@com>
	<4616B4D6.5020505@jps.net>
Message-ID: <01f901c77890$ce011670$6a034350$@com>

If I didn't understand you and your insecurities I'd might take offense to
your comments.   Last time I checked, this list was for sharing knowledge
when one asks the question.   I didn't know anyone was keeping score.

If you need to discuss this further, you know my email address so we can
take this offline.

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Steph Kerman
Sent: Friday, April 06, 2007 5:00 PM
To: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Jane Barbe

On further reflection, that strikes me as a particularly ironic question 
coming from someone like yourself who asks for a lot more help on this 
list than he gives to others.  There are a whole host of others on this 
list from whom the question would seem considerably less bizarre.

SK

Lee Spenadel wrote:
> There you go demonstrating your usefulness again.  
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Steph Kerman
> Sent: Friday, April 06, 2007 4:35 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Jane Barbe
>
> I've been waiting patiently for a good opportunity to use this phrase:  
> What part of "I haven't the faintest idea.  Under the present 
> circumstances it enjoys an especially low level of priority." don't you 
> understand?  ;-)
>
> SK
>
> Lee Spenadel wrote:
>   
>> It sure would be nice for us to enjoy clean crisp recordings of Jane.
>>     
> What
>   
>> would be a possible time line on this?
>>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
>> Steph Kerman
>> Sent: Friday, April 06, 2007 3:57 PM
>> To: Voice Over IP Tandem for Analog Switches
>> Subject: Re: [VoIP] Jane Barbe
>>
>> I haven't the faintest idea.  Under the present circumstances it enjoys 
>> an especially low level of priority.
>>
>> SK
>>
>> David Josephson wrote:
>>   
>>     
>>> Fine. When can you, I or we do this?
>>>
>>> Steph Kerman wrote:
>>>   
>>>     
>>>       
>>>> David Josephson wrote:
>>>>   
>>>>     
>>>>       
>>>>         
>>>>> Is that in the warehouse I know about?
>>>>>     
>>>>>       
>>>>>         
>>>>>           
>>>> No.
>>>>   
>>>>     
>>>>       
>>>>         
>>>>> We have a large assortment of analog magnetic tape repro equipment 
>>>>> here. If we can get the transports (tape, drum or belt) running, we 
>>>>> can probably extract the sounds recorded on those tracks with good 
>>>>> fidelity. That would be an interesting project.
>>>>>     
>>>>>       
>>>>>         
>>>>>           
>>>> Thanks for offering.  I'm sure there are enough channels of the
original
>>>>         
>
>   
>>>> electronics that work just fine to dub all 96 tracks serially.  If not,

>>>> I have appropriate preamps that I could substitute.  If there *are* 
>>>> complications they might be related to control of the drive mechanism, 
>>>> probably not something you are particularly prepared to deal with any 
>>>> better than I am.
>>>>   
>>>>     
>>>>       
>>>>         
>>>   
>>>     
>>>       
>> _______________________________________________
>> 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 stfkerman at jps.net  Fri Apr  6 16:33:24 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Fri, 06 Apr 2007 17:33:24 -0400
Subject: [VoIP] Jane Barbe
In-Reply-To: <01f901c77890$ce011670$6a034350$@com>
References: <E1HZe5P-0001jk-3C@telstar.blackratchet.org>	<000d01c777f4$b509ea80$0600a8c0@bluegrasspals.com>	<4615D34A.5090409@jps.net>	<000901c77808$ed76b600$0600a8c0@bluegrasspals.com>	<4615D547.5000501@jps.net>	<4616A1C2.2060501@josephson.com>	<4616A3E8.1010803@jps.net>	<4616A4AC.9020402@josephson.com>	<4616A60F.1050503@jps.net>	<01e901c77888$5b46cb00$11d46100$@com>	<4616AF0D.8040902@jps.net>	<01eb01c7788c$95b59b50$c120d1f0$@com>	<4616B4D6.5020505@jps.net>
	<01f901c77890$ce011670$6a034350$@com>
Message-ID: <4616BCA4.8090506@jps.net>

I didn't realize the list's scope included lame attempts at 
psychoanalysis based on scant information!  That's fascinating.   
Thank's for cluing me in to that.  I'll feel more free in the future to 
moderate the on-line group therapy.

The point of the list is indeed to ask questions and share information.  
However that does not include dismissing people as being useless who 
have their own priorities which you know nothing whatever about and 
among which satisfying the whims of others like you does not place very 
high.

SK

Lee Spenadel wrote:
> If I didn't understand you and your insecurities I'd might take offense to
> your comments.   Last time I checked, this list was for sharing knowledge
> when one asks the question.   I didn't know anyone was keeping score.
>
> If you need to discuss this further, you know my email address so we can
> take this offline.
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Steph Kerman
> Sent: Friday, April 06, 2007 5:00 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Jane Barbe
>
> On further reflection, that strikes me as a particularly ironic question 
> coming from someone like yourself who asks for a lot more help on this 
> list than he gives to others.  There are a whole host of others on this 
> list from whom the question would seem considerably less bizarre.
>
> SK
>
> Lee Spenadel wrote:
>   
>> There you go demonstrating your usefulness again.  
>>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
>> Steph Kerman
>> Sent: Friday, April 06, 2007 4:35 PM
>> To: Voice Over IP Tandem for Analog Switches
>> Subject: Re: [VoIP] Jane Barbe
>>
>> I've been waiting patiently for a good opportunity to use this phrase:  
>> What part of "I haven't the faintest idea.  Under the present 
>> circumstances it enjoys an especially low level of priority." don't you 
>> understand?  ;-)
>>
>> SK
>>
>> Lee Spenadel wrote:
>>   
>>     
>>> It sure would be nice for us to enjoy clean crisp recordings of Jane.
>>>     
>>>       
>> What
>>   
>>     
>>>