From jjones3601 at yahoo.com  Sun Jul  1 09:26:51 2007
From: jjones3601 at yahoo.com (john jones)
Date: Sun, 1 Jul 2007 07:26:51 -0700 (PDT)
Subject: [VoIP] OT: Need help w/Linux
Message-ID: <634843.86678.qm@web34314.mail.mud.yahoo.com>

This keyboard business is another reason I switched from a PC to a router to host Asterisk.  

John

----- Original Message ----
From: Duncan Smith <duncan.b.smith at gmail.com>
To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
Sent: Saturday, June 30, 2007 9:50:38 PM
Subject: Re: [VoIP] OT: Need help w/Linux

On Sat, Jun 30, 2007 at 11:58:20AM -0700, Mark Rudholm wrote:
> Worst-case scenario would be that you'd have to leave an old keyboard
> plugged into it.

What I do is take an old keyboard, and get rid of all the keys.  Then
it's just a circuit board on the end of a keyboard cable, but it looks
like a keyboard to the computer.  You can wrap it in electrical tape
or heat-shrink tubing, and put it inside the machine (cable through
the back) to save even more space.

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




From mark at rudholm.com  Sun Jul  1 11:37:45 2007
From: mark at rudholm.com (Mark Rudholm)
Date: Sun, 01 Jul 2007 09:37:45 -0700
Subject: [VoIP] OT: Need help w/Linux
In-Reply-To: <634843.86678.qm@web34314.mail.mud.yahoo.com>
References: <634843.86678.qm@web34314.mail.mud.yahoo.com>
Message-ID: <4687D859.7090107@rudholm.com>

I can definitely think of some reasons to experiment with
Asterisk on my LinkSys (low power, no noise, cheap, small,
it'd just be cool) but keyboard abatement wasn't one of them.
I guess I haven't really dealt with motherboards that don't
let you turn the keyboard presence test off.  Seems like
it'd be easier to just replace the motherboard with one
that isn't quite so brain-dead.

-Mark

john jones wrote:
> This keyboard business is another reason I switched from a PC to a router to host Asterisk.  
> 
> John
> 
> ----- Original Message ----
> From: Duncan Smith <duncan.b.smith at gmail.com>
> To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
> Sent: Saturday, June 30, 2007 9:50:38 PM
> Subject: Re: [VoIP] OT: Need help w/Linux
> 
> On Sat, Jun 30, 2007 at 11:58:20AM -0700, Mark Rudholm wrote:
>> Worst-case scenario would be that you'd have to leave an old keyboard
>> plugged into it.
> 
> What I do is take an old keyboard, and get rid of all the keys.  Then
> it's just a circuit board on the end of a keyboard cable, but it looks
> like a keyboard to the computer.  You can wrap it in electrical tape
> or heat-shrink tubing, and put it inside the machine (cable through
> the back) to save even more space.
> 


From lee at spenadel.com  Sun Jul  1 11:58:58 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Sun, 1 Jul 2007 12:58:58 -0400
Subject: [VoIP] OT: Need help w/Linux
In-Reply-To: <634843.86678.qm@web34314.mail.mud.yahoo.com>
References: <634843.86678.qm@web34314.mail.mud.yahoo.com>
Message-ID: <033d01c7bc01$1dea6c00$59bf4400$@com>

Not sure why this is a problem.  I use an old eMachines PC with CentOS 3
installed on it.  The machine boots normally with no keyboard / mouse
plugged in.  

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
john jones
Sent: Sunday, July 01, 2007 10:27 AM
To: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] OT: Need help w/Linux

This keyboard business is another reason I switched from a PC to a router to
host Asterisk.  

John

----- Original Message ----
From: Duncan Smith <duncan.b.smith at gmail.com>
To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
Sent: Saturday, June 30, 2007 9:50:38 PM
Subject: Re: [VoIP] OT: Need help w/Linux

On Sat, Jun 30, 2007 at 11:58:20AM -0700, Mark Rudholm wrote:
> Worst-case scenario would be that you'd have to leave an old keyboard
> plugged into it.

What I do is take an old keyboard, and get rid of all the keys.  Then
it's just a circuit board on the end of a keyboard cable, but it looks
like a keyboard to the computer.  You can wrap it in electrical tape
or heat-shrink tubing, and put it inside the machine (cable through
the back) to save even more space.

-- 
Duncan Smith  --------\    http://students.washington.edu/f/    /---
  ()  ascii ribbon     \--- Signed/encrypted mail preferred ---/
  /\    campaign    [ against html mail ]  [ support open formats ]
_______________________________________________
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  Sun Jul  1 12:53:10 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 01 Jul 2007 13:53:10 -0400
Subject: [VoIP] OT: Need help w/Linux
In-Reply-To: <033d01c7bc01$1dea6c00$59bf4400$@com>
References: <634843.86678.qm@web34314.mail.mud.yahoo.com>
	<033d01c7bc01$1dea6c00$59bf4400$@com>
Message-ID: <4687EA06.9050602@stromberg-carlson.org>

I have used several different motherboards, considered obsolete by 
todays standards, and DID run into one that, even though Bios settings 
said to ignore, when the machine booted into CentOS it wasn't happy with 
no keyboard and went somewhere I have never seen before.
Adding a keyboard to the machine made the problem go away.
It was a Gateway MB
Pentium III MB's are often available on eBay for next to nothing, and if 
the footprint is standard, easily replaceable.
Also the machine could be booted with a keyboard, then removed. Since 
machines work and survive better when on 24/7 that is another solution.

John Novack


Lee Spenadel wrote:
> Not sure why this is a problem.  I use an old eMachines PC with CentOS 3
> installed on it.  The machine boots normally with no keyboard / mouse
> plugged in.  
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> john jones
> Sent: Sunday, July 01, 2007 10:27 AM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] OT: Need help w/Linux
>
> This keyboard business is another reason I switched from a PC to a router to
> host Asterisk.  
>
> John
>
> ----- Original Message ----
> From: Duncan Smith <duncan.b.smith at gmail.com>
> To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
> Sent: Saturday, June 30, 2007 9:50:38 PM
> Subject: Re: [VoIP] OT: Need help w/Linux
>
> On Sat, Jun 30, 2007 at 11:58:20AM -0700, Mark Rudholm wrote:
>   
>> Worst-case scenario would be that you'd have to leave an old keyboard
>> plugged into it.
>>     
>
> What I do is take an old keyboard, and get rid of all the keys.  Then
> it's just a circuit board on the end of a keyboard cable, but it looks
> like a keyboard to the computer.  You can wrap it in electrical tape
> or heat-shrink tubing, and put it inside the machine (cable through
> the back) to save even more space.
>
>   

-- 
Dog is my co-pilot


From mark at rudholm.com  Sun Jul  1 13:22:22 2007
From: mark at rudholm.com (Mark Rudholm)
Date: Sun, 01 Jul 2007 11:22:22 -0700
Subject: [VoIP] OT: Need help w/Linux
In-Reply-To: <4687EA06.9050602@stromberg-carlson.org>
References: <634843.86678.qm@web34314.mail.mud.yahoo.com>	<033d01c7bc01$1dea6c00$59bf4400$@com>
	<4687EA06.9050602@stromberg-carlson.org>
Message-ID: <4687F0DE.9070702@rudholm.com>

John Novack wrote:
> I have used several different motherboards, considered obsolete by 
> todays standards, and DID run into one that, even though Bios settings 
> said to ignore, when the machine booted into CentOS it wasn't happy with 
> no keyboard and went somewhere I have never seen before.
> Adding a keyboard to the machine made the problem go away.
> It was a Gateway MB

That sounds like an OS issue.  You might be able to solve
it by commenting out the console gettys from inittab.

From jnovack at stromberg-carlson.org  Sun Jul  1 14:22:57 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 01 Jul 2007 15:22:57 -0400
Subject: [VoIP] OT: Need help w/Linux
In-Reply-To: <4687F0DE.9070702@rudholm.com>
References: <634843.86678.qm@web34314.mail.mud.yahoo.com>	<033d01c7bc01$1dea6c00$59bf4400$@com>
	<4687EA06.9050602@stromberg-carlson.org>
	<4687F0DE.9070702@rudholm.com>
Message-ID: <4687FF11.4080501@stromberg-carlson.org>



Mark Rudholm wrote:
> John Novack wrote:
>   
>> I have used several different motherboards, considered obsolete by 
>> todays standards, and DID run into one that, even though Bios settings 
>> said to ignore, when the machine booted into CentOS it wasn't happy with 
>> no keyboard and went somewhere I have never seen before.
>> Adding a keyboard to the machine made the problem go away.
>> It was a Gateway MB
>>     
>
> That sounds like an OS issue.  You might be able to solve
> it by commenting out the console gettys from inittab.
Possibly, though the same configuration on other MB's didn't suffer from 
this problem.
The machine is now in service, so if it's not broken, I will leave it 
alone!!

JN
-- 

Dog is my co-pilot


From watson061502 at ameritech.net  Sun Jul  1 15:22:04 2007
From: watson061502 at ameritech.net (Nathaniel D. Watson)
Date: Sun, 01 Jul 2007 16:22:04 -0400
Subject: [VoIP] OT: Need help w/Linux
In-Reply-To: <4687FF11.4080501@stromberg-carlson.org>
Message-ID: <C2AD852C.C1A3%watson061502@ameritech.net>

Here's an update on my situation:

The machine I was using was a Gateway, which whom I had known to be rather
proprietary in their designs. I found that there were no BIOS settings to
make it ignore the presence of a keyboard, and I confirmed this to be true
by speaking with the shop where I bought the machine (a local place selling
used machines & parts). The guy did let me swap/exchange for a comparably
equipped Compaq, and I got it to work the way I want just fine.

Thanks for everyone's responses & suggestions! My next step is to install
and dive into Asterisk. I'm also awaiting the delivery of a Cisco 3810 --
the one which I asked about a few days ago.

(John N.: I did follow your advice, and the guy accepted my best-offer
price.)


Thanks,

Nathan 



From jjones3601 at yahoo.com  Sun Jul  1 15:52:13 2007
From: jjones3601 at yahoo.com (john jones)
Date: Sun, 1 Jul 2007 13:52:13 -0700 (PDT)
Subject: [VoIP] OT: Need help w/Linux
Message-ID: <20070701205213.93377.qmail@web34309.mail.mud.yahoo.com>

Nathan,

Let me know what software image the the 3810 has when it arrives ( just type in dir from a command prompt )

You may want to get a console cable since most of these routers don't come with them.

http://cgi.ebay.com/Cisco-Console-Cable-72-3383-01_W0QQitemZ180133729957QQihZ008QQcategoryZ51264QQrdZ1QQcmdZViewItem




John

----- Original Message ----
From: Nathaniel D. Watson <watson061502 at ameritech.net>
To: voip at ckts.info
Sent: Sunday, July 1, 2007 4:22:04 PM
Subject: Re: [VoIP] OT: Need help w/Linux

Here's an update on my situation:

The machine I was using was a Gateway, which whom I had known to be rather
proprietary in their designs. I found that there were no BIOS settings to
make it ignore the presence of a keyboard, and I confirmed this to be true
by speaking with the shop where I bought the machine (a local place selling
used machines & parts). The guy did let me swap/exchange for a comparably
equipped Compaq, and I got it to work the way I want just fine.

Thanks for everyone's responses & suggestions! My next step is to install
and dive into Asterisk. I'm also awaiting the delivery of a Cisco 3810 --
the one which I asked about a few days ago.

(John N.: I did follow your advice, and the guy accepted my best-offer
price.)


Thanks,

Nathan 


_______________________________________________
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  Sun Jul  1 16:46:50 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 01 Jul 2007 17:46:50 -0400
Subject: [VoIP] OT: Need help w/Linux
In-Reply-To: <C2AD852C.C1A3%watson061502@ameritech.net>
References: <C2AD852C.C1A3%watson061502@ameritech.net>
Message-ID: <468820CA.802@stromberg-carlson.org>

Great
While you are waiting, you may want to consider doing a new install with 
CentOS 3,4, or even 5. RH 9 is OK, but very limited.
You also haven't yet mentioned the CPU speed and memory.
Another issue which MAY come up is the type of USB hardware. I seem to 
remember that, if you don't have at least one FXO or FXS card, you will 
need to compile Asterisk with ztdummy enabled, which requires a small 
edit to the make file, that gives you a timing source needed for certain 
applications within Asterisk.

John Novack


Nathaniel D. Watson wrote:
> Here's an update on my situation:
>
> The machine I was using was a Gateway, which whom I had known to be rather proprietary in their designs. I found that there were no BIOS settings to make it ignore the presence of a keyboard, and I confirmed this to be true by speaking with the shop where I bought the machine (a local place selling used machines & parts). The guy did let me swap/exchange for a comparably
> equipped Compaq, and I got it to work the way I want just fine.
>
> Thanks for everyone's responses & suggestions! My next step is to install
> and dive into Asterisk. I'm also awaiting the delivery of a Cisco 3810 --
> the one which I asked about a few days ago.
>
> (John N.: I did follow your advice, and the guy accepted my best-offer
> price.)
>
>
> Thanks,
>
> Nathan 
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From watson061502 at ameritech.net  Sun Jul  1 17:15:01 2007
From: watson061502 at ameritech.net (Nathaniel D. Watson)
Date: Sun, 01 Jul 2007 18:15:01 -0400
Subject: [VoIP] OT: Need help w/Linux
In-Reply-To: <468820CA.802@stromberg-carlson.org>
Message-ID: <C2AD9FA5.C1AC%watson061502@ameritech.net>

My machine is a Compaq Deskpro EXm; P3 @ 733 MHz, 128 MB RAM (but I have
old/spare RAM I can add if needed), 10 GB HD. Last night I downloaded CentOS
5, so I have that to install if I need to switch to it.

Nathan


on 7/1/07 5:46 PM, John Novack at jnovack at stromberg-carlson.org wrote:

> Great
> While you are waiting, you may want to consider doing a new install with
> CentOS 3,4, or even 5. RH 9 is OK, but very limited.
> You also haven't yet mentioned the CPU speed and memory.
> Another issue which MAY come up is the type of USB hardware. I seem to
> remember that, if you don't have at least one FXO or FXS card, you will
> need to compile Asterisk with ztdummy enabled, which requires a small
> edit to the make file, that gives you a timing source needed for certain
> applications within Asterisk.
> 
> John Novack
> 
> 
> Nathaniel D. Watson wrote:
>> Here's an update on my situation:
>> 
>> The machine I was using was a Gateway, which whom I had known to be rather
>> proprietary in their designs. I found that there were no BIOS settings to
>> make it ignore the presence of a keyboard, and I confirmed this to be true by
>> speaking with the shop where I bought the machine (a local place selling used
>> machines & parts). The guy did let me swap/exchange for a comparably
>> equipped Compaq, and I got it to work the way I want just fine.
>> 
>> Thanks for everyone's responses & suggestions! My next step is to install
>> and dive into Asterisk. I'm also awaiting the delivery of a Cisco 3810 --
>> the one which I asked about a few days ago.
>> 
>> (John N.: I did follow your advice, and the guy accepted my best-offer
>> price.)
>> 
>> 
>> Thanks,
>> 
>> Nathan 
>> 
>> 
>> _______________________________________________
>> 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  Sun Jul  1 22:16:56 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 01 Jul 2007 23:16:56 -0400
Subject: [VoIP] OT: Need help w/Linux
In-Reply-To: <C2AD9FA5.C1AC%watson061502@ameritech.net>
References: <C2AD9FA5.C1AC%watson061502@ameritech.net>
Message-ID: <46886E28.5090806@stromberg-carlson.org>

You may find the 10 gig HD a little on the tight side.
Since I am not really a Linux guru, I tell the install to install 
everything, then turn off services not needed once it is up and running. 
In the past when I didn't do that, somewhere along the way, SOMETHING 
would complain about some missing file or other, then I would have to go 
searching around for what was missing, try and install it and hope it 
got into the right place in the directory structure.
CentOS 3 with asterisk and add ons has plenty of left over space for 
logs and such on a 20 Gig HD, and from what I have seen so does CentOS 
4. Cant say about CentOS 5 though. No experience with that ( yet )
Used 20-40 Gig hard drives are available on eBay for 20-30 bucks

John Novack


Nathaniel D. Watson wrote:
> My machine is a Compaq Deskpro EXm; P3 @ 733 MHz, 128 MB RAM (but I have
> old/spare RAM I can add if needed), 10 GB HD. Last night I downloaded CentOS
> 5, so I have that to install if I need to switch to it.
>
> Nathan
>
>
> on 7/1/07 5:46 PM, John Novack at jnovack at stromberg-carlson.org wrote:
>
>   
>> Great
>> While you are waiting, you may want to consider doing a new install with
>> CentOS 3,4, or even 5. RH 9 is OK, but very limited.
>> You also haven't yet mentioned the CPU speed and memory.
>> Another issue which MAY come up is the type of USB hardware. I seem to
>> remember that, if you don't have at least one FXO or FXS card, you will
>> need to compile Asterisk with ztdummy enabled, which requires a small
>> edit to the make file, that gives you a timing source needed for certain
>> applications within Asterisk.
>>
>> John Novack
>>
>>
>> Nathaniel D. Watson wrote:
>>     
>>> Here's an update on my situation:
>>>
>>> The machine I was using was a Gateway, which whom I had known to be rather
>>> proprietary in their designs. I found that there were no BIOS settings to
>>> make it ignore the presence of a keyboard, and I confirmed this to be true by
>>> speaking with the shop where I bought the machine (a local place selling used
>>> machines & parts). The guy did let me swap/exchange for a comparably
>>> equipped Compaq, and I got it to work the way I want just fine.
>>>
>>> Thanks for everyone's responses & suggestions! My next step is to install
>>> and dive into Asterisk. I'm also awaiting the delivery of a Cisco 3810 --
>>> the one which I asked about a few days ago.
>>>
>>> (John N.: I did follow your advice, and the guy accepted my best-offer
>>> price.)
>>>
>>>
>>> Thanks,
>>>
>>> Nathan 
>>>
>>>
>>> _______________________________________________
>>> 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/
>
>   

-- 
Dog is my co-pilot


From hockd at dteenergy.com  Mon Jul  2 03:57:21 2007
From: hockd at dteenergy.com (Dennis D Hock)
Date: Mon, 2 Jul 2007 04:57:21 -0400
Subject: [VoIP] New member now online
Message-ID: <OF3C2028BA.ACF2AA40-ON8525730C.00313232-8525730C.0031323B@dteenergy.com>


Dimitri,

Welcome to the group again.  I dare say you are probably the youngest
switcher here.  Congrats on just having graduated from high school.  Where
do you plan on going to pursue your interest in obtaining a job in Telecom?

I am 54 years old and have worked in the "Enterprise" side of Telephony all
my carrer life.  Just completed 28 years at the same firm in the same area,
telecom.  I started as an Assistant Engineer in 1979 and have worked my way
up to Principal Engineer.  I work for a large Midwest Energy company DTE
Energy.  probably a dinosaur in what is becoming your world, smae place for
28 years with increasing responsibilities/  Must be womething wrong with
me.  ;-)

Our voice network is composed of some 42 switches, 6 of which are tandems,
some 22,000 stations spread mostly over SE Michigan.  Telecom is still a
very interesting job and a good choice in spite of the movement toward the
converged network.  The data folks sometimes think voice is very simple to
do but don't always knwo how.  I am all for the convergence its just that
after 28 years the politics starts to wear on you.

Learn to keep one foot in each house and you can make yourself valuable to
your employer.  CCisco is the big dog right now but I don't think they will
remain for very long.  They have too much to learn and are not very open
minded.  They don't want to hear about how and why things have been done
only their attempt to redefine some of the basic tenants.  Keep your ear to
the ground and an open mind and drink up everything you can about
everybodies equipment.  There just might be a better way to do something
and you may be the one to offer it to your employer.  Plus there is the
pride of working on such a large network and watching it grow.  Money is
nice but as the old saw goes it isn't everything!

Anyway, I am here at this time of the morning for three reasons, 1) I love
what I do, 2) It is the only time I can get anything done, 3) I am stupid.

Welcome aboard again and enjoy the group.  A wealth of knowledge exists
here.

Dennis Hock

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


To: voip at ckts.info
From: Dimitri Ressetar <mitya89 at aim.com>
Sent by: voip-bounces at ckts.info
Date: 07/01/2007 12:45AM
Subject: [VoIP] New member now online

Hello everyone.

I've posted on here once before but I thought I should introduce myself
here. My name is Dimitri Ressetar, I live in Harrisburg, Pennsylvania,
and I found out about this network last year when searching around the
internet. I was experimenting with Asterisk at Home at the time and was
amazed by the network that you all put together. I've been interested in
telephones and how the telephone network works since I was a kid, and
finding so much cool information on the internet in the past few years
has been awesome...reading about the blue-boxing and such really shows
how interesting the telephone network is/was. I just graduated from high
school this past May, so I'm guessing I'm one of the youngest people on
the network. I don't have/had a job in the telecommunications industry
(at least not yet!), I'm just really interested on how it all works.
Even though I have no electromechanical switch of my own, I contacted
Greg last November and he connected me to the network with office code
347 (for my initials DGR). I was experimenting around with Asterisk and
the cnet throughout the school year, and more recently (March/April),
switched from using Asterisk at Home on a really old Cyrix 686 to using
Asterisk installed on my Ubuntu Pentium II server (built from old spare
parts).

My server originally was only powered on when I was using it, since it
was too buggy to keep going all the time. This caused some problems,
especially since calling back numbers from my exchange would just
connect to nowhere. (Sorry John Novack!) By the way, checking my CDR, I
found out that "ALDERDICE DOUG" <3663212> was able to get through a long
time ago and heard my French/English Out-of-Service intercept from
Quebec, Canada, an old recording I downloaded.

Now I think I have my server running well enough to keep it on 24x7. I
have ddclient polling my router to keep the dyndns accounts updated, and
the server and router are plugged into my battery backup so they survive
the power blips we have here during the summer. I also have some PSTN
connectivity working thru Asterisk in addition to cnet, but I'm still
experimenting with that. I'll have to contact Greg to get a username and
password to put listings into the phone book, but if you'd like to call
here are some numbers that work: 347-0001 has a short informational
announcement, and 347-1601 thru 1609 are my 'dial-a-disc' lines, which
play a few tunes imported from my music collection. My IP-Phone is
connected to 347-9454, and you're welcome to call it anytime before
midnight (eastern time). If I'm not there, it will go to Asterisk
voicemail. (Please also note that I don't have all the standard cnet
test numbers (time of day, etc.) working yet...hopefully I will code
them in soon.)

I'll be away for the next few days, then back home for a few days, then
gone for a week, but I'll keep the server running while I'm gone to see
how it works connecting remotely. I'll still be checking my e-mail, so
let me know either on the list or off if you notice any more problems
with my server and stuff (I think the caller ID issue is fixed, though
I'll still have to check that again sometime). My e-mail address is
dimitri at ressetar dot com, but I use my aim account mitya89 at aim.com
for the list to avoid getting spam.

Dimitri Ressetar
cnet 347-9454
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voipProject Web Page:
http://www.ckts.info/

From mzaiontz at bellsouth.net  Mon Jul  2 11:39:20 2007
From: mzaiontz at bellsouth.net (Michael Zaiontz)
Date: Mon, 2 Jul 2007 11:39:20 -0500
Subject: [VoIP] Asterisk error message from Entensions.conf
Message-ID: <001001c7bcc7$8a1d0560$6400a8c0@zaiontz1>

I'm getting the following error message when I try to dial a test number on cnet:

WARNING[3884]: pbx.c:1797 pbx_extension_helper: No application ' '  for extension (macro-dialcnet, s, 4)

The line it is complaining about is:

exten  => s,4(startloop),Set(ENUM=${ENUMLOOKUP(${ARG1},ALL,,std.ckts.info)})

I have the latest asterisk and Centos.  
I used the configuration tools on ckts.info to generate the base config files.

Any ideas?

Mike Zaiontz

From g4vft at btinternet.com  Mon Jul  2 12:16:45 2007
From: g4vft at btinternet.com (Jonathan Kay)
Date: Mon, 02 Jul 2007 18:16:45 +0100
Subject: [VoIP] Asterisk error message from Entensions.conf
In-Reply-To: <001001c7bcc7$8a1d0560$6400a8c0@zaiontz1>
References: <001001c7bcc7$8a1d0560$6400a8c0@zaiontz1>
Message-ID: <468932FD.9070001@btinternet.com>

Michael Zaiontz wrote:
> I'm getting the following error message when I try to dial a test number on cnet:
>
> WARNING[3884]: pbx.c:1797 pbx_extension_helper: No application ' '  for extension (macro-dialcnet, s, 4)
>
> The line it is complaining about is:
>
> exten  => s,4(startloop),Set(ENUM=${ENUMLOOKUP(${ARG1},ALL,,std.ckts.info)})
>
> I have the latest asterisk and Centos.  
> I used the configuration tools on ckts.info to generate the base config files.
>
> Any ideas?
>   
Hi Michael,
The latest version of Asterisk, has an extra parameter in its ENUMLOOKUP 
function.
So, you need an extra comma, between ALL and std.ckts.info
A friend of mine here in the UK, Andy Coleman, found he needed to 
specify '1' to use the first choice lookup, also.

exten => s,4(startloop),Set(ENUM=${ENUMLOOKUP(${ARG1},ALL,,1,std.ckts.info)})

HTH

Jon Kay

From mzaiontz at bellsouth.net  Mon Jul  2 12:58:47 2007
From: mzaiontz at bellsouth.net (Michael Zaiontz)
Date: Mon, 2 Jul 2007 12:58:47 -0500
Subject: [VoIP] Asterisk error message from Entensions.conf
References: <001001c7bcc7$8a1d0560$6400a8c0@zaiontz1>
	<468932FD.9070001@btinternet.com>
Message-ID: <004601c7bcd2$a3d926e0$6400a8c0@zaiontz1>

Jon,

Thanks for the speedy reply. I tried what you suggested, but I have the same 
results. I double checked to make sure Asterisk picked up my changes and it 
did, but I get the same error message.

Mike
----- Original Message ----- 
From: "Jonathan Kay" <g4vft at btinternet.com>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Monday, July 02, 2007 12:16 PM
Subject: Re: [VoIP] Asterisk error message from Entensions.conf


> Michael Zaiontz wrote:
>> I'm getting the following error message when I try to dial a test number 
>> on cnet:
>>
>> WARNING[3884]: pbx.c:1797 pbx_extension_helper: No application ' '  for 
>> extension (macro-dialcnet, s, 4)
>>
>> The line it is complaining about is:
>>
>> exten  => 
>> s,4(startloop),Set(ENUM=${ENUMLOOKUP(${ARG1},ALL,,std.ckts.info)})
>>
>> I have the latest asterisk and Centos.
>> I used the configuration tools on ckts.info to generate the base config 
>> files.
>>
>> Any ideas?
>>
> Hi Michael,
> The latest version of Asterisk, has an extra parameter in its ENUMLOOKUP
> function.
> So, you need an extra comma, between ALL and std.ckts.info
> A friend of mine here in the UK, Andy Coleman, found he needed to
> specify '1' to use the first choice lookup, also.
>
> exten => 
> s,4(startloop),Set(ENUM=${ENUMLOOKUP(${ARG1},ALL,,1,std.ckts.info)})
>
> HTH
>
> Jon Kay
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/ 


From mzaiontz at bellsouth.net  Mon Jul  2 14:13:16 2007
From: mzaiontz at bellsouth.net (Michael Zaiontz)
Date: Mon, 2 Jul 2007 14:13:16 -0500
Subject: [VoIP] Asterisk error message from Entensions.conf
References: <001001c7bcc7$8a1d0560$6400a8c0@zaiontz1><468932FD.9070001@btinternet.com>
	<004601c7bcd2$a3d926e0$6400a8c0@zaiontz1>
Message-ID: <006301c7bcdd$0b51c110$6400a8c0@zaiontz1>

Found it!!

My fault. I need to get new glasses.  One too many commas.
Jon, I did need your fix after all

Thanks all. One step closer to joining the net.

Mike Zaiontz
----- Original Message ----- 
From: "Michael Zaiontz" <mzaiontz at bellsouth.net>
To: <g4vft at btinternet.com>; "Voice Over IP Tandem for Analog Switches" 
<voip at ckts.info>
Sent: Monday, July 02, 2007 12:58 PM
Subject: Re: [VoIP] Asterisk error message from Entensions.conf


> Jon,
>
> Thanks for the speedy reply. I tried what you suggested, but I have the 
> same
> results. I double checked to make sure Asterisk picked up my changes and 
> it
> did, but I get the same error message.
>
> Mike
> ----- Original Message ----- 
> From: "Jonathan Kay" <g4vft at btinternet.com>
> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> Sent: Monday, July 02, 2007 12:16 PM
> Subject: Re: [VoIP] Asterisk error message from Entensions.conf
>
>
>> Michael Zaiontz wrote:
>>> I'm getting the following error message when I try to dial a test number
>>> on cnet:
>>>
>>> WARNING[3884]: pbx.c:1797 pbx_extension_helper: No application ' '  for
>>> extension (macro-dialcnet, s, 4)
>>>
>>> The line it is complaining about is:
>>>
>>> exten  =>
>>> s,4(startloop),Set(ENUM=${ENUMLOOKUP(${ARG1},ALL,,std.ckts.info)})
>>>
>>> I have the latest asterisk and Centos.
>>> I used the configuration tools on ckts.info to generate the base config
>>> files.
>>>
>>> Any ideas?
>>>
>> Hi Michael,
>> The latest version of Asterisk, has an extra parameter in its ENUMLOOKUP
>> function.
>> So, you need an extra comma, between ALL and std.ckts.info
>> A friend of mine here in the UK, Andy Coleman, found he needed to
>> specify '1' to use the first choice lookup, also.
>>
>> exten =>
>> s,4(startloop),Set(ENUM=${ENUMLOOKUP(${ARG1},ALL,,1,std.ckts.info)})
>>
>> HTH
>>
>> Jon Kay
>> _______________________________________________
>> 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 g4vft at btinternet.com  Mon Jul  2 14:25:01 2007
From: g4vft at btinternet.com (Jonathan Kay)
Date: Mon, 02 Jul 2007 20:25:01 +0100
Subject: [VoIP] Asterisk error message from Entensions.conf
In-Reply-To: <006301c7bcdd$0b51c110$6400a8c0@zaiontz1>
References: <001001c7bcc7$8a1d0560$6400a8c0@zaiontz1><468932FD.9070001@btinternet.com>	<004601c7bcd2$a3d926e0$6400a8c0@zaiontz1>
	<006301c7bcdd$0b51c110$6400a8c0@zaiontz1>
Message-ID: <4689510D.8050709@btinternet.com>

Well done Michael.
I won't claim any credit for finding the fix. Just repeating what I had 
heard.

JonK


From WATSON061502 at ameritech.net  Mon Jul  2 17:12:01 2007
From: WATSON061502 at ameritech.net (Nathan Watson)
Date: Mon, 02 Jul 2007 14:12:01 -0800
Subject: [VoIP] Question about Asterisk & Wireless IP Phones
Message-ID: <AA-B49395EB7C19126072BA4DDF7EB35A74-ZZ@fipprd24>

Hello,

I am wondering if it would be possible to use a wireless IP phone in 
conjunction 
with Asterisk such that I could take the wireless IP phone and use it 
anywhere I can 
get wireless internet access (to subsequently use my Asterisk setup).

I do not know the details of the wireless IP phones, but I've seen them on 
the 
shelves at computer stores. I don't mean to ask a trivial question, but I'm 
at a point 
with VoIP that I don't know what I don't know. I'm wondering if there are 
any 
wireless handsets that are "open" to custom configuring.

If this is possible, due to the increasing presence of wireless hot spots, 
it would be 
almost like a cellphone without a monthly fee (that is, a fee beyond the 
obvious 
costs of maintaining your own network and computer equipment).

Thanks,

Nathan


From dfroula at sbcglobal.net  Mon Jul  2 16:21:58 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Mon, 2 Jul 2007 14:21:58 -0700 (PDT)
Subject: [VoIP] Question about Asterisk & Wireless IP Phones
In-Reply-To: <AA-B49395EB7C19126072BA4DDF7EB35A74-ZZ@fipprd24>
Message-ID: <17998.83229.qm@web83205.mail.mud.yahoo.com>

I've beeen trying to configure my HP Ipaq Pocket PC
with wireless LAN capability to do just that. I
already use it in the house on the local wireless LAN
as a cordless VOIP phone with no problems, but
configuring the client to work on an outside LAN is
causing problems. The client I am using is pretty
buggy, but the best available at the moment (Express
Talk by NCH for Pocket PC).

I seems I should be able to use SIP to connect
directly to my server using the DynDNS address, but am
having problems getting this to work.

I'll report on any success with outside access.

Best,

Don


--- Nathan Watson <WATSON061502 at ameritech.net> wrote:

> Hello,
> 
> I am wondering if it would be possible to use a
> wireless IP phone in 
> conjunction 
> with Asterisk such that I could take the wireless IP
> phone and use it 
> anywhere I can 
> get wireless internet access (to subsequently use my
> Asterisk setup).
> 
> I do not know the details of the wireless IP phones,
> but I've seen them on 
> the 
> shelves at computer stores. I don't mean to ask a
> trivial question, but I'm 
> at a point 
> with VoIP that I don't know what I don't know. I'm
> wondering if there are 
> any 
> wireless handsets that are "open" to custom
> configuring.
> 
> If this is possible, due to the increasing presence
> of wireless hot spots, 
> it would be 
> almost like a cellphone without a monthly fee (that
> is, a fee beyond the 
> obvious 
> costs of maintaining your own network and computer
> equipment).
> 
> Thanks,
> 
> Nathan
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From ian at uax.org.uk  Mon Jul  2 16:23:34 2007
From: ian at uax.org.uk (Ian Jolly)
Date: Mon, 2 Jul 2007 22:23:34 +0100
Subject: [VoIP] Question about Asterisk & Wireless IP Phones
References: <AA-B49395EB7C19126072BA4DDF7EB35A74-ZZ@fipprd24>
Message-ID: <026f01c7bcef$3f5e0380$0c01a8c0@acer1dd0bbc6d0>

Hi Nathan

I've got a Linksys 330 http://www.provu.co.uk/linksys_WIP330.html  WiFi 
phone on +44 352 2121 on CNET .  I use it mainly at home but have found I 
can walk around the local town where there are plenty  of businesses/private 
houses with no security on their routers and hence I can make/receive calls 
on my WiFi phone !!

Most of the cheaper WiFi phones are designed for 'Skype only' - steer clear 
of those but there are plenty of SIP WiFi phones that could be set up on 
CNET.

Ian Jolly

+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: "Nathan Watson" <WATSON061502 at ameritech.net>
To: <voip at ckts.info>
Sent: Monday, July 02, 2007 11:12 PM
Subject: [VoIP] Question about Asterisk & Wireless IP Phones


> Hello,
>
> I am wondering if it would be possible to use a wireless IP phone in
> conjunction
> with Asterisk such that I could take the wireless IP phone and use it
> anywhere I can
> get wireless internet access (to subsequently use my Asterisk setup).
>
> I do not know the details of the wireless IP phones, but I've seen them on
> the
> shelves at computer stores. I don't mean to ask a trivial question, but 
> I'm
> at a point
> with VoIP that I don't know what I don't know. I'm wondering if there are
> any
> wireless handsets that are "open" to custom configuring.
>
> If this is possible, due to the increasing presence of wireless hot spots,
> it would be
> almost like a cellphone without a monthly fee (that is, a fee beyond the
> obvious
> costs of maintaining your own network and computer equipment).
>
> Thanks,
>
> Nathan
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.5.476 / Virus Database: 269.9.14/883 - Release Date: 01/07/2007 
> 12:19
>
> 


From mzaiontz at bellsouth.net  Mon Jul  2 16:38:38 2007
From: mzaiontz at bellsouth.net (Michael Zaiontz)
Date: Mon, 2 Jul 2007 16:38:38 -0500
Subject: [VoIP] Ti and Asterisk
Message-ID: <00d901c7bcf1$5a36a7a0$6400a8c0@zaiontz1>

Hi all!

You can tell I have too much time on my hands by the number of posts I've made lately.

I've been dying to try something with T1. There's a Adtran 850 with 4 FXS cards at a good price on ebay.  Am I right in thinking all I'll need is the 850 and a T1 card for my Asterisk box?  Till I get my step switch up and running, at least I'll have some extra extensions. (and something else to play with).

Thanks,
Mike

From lee at spenadel.com  Mon Jul  2 16:56:28 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Mon, 2 Jul 2007 17:56:28 -0400
Subject: [VoIP] Ti and Asterisk
In-Reply-To: <00d901c7bcf1$5a36a7a0$6400a8c0@zaiontz1>
References: <00d901c7bcf1$5a36a7a0$6400a8c0@zaiontz1>
Message-ID: <049501c7bcf3$d8748c20$895da460$@com>

Mike,

I purchased an Adtran 850 with 4 FXS and one FXO.  You will need FXO ports
to connect to your Step switch.  I connected my SxS to Asterisk using an
extension off of the SxS, therefore I needed an FXO to do it.  The T1 ends
up being very inexpensive on a port by port basis and I've ended up
deploying more Asterisk extensions that way.  So your decision to go Adtran
is not a bad one.

Lee

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Michael Zaiontz
Sent: Monday, July 02, 2007 5:39 PM
To: Voice Over IP Tandem for Analog Switches
Subject: [VoIP] Ti and Asterisk

Hi all!

You can tell I have too much time on my hands by the number of posts I've
made lately.

I've been dying to try something with T1. There's a Adtran 850 with 4 FXS
cards at a good price on ebay.  Am I right in thinking all I'll need is the
850 and a T1 card for my Asterisk box?  Till I get my step switch up and
running, at least I'll have some extra extensions. (and something else to
play with).

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


From hockd at dteenergy.com  Tue Jul  3 04:12:42 2007
From: hockd at dteenergy.com (Dennis D Hock)
Date: Tue, 3 Jul 2007 05:12:42 -0400
Subject: [VoIP] Ti and Asterisk
Message-ID: <OFE1E070EA.BFB79477-ON8525730D.00329A41-8525730D.00329A4B@dteenergy.com>


Mike,

Adtran makes good equipment and will serve you well.  As has been said you
will need an FXO to tie to your other switch but all in good time.  I
believe all the recentAdtran documentation is on line making it easy to
access and use.  Od course as you said you will ned to terminate the T1 ona
dedsicated board / card or with a multiprocess devicesuch as the Cisco 3810
also on the eplace .  John and others can tell you more about the
requirements for the 3810 in terms of memory and release etc.

Good luck with the eplace.

Dennis Hock

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


To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
From: "Michael Zaiontz" <mzaiontz at bellsouth.net>
Sent by: voip-bounces at ckts.info
Date: 07/02/2007 05:38PM
Subject: [VoIP] Ti and Asterisk

Hi all!

You can tell I have too much time on my hands by the number of posts I've
made lately.

I've been dying to try something with T1. There's a Adtran 850 with 4 FXS
cards at a good price on ebay.  Am I right in thinking all I'll need is the
850 and a T1 card for my Asterisk box?  Till I get my step switch up and
running, at least I'll have some extra extensions. (and something else to
play with).

Thanks,
Mike
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voipProject Web Page:
http://www.ckts.info/

From mzaiontz at bellsouth.net  Wed Jul  4 11:49:24 2007
From: mzaiontz at bellsouth.net (Michael Zaiontz)
Date: Wed, 4 Jul 2007 11:49:24 -0500
Subject: [VoIP] Strange error message
Message-ID: <006501c7be5b$46f34d40$6400a8c0@zaiontz1>

Hi All!

I'm one or two steps closer to re-joining cnet.

When I attemp to call a local number on my switch, it ffails with this message:

called guest at twinbrook.dnsdojo.net/15255427
NOTICE[2674]: chan_iax2.c:934 Socket_process: Rejected connect attemp from 192.168.0.1, Request  '15255427 at default ' does not exist.

WARNING[2675]: chan_iax2.c:7181 Socket process: Call rejected by 68.222.56.129 No such context/extension

The 192.168.0.1 address is my router. The 68.222.56.129 is my wan address.
The asterisk box has a fixed IP address.

The extension does exist. I do have the correct port forwarded.  

I do have a guest account in my iax.conf file.

I can call test numbers on other switches in cnet.

What do I not see here?

Mike

From jjones3601 at yahoo.com  Wed Jul  4 12:47:38 2007
From: jjones3601 at yahoo.com (john jones)
Date: Wed, 4 Jul 2007 10:47:38 -0700 (PDT)
Subject: [VoIP] Strange error message
Message-ID: <860105.8056.qm@web34312.mail.mud.yahoo.com>

Do you need to strip off the 1 or do you have 
some thing like this in extensions.conf

[cnet-in]                                                                                 
exten => _15255427,1,Goto(internal,5427,1)     


John


----- Original Message ----
From: Michael Zaiontz <mzaiontz at bellsouth.net>
To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
Sent: Wednesday, July 4, 2007 12:49:24 PM
Subject: [VoIP] Strange error message

Hi All!

I'm one or two steps closer to re-joining cnet.

When I attemp to call a local number on my switch, it ffails with this message:

called guest at twinbrook.dnsdojo.net/15255427
NOTICE[2674]: chan_iax2.c:934 Socket_process: Rejected connect attemp from 192.168.0.1, Request  '15255427 at default ' does not exist.

WARNING[2675]: chan_iax2.c:7181 Socket process: Call rejected by 68.222.56.129 No such context/extension

The 192.168.0.1 address is my router. The 68.222.56.129 is my wan address.
The asterisk box has a fixed IP address.

The extension does exist. I do have the correct port forwarded.  

I do have a guest account in my iax.conf file.

I can call test numbers on other switches in cnet.

What do I not see here?

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




From dfroula at sbcglobal.net  Wed Jul  4 12:51:50 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Wed, 4 Jul 2007 10:51:50 -0700 (PDT)
Subject: [VoIP] Strange error message
In-Reply-To: <006501c7be5b$46f34d40$6400a8c0@zaiontz1>
Message-ID: <765588.61814.qm@web83204.mail.mud.yahoo.com>

It looks like you are missing a 1-525-5427 extension
in your extensions.conf "default" context.

Actually, a better way would be to create a new
"guest" context in extensions.conf and put the
extension there. I think Asterisk is trying to find an
extension in the [guest] context, since the incoming
call is going to guest at twinbrook.dnsdojo.net/15255427.
Failing that, it is trying the "default" context.

Did you put the leading "1" in your extension
definition?

Don

--- Michael Zaiontz <mzaiontz at bellsouth.net> wrote:

> Hi All!
> 
> I'm one or two steps closer to re-joining cnet.
> 
> When I attemp to call a local number on my switch,
> it ffails with this message:
> 
> called guest at twinbrook.dnsdojo.net/15255427
> NOTICE[2674]: chan_iax2.c:934 Socket_process:
> Rejected connect attemp from 192.168.0.1, Request 
> '15255427 at default ' does not exist.
> 
> WARNING[2675]: chan_iax2.c:7181 Socket process: Call
> rejected by 68.222.56.129 No such context/extension
> 
> The 192.168.0.1 address is my router. The
> 68.222.56.129 is my wan address.
> The asterisk box has a fixed IP address.
> 
> The extension does exist. I do have the correct port
> forwarded.  
> 
> I do have a guest account in my iax.conf file.
> 
> I can call test numbers on other switches in cnet.
> 
> What do I not see here?
> 
> Mike
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From jjones3601 at yahoo.com  Wed Jul  4 12:54:08 2007
From: jjones3601 at yahoo.com (john jones)
Date: Wed, 4 Jul 2007 10:54:08 -0700 (PDT)
Subject: [VoIP] Strange error message
Message-ID: <203706.16072.qm@web34310.mail.mud.yahoo.com>

Here is my debug output.


OpenWrt*CLI> 
    -- Executing Macro("SIP/192.168.0.99-100745e0", "dialcnet|15255427") in new stack
    -- Executing Set("SIP/192.168.0.99-100745e0", "E164NETWORKS=std.ckts.info") in new stack
    -- Executing Set("SIP/192.168.0.99-100745e0", "CALLERID(name)=John Jones") in new stack
    -- Executing Set("SIP/192.168.0.99-100745e0", "CALLERID(number)=687-7700") in new stack
    -- Executing GotoIf("SIP/192.168.0.99-100745e0", "0?startloop") in new stack
    -- Executing Set("SIP/192.168.0.99-100745e0", "ARG1=+15255427") in new stack
    -- Executing Set("SIP/192.168.0.99-100745e0", "ENUM=iax2:guest at twinbrook.dnsdojo.net/15255427") in new stack
    -- Executing GotoIf("SIP/192.168.0.99-100745e0", "0?continue") in new stack
    -- Executing GotoIf("SIP/192.168.0.99-100745e0", "0?sipuri") in new stack
    -- Executing GotoIf("SIP/192.168.0.99-100745e0", "1?iaxuri") in new stack
    -- Goto (macro-dialcnet,s,16)
    -- Executing Set("SIP/192.168.0.99-100745e0", "DIALSTR=IAX2/guest at twinbrook.dnsdojo.net/15255427") in new stack
    -- Executing Goto("SIP/192.168.0.99-100745e0", "dodial") in new stack
    -- Goto (macro-dialcnet,s,19)
    -- Executing Dial("SIP/192.168.0.99-100745e0", "IAX2/guest at twinbrook.dnsdojo.net/15255427") in new stack
Jul  4 12:51:33 WARNING[9115]: chan_iax2.c:2747 create_addr: No such host: twinbrook.dnsdojo.net
Jul  4 12:51:33 NOTICE[9115]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Hangup("SIP/192.168.0.99-100745e0", "") in new stack
OpenWrt*CLI> 


Looks like your DNS resolution is wrong.

John

----- Original Message ----
From: Michael Zaiontz <mzaiontz at bellsouth.net>
To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
Sent: Wednesday, July 4, 2007 12:49:24 PM
Subject: [VoIP] Strange error message

Hi All!

I'm one or two steps closer to re-joining cnet.

When I attemp to call a local number on my switch, it ffails with this message:

called guest at twinbrook.dnsdojo.net/15255427
NOTICE[2674]: chan_iax2.c:934 Socket_process: Rejected connect attemp from 192.168.0.1, Request  '15255427 at default ' does not exist.

WARNING[2675]: chan_iax2.c:7181 Socket process: Call rejected by 68.222.56.129 No such context/extension

The 192.168.0.1 address is my router. The 68.222.56.129 is my wan address.
The asterisk box has a fixed IP address.

The extension does exist. I do have the correct port forwarded.  

I do have a guest account in my iax.conf file.

I can call test numbers on other switches in cnet.

What do I not see here?

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




From jjones3601 at yahoo.com  Wed Jul  4 13:22:08 2007
From: jjones3601 at yahoo.com (john jones)
Date: Wed, 4 Jul 2007 11:22:08 -0700 (PDT)
Subject: [VoIP] Strange error message
Message-ID: <328380.17095.qm@web34307.mail.mud.yahoo.com>

The reason I say you DNS is wrong is because

C:\Documents and Settings\Administrator>ping twinbrook.dnsdojo.net

Pinging twinbrook.dnsdojo.net [67.35.240.96] with 32 bytes of data:

Request timed out.
Request timed out.
Request timed out.
Request timed out.

Ping statistics for 67.35.240.96:
    Packets: Sent = 4, Received = 0, Lost = 4 (100% loss),

C:\Documents and Settings\Administrator>


doesn't match your fixed IP of 68.222.56.129


John

----- Original Message ----
From: john jones <jjones3601 at yahoo.com>
To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
Sent: Wednesday, July 4, 2007 1:54:08 PM
Subject: Re: [VoIP] Strange error message

Here is my debug output.


OpenWrt*CLI> 
    -- Executing Macro("SIP/192.168.0.99-100745e0", "dialcnet|15255427") in new stack
    -- Executing Set("SIP/192.168.0.99-100745e0", "E164NETWORKS=std.ckts.info") in new stack
    -- Executing Set("SIP/192.168.0.99-100745e0", "CALLERID(name)=John Jones") in new stack
    -- Executing Set("SIP/192.168.0.99-100745e0", "CALLERID(number)=687-7700") in new stack
    -- Executing GotoIf("SIP/192.168.0.99-100745e0", "0?startloop") in new stack
    -- Executing Set("SIP/192.168.0.99-100745e0", "ARG1=+15255427") in new stack
    -- Executing Set("SIP/192.168.0.99-100745e0", "ENUM=iax2:guest at twinbrook.dnsdojo.net/15255427") in new stack
    -- Executing GotoIf("SIP/192.168.0.99-100745e0", "0?continue") in new stack
    -- Executing GotoIf("SIP/192.168.0.99-100745e0", "0?sipuri") in new stack
    -- Executing GotoIf("SIP/192.168.0.99-100745e0", "1?iaxuri") in new stack
    -- Goto (macro-dialcnet,s,16)
    -- Executing Set("SIP/192.168.0.99-100745e0", "DIALSTR=IAX2/guest at twinbrook.dnsdojo.net/15255427") in new stack
    -- Executing Goto("SIP/192.168.0.99-100745e0", "dodial") in new stack
    -- Goto (macro-dialcnet,s,19)
    -- Executing Dial("SIP/192.168.0.99-100745e0", "IAX2/guest at twinbrook.dnsdojo.net/15255427") in new stack
Jul  4 12:51:33 WARNING[9115]: chan_iax2.c:2747 create_addr: No such host: twinbrook.dnsdojo.net
Jul  4 12:51:33 NOTICE[9115]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing Hangup("SIP/192.168.0.99-100745e0", "") in new stack
OpenWrt*CLI> 


Looks like your DNS resolution is wrong.

John

----- Original Message ----
From: Michael Zaiontz <mzaiontz at bellsouth.net>
To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
Sent: Wednesday, July 4, 2007 12:49:24 PM
Subject: [VoIP] Strange error message

Hi All!

I'm one or two steps closer to re-joining cnet.

When I attemp to call a local number on my switch, it ffails with this message:

called guest at twinbrook.dnsdojo.net/15255427
NOTICE[2674]: chan_iax2.c:934 Socket_process: Rejected connect attemp from 192.168.0.1, Request  '15255427 at default ' does not exist.

WARNING[2675]: chan_iax2.c:7181 Socket process: Call rejected by 68.222.56.129 No such context/extension

The 192.168.0.1 address is my router. The 68.222.56.129 is my wan address.
The asterisk box has a fixed IP address.

The extension does exist. I do have the correct port forwarded.  

I do have a guest account in my iax.conf file.

I can call test numbers on other switches in cnet.

What do I not see here?

Mike
_______________________________________________
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  Wed Jul  4 13:23:29 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Wed, 04 Jul 2007 14:23:29 -0400
Subject: [VoIP] Strange error message
In-Reply-To: <203706.16072.qm@web34310.mail.mud.yahoo.com>
References: <203706.16072.qm@web34310.mail.mud.yahoo.com>
Message-ID: <468BE5A1.8090609@stromberg-carlson.org>

Looks like Mike has made some progress.
I received no ringback or any audio, but the call was accepted and 
resolved to what I assume is his IP address.

John Novack
 

 -- Executing Macro("SIP/9900-09b37430", "dialcnet|15255427") in new stack
    -- Executing Set("SIP/9900-09b37430", "E164NETWORKS=std.ckts.info") 
in new stack
    -- Executing GotoIf("SIP/9900-09b37430", "0?startloop") in new stack
    -- Executing Set("SIP/9900-09b37430", "ARG1=+15255427") in new stack
    -- Executing Set("SIP/9900-09b37430", 
"ENUM=iax2:guest at twinbrook.dnsdojo.net/15255427") in new stack
    -- Executing GotoIf("SIP/9900-09b37430", "0?continue") in new stack
    -- Executing GotoIf("SIP/9900-09b37430", "0?sipuri") in new stack
    -- Executing GotoIf("SIP/9900-09b37430", "1?iaxuri") in new stack
    -- Goto (macro-dialcnet,s,14)
    -- Executing Set("SIP/9900-09b37430", 
"DIALSTR=IAX2/guest at twinbrook.dnsdojo.net/15255427") in new stack
    -- Executing Goto("SIP/9900-09b37430", "dodial") in new stack
    -- Goto (macro-dialcnet,s,17)
    -- Executing Dial("SIP/9900-09b37430", 
"IAX2/guest at twinbrook.dnsdojo.net/15255427") in new stack
    -- Called guest at twinbrook.dnsdojo.net/15255427
    -- Call accepted by 67.35.240.96 (format gsm)
    -- Format for call is gsm
    -- IAX2/67.35.240.96:4569-7 answered SIP/9900-09b37430
    -- Hungup 'IAX2/67.35.240.96:4569-7'


john jones wrote:
> Here is my debug output.
>
>
> OpenWrt*CLI> 
>     -- Executing Macro("SIP/192.168.0.99-100745e0", "dialcnet|15255427") in new stack
>     -- Executing Set("SIP/192.168.0.99-100745e0", "E164NETWORKS=std.ckts.info") in new stack
>     -- Executing Set("SIP/192.168.0.99-100745e0", "CALLERID(name)=John Jones") in new stack
>     -- Executing Set("SIP/192.168.0.99-100745e0", "CALLERID(number)=687-7700") in new stack
>     -- Executing GotoIf("SIP/192.168.0.99-100745e0", "0?startloop") in new stack
>     -- Executing Set("SIP/192.168.0.99-100745e0", "ARG1=+15255427") in new stack
>     -- Executing Set("SIP/192.168.0.99-100745e0", "ENUM=iax2:guest at twinbrook.dnsdojo.net/15255427") in new stack
>     -- Executing GotoIf("SIP/192.168.0.99-100745e0", "0?continue") in new stack
>     -- Executing GotoIf("SIP/192.168.0.99-100745e0", "0?sipuri") in new stack
>     -- Executing GotoIf("SIP/192.168.0.99-100745e0", "1?iaxuri") in new stack
>     -- Goto (macro-dialcnet,s,16)
>     -- Executing Set("SIP/192.168.0.99-100745e0", "DIALSTR=IAX2/guest at twinbrook.dnsdojo.net/15255427") in new stack
>     -- Executing Goto("SIP/192.168.0.99-100745e0", "dodial") in new stack
>     -- Goto (macro-dialcnet,s,19)
>     -- Executing Dial("SIP/192.168.0.99-100745e0", "IAX2/guest at twinbrook.dnsdojo.net/15255427") in new stack
> Jul  4 12:51:33 WARNING[9115]: chan_iax2.c:2747 create_addr: No such host: twinbrook.dnsdojo.net
> Jul  4 12:51:33 NOTICE[9115]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination)
>   == Everyone is busy/congested at this time (1:0/0/1)
>     -- Executing Hangup("SIP/192.168.0.99-100745e0", "") in new stack
> OpenWrt*CLI> 
>
>
> Looks like your DNS resolution is wrong.
>
> John
>
> ----- Original Message ----
> From: Michael Zaiontz <mzaiontz at bellsouth.net>
> To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
> Sent: Wednesday, July 4, 2007 12:49:24 PM
> Subject: [VoIP] Strange error message
>
> Hi All!
>
> I'm one or two steps closer to re-joining cnet.
>
> When I attemp to call a local number on my switch, it ffails with this message:
>
> called guest at twinbrook.dnsdojo.net/15255427
> NOTICE[2674]: chan_iax2.c:934 Socket_process: Rejected connect attemp from 192.168.0.1, Request  '15255427 at default ' does not exist.
>
> WARNING[2675]: chan_iax2.c:7181 Socket process: Call rejected by 68.222.56.129 No such context/extension
>
> The 192.168.0.1 address is my router. The 68.222.56.129 is my wan address.
> The asterisk box has a fixed IP address.
>
> The extension does exist. I do have the correct port forwarded.  
>
> I do have a guest account in my iax.conf file.
>
> I can call test numbers on other switches in cnet.
>
> What do I not see here?
>
> Mike
> _______________________________________________
> 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/
>
>   

-- 
Dog is my co-pilot


From mzaiontz at bellsouth.net  Wed Jul  4 13:33:38 2007
From: mzaiontz at bellsouth.net (Michael Zaiontz)
Date: Wed, 4 Jul 2007 13:33:38 -0500
Subject: [VoIP] Strange error message
References: <203706.16072.qm@web34310.mail.mud.yahoo.com>
	<468BE5A1.8090609@stromberg-carlson.org>
Message-ID: <000e01c7be69$d6cf93c0$6400a8c0@zaiontz1>

TaDa!!!!!!!!!

For the moment you can call me at 525-5990 (I hope)

I still have some problems with messages it can't find, but give it a try. 
I'll be here,and on cnet, till about 4:00 CDT.

Thanks for all your help
Mike


----- Original Message ----- 
From: "John Novack" <jnovack at stromberg-carlson.org>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Wednesday, July 04, 2007 1:23 PM
Subject: Re: [VoIP] Strange error message


> Looks like Mike has made some progress.
> I received no ringback or any audio, but the call was accepted and
> resolved to what I assume is his IP address.
>
> John Novack
>
>
> -- Executing Macro("SIP/9900-09b37430", "dialcnet|15255427") in new stack
>    -- Executing Set("SIP/9900-09b37430", "E164NETWORKS=std.ckts.info")
> in new stack
>    -- Executing GotoIf("SIP/9900-09b37430", "0?startloop") in new stack
>    -- Executing Set("SIP/9900-09b37430", "ARG1=+15255427") in new stack
>    -- Executing Set("SIP/9900-09b37430",
> "ENUM=iax2:guest at twinbrook.dnsdojo.net/15255427") in new stack
>    -- Executing GotoIf("SIP/9900-09b37430", "0?continue") in new stack
>    -- Executing GotoIf("SIP/9900-09b37430", "0?sipuri") in new stack
>    -- Executing GotoIf("SIP/9900-09b37430", "1?iaxuri") in new stack
>    -- Goto (macro-dialcnet,s,14)
>    -- Executing Set("SIP/9900-09b37430",
> "DIALSTR=IAX2/guest at twinbrook.dnsdojo.net/15255427") in new stack
>    -- Executing Goto("SIP/9900-09b37430", "dodial") in new stack
>    -- Goto (macro-dialcnet,s,17)
>    -- Executing Dial("SIP/9900-09b37430",
> "IAX2/guest at twinbrook.dnsdojo.net/15255427") in new stack
>    -- Called guest at twinbrook.dnsdojo.net/15255427
>    -- Call accepted by 67.35.240.96 (format gsm)
>    -- Format for call is gsm
>    -- IAX2/67.35.240.96:4569-7 answered SIP/9900-09b37430
>    -- Hungup 'IAX2/67.35.240.96:4569-7'
>
>
> john jones wrote:
>> Here is my debug output.
>>
>>
>> OpenWrt*CLI>
>>     -- Executing Macro("SIP/192.168.0.99-100745e0", "dialcnet|15255427") 
>> in new stack
>>     -- Executing Set("SIP/192.168.0.99-100745e0", 
>> "E164NETWORKS=std.ckts.info") in new stack
>>     -- Executing Set("SIP/192.168.0.99-100745e0", "CALLERID(name)=John 
>> Jones") in new stack
>>     -- Executing Set("SIP/192.168.0.99-100745e0", 
>> "CALLERID(number)=687-7700") in new stack
>>     -- Executing GotoIf("SIP/192.168.0.99-100745e0", "0?startloop") in 
>> new stack
>>     -- Executing Set("SIP/192.168.0.99-100745e0", "ARG1=+15255427") in 
>> new stack
>>     -- Executing Set("SIP/192.168.0.99-100745e0", 
>> "ENUM=iax2:guest at twinbrook.dnsdojo.net/15255427") in new stack
>>     -- Executing GotoIf("SIP/192.168.0.99-100745e0", "0?continue") in new 
>> stack
>>     -- Executing GotoIf("SIP/192.168.0.99-100745e0", "0?sipuri") in new 
>> stack
>>     -- Executing GotoIf("SIP/192.168.0.99-100745e0", "1?iaxuri") in new 
>> stack
>>     -- Goto (macro-dialcnet,s,16)
>>     -- Executing Set("SIP/192.168.0.99-100745e0", 
>> "DIALSTR=IAX2/guest at twinbrook.dnsdojo.net/15255427") in new stack
>>     -- Executing Goto("SIP/192.168.0.99-100745e0", "dodial") in new stack
>>     -- Goto (macro-dialcnet,s,19)
>>     -- Executing Dial("SIP/192.168.0.99-100745e0", 
>> "IAX2/guest at twinbrook.dnsdojo.net/15255427") in new stack
>> Jul  4 12:51:33 WARNING[9115]: chan_iax2.c:2747 create_addr: No such 
>> host: twinbrook.dnsdojo.net
>> Jul  4 12:51:33 NOTICE[9115]: app_dial.c:1010 dial_exec_full: Unable to 
>> create channel of type 'IAX2' (cause 3 - No route to destination)
>>   == Everyone is busy/congested at this time (1:0/0/1)
>>     -- Executing Hangup("SIP/192.168.0.99-100745e0", "") in new stack
>> OpenWrt*CLI>
>>
>>
>> Looks like your DNS resolution is wrong.
>>
>> John
>>
>> ----- Original Message ----
>> From: Michael Zaiontz <mzaiontz at bellsouth.net>
>> To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
>> Sent: Wednesday, July 4, 2007 12:49:24 PM
>> Subject: [VoIP] Strange error message
>>
>> Hi All!
>>
>> I'm one or two steps closer to re-joining cnet.
>>
>> When I attemp to call a local number on my switch, it ffails with this 
>> message:
>>
>> called guest at twinbrook.dnsdojo.net/15255427
>> NOTICE[2674]: chan_iax2.c:934 Socket_process: Rejected connect attemp 
>> from 192.168.0.1, Request  '15255427 at default ' does not exist.
>>
>> WARNING[2675]: chan_iax2.c:7181 Socket process: Call rejected by 
>> 68.222.56.129 No such context/extension
>>
>> The 192.168.0.1 address is my router. The 68.222.56.129 is my wan 
>> address.
>> The asterisk box has a fixed IP address.
>>
>> The extension does exist. I do have the correct port forwarded.
>>
>> I do have a guest account in my iax.conf file.
>>
>> I can call test numbers on other switches in cnet.
>>
>> What do I not see here?
>>
>> Mike
>> _______________________________________________
>> 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/
>>
>>
>
> -- 
> Dog is my co-pilot
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/ 


From mzaiontz at bellsouth.net  Wed Jul  4 15:52:35 2007
From: mzaiontz at bellsouth.net (Michael Zaiontz)
Date: Wed, 4 Jul 2007 15:52:35 -0500
Subject: [VoIP] My switch works!
Message-ID: <000801c7be7d$403a23d0$6400a8c0@zaiontz1>

Thanks for the help, everyone!!!!!

I still have a few loose ends to tie up, but it appears to be working OK. I'll be down until I get a UPS/surge protector for that PC. I'll do that either tomorrow or Friday then I'll be on the net 7/24.
I'll post again when that happens.

Thanks again,

Mike Zaiontz


From greg at vyger.net  Wed Jul  4 18:59:27 2007
From: greg at vyger.net (Greg Blakely)
Date: Wed, 4 Jul 2007 18:59:27 -0500
Subject: [VoIP] Officde Code Activations
Message-ID: <A691C3685A152E4AAC9608B8D8E113166863A2@godzilla.hickory.vyger.net>

Hey, folks.  I finally figured out what was going on with the mysterious
disapearances of the forms that people have filled out asking for office
code activations or account setups.
 
The mail was going to an email account that I haven't used in a very
long time.
 
Today, I stumbled across 1,590 messages that have come in since April.
Most of them were from one scam or another, but there are a fair amount
of them that were CNETters.
 
So, I'm wading through them, and will be sending email to individuals
who didn't end up getting in touch with me despite the misdirected form
mail.
 
 

From jjones3601 at yahoo.com  Wed Jul  4 20:16:08 2007
From: jjones3601 at yahoo.com (john jones)
Date: Wed, 4 Jul 2007 18:16:08 -0700 (PDT)
Subject: [VoIP] ENUM Question for Greg
Message-ID: <309041.55030.qm@web34313.mail.mud.yahoo.com>

Greg,

Is the ENUM server a function of your Asterisk system or is it a separate system?

Thanks!

John




From greg at vyger.net  Wed Jul  4 20:17:31 2007
From: greg at vyger.net (Greg Blakely)
Date: Wed, 4 Jul 2007 20:17:31 -0500
Subject: [VoIP] ENUM Question for Greg
Message-ID: <A691C3685A152E4AAC9608B8D8E113166863A5@godzilla.hickory.vyger.net>

It is a standard DNS server running on a separate Linux PC. 

> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] 
> On Behalf Of john jones
> Sent: Wednesday, July 04, 2007 8:16 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: [VoIP] ENUM Question for Greg
> 
> Greg,
> 
> Is the ENUM server a function of your Asterisk system or is 
> it a separate system?
> 
> Thanks!
> 
> John
> 
> 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 

From mzaiontz at bellsouth.net  Fri Jul  6 09:44:39 2007
From: mzaiontz at bellsouth.net (Michael Zaiontz)
Date: Fri, 6 Jul 2007 09:44:39 -0500
Subject: [VoIP] Up and running!
Message-ID: <003e01c7bfdc$2e3a9510$6400a8c0@zaiontz1>

Hi All!

Just a quick word to say I'm on the net. 5255990 will get you me, if I'm in the office, or my answering machine.  I'm on my way out to check on my new house across the lake, well above sea level, so have fun.

Thanks,
Mike Zaiontz

From ratguy at bellsouth.net  Sat Jul  7 05:12:18 2007
From: ratguy at bellsouth.net (Jayson smith)
Date: Sat, 7 Jul 2007 06:12:18 -0400
Subject: [VoIP] Strange happening with DSL service
Message-ID: <000801c7c07f$5f1a7ee0$6600a8c0@bluegrasspals.com>

Hi all,
     This weekend, we've been listening to the annual men's Barbershop Harmony Society convention webcast. So it's going along nicely, and poof! the stream dies. No big deal, I'll just restart it. No dice. Reboot the router and computer, no dice. Finally I realize we've got no DSL, and for a very good reason. There's static on the line. That's strange. We sometimes get static after rain, but it hasn't rained here recently. Anyway, a couple of calls to the ringback circuit strangely takes care of the static problem, but still no DSL. So now we're stuck with dial-up, which is how I'm sending this message. I've tried all sorts of stuff with the modem and router, all to no avail. So finally this morning I get the bright idea to take a phone and plug it into the DSL jack. I plug the phone in, pick it up and... Nothing. No dialtone, no battery, just as if it wasn't plugged in. Further tests reveal, it's not the phone cord, it's not the phone. After all this frustration, I find out the DSL jack has decided to go to that place in the sky where all good phone jacks which have lived full lives go. My question is, what would cause the jack to do this, just suddenly? Can something actually happen to the jack or its wires, just like that, or does something have to have gotten disconnected somehow? As far as I know, all the other jacks on that line work. Looks like we're stuck with dial-up until sometime next week. Man, being blind is sometimes a pain! If I could see, and knew halfway what I was doing, I'd go out to the box and see what was what. BTW, our line is the type where there's a DSL jack, then a DSL filter, behind which are all the other phone jacks. If the DSL filter got fried, that would do just the opposite, DSL service but no other phone service, right?
Thanks for any help on this.
Jayson

From dfroula at sbcglobal.net  Sat Jul  7 07:11:52 2007
From: dfroula at sbcglobal.net (Donald Froula)
Date: Sat, 7 Jul 2007 05:11:52 -0700 (PDT)
Subject: [VoIP] Strange happening with DSL service
In-Reply-To: <000801c7c07f$5f1a7ee0$6600a8c0@bluegrasspals.com>
Message-ID: <554136.27054.qm@web83201.mail.mud.yahoo.com>

Jayson,

It could be your filter. The DSL installation kit I
received from SBC contained a bag of filters that were
designed to be installed at every phone jack in the
house. I originally tried using one of these as a
whole-house filter. It fried after a few weeks with
the same symptoms as yours. I finally purchased a
whole-house filter on Ebay. The one I purchased is
from Siecor. It has three sets of terminals: Tip/Ring
in, filtered Tip/Ring out for the phone lines in the
rest of the house, and unfiltered Tip/Ring out for the
DSL modem. It also has a filtered and unfiltered phone
jack on the fromt for testing. This has been
trouble-free for many years.

Is the dead jack you plugged the phone into before or
after the filter? 

If before, no failure of the filter should cause the
problem. It's probably a bad connection at the jack.

If the jack is after the filter, it could well be the
filter.

Don


--- Jayson smith <ratguy at bellsouth.net> wrote:

> Hi all,
>      This weekend, we've been listening to the
> annual men's Barbershop Harmony Society convention
> webcast. So it's going along nicely, and poof! the
> stream dies. No big deal, I'll just restart it. No
> dice. Reboot the router and computer, no dice.
> Finally I realize we've got no DSL, and for a very
> good reason. There's static on the line. That's
> strange. We sometimes get static after rain, but it
> hasn't rained here recently. Anyway, a couple of
> calls to the ringback circuit strangely takes care
> of the static problem, but still no DSL. So now
> we're stuck with dial-up, which is how I'm sending
> this message. I've tried all sorts of stuff with the
> modem and router, all to no avail. So finally this
> morning I get the bright idea to take a phone and
> plug it into the DSL jack. I plug the phone in, pick
> it up and... Nothing. No dialtone, no battery, just
> as if it wasn't plugged in. Further tests reveal,
> it's not the phone cord, it's not the phone. After
> all this frustration, I find out 
>  the DSL jack has decided to go to that place in the
> sky where all good phone jacks which have lived full
> lives go. My question is, what would cause the jack
> to do this, just suddenly? Can something actually
> happen to the jack or its wires, just like that, or
> does something have to have gotten disconnected
> somehow? As far as I know, all the other jacks on
> that line work. Looks like we're stuck with dial-up
> until sometime next week. Man, being blind is
> sometimes a pain! If I could see, and knew halfway
> what I was doing, I'd go out to the box and see what
> was what. BTW, our line is the type where there's a
> DSL jack, then a DSL filter, behind which are all
> the other phone jacks. If the DSL filter got fried,
> that would do just the opposite, DSL service but no
> other phone service, right?
> Thanks for any help on this.
> Jayson
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From hockd at dteenergy.com  Sat Jul  7 09:16:03 2007
From: hockd at dteenergy.com (Dennis D Hock)
Date: Sat, 7 Jul 2007 10:16:03 -0400
Subject: [VoIP] Strange happening with DSL service
Message-ID: <OF7F306905.BA076C5D-ON85257311.004E5FEA-85257311.004E5FEF@dteenergy.com>


I agree with Don saounds like the filter is the prime suspect at this
point.  The Siecor and other whole house ones are designed a little better
I believe, although there was a discussion on the TCI list I think some
time ago about filters.

Good luck,

Dennis Hock

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


To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
From: Donald Froula <dfroula at sbcglobal.net>
Sent by: voip-bounces at ckts.info
Date: 07/07/2007 08:11AM
Subject: Re: [VoIP] Strange happening with DSL service

Jayson,

It could be your filter. The DSL installation kit I
received from SBC contained a bag of filters that were
designed to be installed at every phone jack in the
house. I originally tried using one of these as a
whole-house filter. It fried after a few weeks with
the same symptoms as yours. I finally purchased a
whole-house filter on Ebay. The one I purchased is
from Siecor. It has three sets of terminals: Tip/Ring
in, filtered Tip/Ring out for the phone lines in the
rest of the house, and unfiltered Tip/Ring out for the
DSL modem. It also has a filtered and unfiltered phone
jack on the fromt for testing. This has been
trouble-free for many years.

Is the dead jack you plugged the phone into before or
after the filter?

If before, no failure of the filter should cause the
problem. It's probably a bad connection at the jack.

If the jack is after the filter, it could well be the
filter.

Don


--- Jayson smith <ratguy at bellsouth.net> wrote:

> Hi all,
>      This weekend, we've been listening to the
> annual men's Barbershop Harmony Society convention
> webcast. So it's going along nicely, and poof! the
> stream dies. No big deal, I'll just restart it. No
> dice. Reboot the router and computer, no dice.
> Finally I realize we've got no DSL, and for a very
> good reason. There's static on the line. That's
> strange. We sometimes get static after rain, but it
> hasn't rained here recently. Anyway, a couple of
> calls to the ringback circuit strangely takes care
> of the static problem, but still no DSL. So now
> we're stuck with dial-up, which is how I'm sending
> this message. I've tried all sorts of stuff with the
> modem and router, all to no avail. So finally this
> morning I get the bright idea to take a phone and
> plug it into the DSL jack. I plug the phone in, pick
> it up and... Nothing. No dialtone, no battery, just
> as if it wasn't plugged in. Further tests reveal,
> it's not the phone cord, it's not the phone. After
> all this frustration, I find out
>  the DSL jack has decided to go to that place in the
> sky where all good phone jacks which have lived full
> lives go. My question is, what would cause the jack
> to do this, just suddenly? Can something actually
> happen to the jack or its wires, just like that, or
> does something have to have gotten disconnected
> somehow? As far as I know, all the other jacks on
> that line work. Looks like we're stuck with dial-up
> until sometime next week. Man, being blind is
> sometimes a pain! If I could see, and knew halfway
> what I was doing, I'd go out to the box and see what
> was what. BTW, our line is the type where there's a
> DSL jack, then a DSL filter, behind which are all
> the other phone jacks. If the DSL filter got fried,
> that would do just the opposite, DSL service but no
> other phone service, right?
> Thanks for any help on this.
> Jayson
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>

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

From n00dle at yahoo.com  Sat Jul  7 20:51:04 2007
From: n00dle at yahoo.com (Chris Craft)
Date: Sat, 7 Jul 2007 18:51:04 -0700 (PDT)
Subject: [VoIP] New to list/subject.
Message-ID: <163368.75041.qm@web50608.mail.re2.yahoo.com>

 I'm new to this list but have a few small PBXs that
are almost old enough to vote, wishing I had an older
switch to play with.  I work in the internet service
provider industry, but have some experience with telco
stuff, mainly from the PBX point ov view.  The first
PBX I was administrator for was an AT&T System25 back
in '94.
  Nowadays I'm building an Asterisk PBX to replace the
Lucent Partner system at work.  My current home pbx is
a Vodavi StarPlus SPD 1428, which I have
frankensteined together with my home Asterisk box via
a Sipura 2000 ATA.  I've been using linux since 1992
and Asterisk since 2003, so I can help people from
more switching experience with Asterisk, and maybe
some switchers can help me out with the history bits.

Cheers!
Chris Craft, who'll be reserving an office code soon.


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

From kb0tdf at yahoo.com  Sat Jul  7 21:22:21 2007
From: kb0tdf at yahoo.com (Greg Blakely)
Date: Sat, 7 Jul 2007 21:22:21 -0500
Subject: [VoIP] New to list/subject.
In-Reply-To: <163368.75041.qm@web50608.mail.re2.yahoo.com>
References: <163368.75041.qm@web50608.mail.re2.yahoo.com>
Message-ID: <011b01c7c106$d1e94a00$750a0a0a@hickory.vyger.net>

Welcome, Chris.

Some of us have loads of Linux and Asterisk experience; others have some
pretty awe inspiring electromechanical switches working; and yet others know
the whole thing -- not me, though.

This list works to answer questions related to all aspects of
electromechanical switches using internet connections, so the depth and
range of the discussion is fairly complete.

We certainly can use another asterisk guru.  (I am always on the lookout for
cool AGI and 'manager' hooks into asterisk.)  And we have a few who have
actually set asterisk to work on a router -- no need for the PC.

I mostly lurk on this list, and am very often amazed at some of the things
that I learn from it.

Again, welcome.  

 

> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] 
> On Behalf Of Chris Craft
> Sent: Saturday, July 07, 2007 8:51 PM
> To: voip at ckts.info
> Subject: [VoIP] New to list/subject.
> 
>  I'm new to this list but have a few small PBXs that are 
> almost old enough to vote, wishing I had an older switch to 
> play with.  I work in the internet service provider industry, 
> but have some experience with telco stuff, mainly from the 
> PBX point ov view.  The first PBX I was administrator for was 
> an AT&T System25 back in '94.
>   Nowadays I'm building an Asterisk PBX to replace the Lucent 
> Partner system at work.  My current home pbx is a Vodavi 
> StarPlus SPD 1428, which I have frankensteined together with 
> my home Asterisk box via a Sipura 2000 ATA.  I've been using 
> linux since 1992 and Asterisk since 2003, so I can help 
> people from more switching experience with Asterisk, and 
> maybe some switchers can help me out with the history bits.
> 
> Cheers!
> Chris Craft, who'll be reserving an office code soon.
> 
> 


From donw at engineeringinc.com  Sat Jul  7 21:34:38 2007
From: donw at engineeringinc.com (Don Wisdom)
Date: Sat, 07 Jul 2007 20:34:38 -0600
Subject: [VoIP] Asterisk Questions
Message-ID: <C2B5A95E.23483%donw@engineeringinc.com>

Hi All,
Has anyone here setup cnet on a machine running freePBX ?
If you have please drop me a line off list. My email address is: donw _ at _
engineeringinc _._ com (remove the _'s

Thanks
--Don


From ikj1234i at yahoo.com  Sun Jul  8 08:00:28 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Sun, 8 Jul 2007 06:00:28 -0700 (PDT)
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <468FD867.1070306@comcast.net>
Message-ID: <20070708130028.48061.qmail@web51601.mail.re2.yahoo.com>


Replying on-list in the event that anyone else has
commentary to add.............

I believe there may be two separate issues here. 
First, the parameters you mention should be tweakable
via zaptel.h, specifically
zaptel.h:#define        ZT_DEFAULT_PULSEMAKETIME 50   
 /* 50 ms of line closed when dial pulsing */
zaptel.h:#define        ZT_DEFAULT_PULSEBREAKTIME 50  
 /* 50 ms of line open when dial pulsing */
zaptel.h:#define        ZT_DEFAULT_PULSEAFTERTIME 750 
 /* 750ms between dial pulse digits */

[Note, it is theoretically possible for user-space
(ztcfg) to override these, although I don't believe
this is currently implemented]

[Note 2, I believe that these values are not correct
for true Bell System dials, the ratio should be ~60/40
not 50/50]

The second issue is that several users have reported
erratic pulsing, which I have noticed too, just by
listening the problem is clearly audible.  Instead of
hearing regularly spaced pulses you hear plainly that
the spacing is irregular.

Try connecting a butt set (in monitor mode) to the
connection and listen to the outpulsing.   Best way to
test this is to use a string of several 'zero'
digits...

I recently was able to grab an FXO card for my channel
bank.  When it shows up (and when I get it set up)
I'll try to duplicate the problem.  I have released an
asterisk dial speed app that will also help pinpoint
it.  See URL

http://www.lightlink.com/mhp/dialtest/

Max

--- Kirt Stanfield <kirtley.stanfield at comcast.net>
wrote:

> Max,
> 
> I hope all is well with you.
> 
> Early this year I installed a Varian T1 card in my
> asterisk machine, and 
> connected it to an adtran channel bank. The channel
> bank has 5 FXS cards 
> and 1 FXO card. The former is connected to ground
> start 2 way trunks on 
> my step switch. The latter is connected to the
> incoming selectors on my 
> step switch.
> 
> In general the above configuration has performed
> well, with a tremendous 
> improvement in voice quality.
> 
> I have however seen evidence of misdialing into the
> switch via the FXO 
> card. I finally wrote a php script to dial over and
> over again, and from 
> that I have deduced that there is about a 5-10%
> misdial rate. I have now 
> place a Metro-Tel TPM32  'digit grabber' in the
> circuit. That indicates 
> that dialing is taking place at a rate of 13pps,
> instead of 10pps, which 
> I suspect is the reason why I am seeing the error
> rate that I am.
> 
> So my question is - can you tell me any place(s) in
> the asterisk or the 
> drivers that I could adjust so as to slow down the
> dial rate?
> 
> Thanks,
> Kirt
> 
> 



 
____________________________________________________________________________________
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 stfkerman at jps.net  Sun Jul  8 10:26:27 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Sun, 08 Jul 2007 11:26:27 -0400
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <20070708130028.48061.qmail@web51601.mail.re2.yahoo.com>
References: <20070708130028.48061.qmail@web51601.mail.re2.yahoo.com>
Message-ID: <46910223.8030009@jps.net>

Wasn't there a dial pulse analysis program floating around out there on 
TCI that reported the duration of each make and break pulse in a digit 
as a list of values?  Seems like that would prove whether the pulsing is 
erratic and identify just how bad it is.  Was it one Dennis Hallworth or 
Steve Flocke wrote?

Steph

ikjtel wrote:
> Replying on-list in the event that anyone else has
> commentary to add.............
>
> I believe there may be two separate issues here. 
> First, the parameters you mention should be tweakable
> via zaptel.h, specifically
> zaptel.h:#define        ZT_DEFAULT_PULSEMAKETIME 50   
>  /* 50 ms of line closed when dial pulsing */
> zaptel.h:#define        ZT_DEFAULT_PULSEBREAKTIME 50  
>  /* 50 ms of line open when dial pulsing */
> zaptel.h:#define        ZT_DEFAULT_PULSEAFTERTIME 750 
>  /* 750ms between dial pulse digits */
>
> [Note, it is theoretically possible for user-space
> (ztcfg) to override these, although I don't believe
> this is currently implemented]
>
> [Note 2, I believe that these values are not correct
> for true Bell System dials, the ratio should be ~60/40
> not 50/50]
>
> The second issue is that several users have reported
> erratic pulsing, which I have noticed too, just by
> listening the problem is clearly audible.  Instead of
> hearing regularly spaced pulses you hear plainly that
> the spacing is irregular.
>
> Try connecting a butt set (in monitor mode) to the
> connection and listen to the outpulsing.   Best way to
> test this is to use a string of several 'zero'
> digits...
>
> I recently was able to grab an FXO card for my channel
> bank.  When it shows up (and when I get it set up)
> I'll try to duplicate the problem.  I have released an
> asterisk dial speed app that will also help pinpoint
> it.  See URL
>
> http://www.lightlink.com/mhp/dialtest/
>
> Max
>
> --- Kirt Stanfield <kirtley.stanfield at comcast.net>
> wrote:
>
>   
>> Max,
>>
>> I hope all is well with you.
>>
>> Early this year I installed a Varian T1 card in my
>> asterisk machine, and 
>> connected it to an adtran channel bank. The channel
>> bank has 5 FXS cards 
>> and 1 FXO card. The former is connected to ground
>> start 2 way trunks on 
>> my step switch. The latter is connected to the
>> incoming selectors on my 
>> step switch.
>>
>> In general the above configuration has performed
>> well, with a tremendous 
>> improvement in voice quality.
>>
>> I have however seen evidence of misdialing into the
>> switch via the FXO 
>> card. I finally wrote a php script to dial over and
>> over again, and from 
>> that I have deduced that there is about a 5-10%
>> misdial rate. I have now 
>> place a Metro-Tel TPM32  'digit grabber' in the
>> circuit. That indicates 
>> that dialing is taking place at a rate of 13pps,
>> instead of 10pps, which 
>> I suspect is the reason why I am seeing the error
>> rate that I am.
>>
>> So my question is - can you tell me any place(s) in
>> the asterisk or the 
>> drivers that I could adjust so as to slow down the
>> dial rate?
>>
>> Thanks,
>> Kirt
>>
>>
>>     
>
>
>
>  
> ____________________________________________________________________________________
> 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 jnovack at stromberg-carlson.org  Sun Jul  8 11:11:09 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Sun, 08 Jul 2007 12:11:09 -0400
Subject: [VoIP] New to list/subject.
In-Reply-To: <163368.75041.qm@web50608.mail.re2.yahoo.com>
References: <163368.75041.qm@web50608.mail.re2.yahoo.com>
Message-ID: <46910C9D.5030405@stromberg-carlson.org>

Welcome to the list and to CNET.
Where in the world are you Chris?

SXS equipment is still out there, if you know where to look and are 
willing to travel to gather it up. Also the space to put something 
together.
Personally, as an interconnect installer with 25 years experience, I 
would not use Asterisk in a small business, given the available products 
on the market today.
Others will disagree, naturally.
I know the Vodavi, and in fact this coming week have to reprogram its 
big brother ( 2856 ) for a new to me customer who bought it with their 
building.

Welcome to CNET

John Novack


Chris Craft wrote:
>  I'm new to this list but have a few small PBXs that
> are almost old enough to vote, wishing I had an older
> switch to play with.  I work in the internet service
> provider industry, but have some experience with telco
> stuff, mainly from the PBX point ov view.  The first
> PBX I was administrator for was an AT&T System25 back
> in '94.
>   Nowadays I'm building an Asterisk PBX to replace the
> Lucent Partner system at work.  My current home pbx is
> a Vodavi StarPlus SPD 1428, which I have
> frankensteined together with my home Asterisk box via
> a Sipura 2000 ATA.  I've been using linux since 1992
> and Asterisk since 2003, so I can help people from
> more switching experience with Asterisk, and maybe
> some switchers can help me out with the history bits.
>
> Cheers!
> Chris Craft, who'll be reserving an office code soon.
>
>
>        
> ____________________________________________________________________________________
> Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.
> http://smallbusiness.yahoo.com/webhosting 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From kirtley.stanfield at comcast.net  Sun Jul  8 13:16:12 2007
From: kirtley.stanfield at comcast.net (Kirt Stanfield)
Date: Sun, 08 Jul 2007 14:16:12 -0400
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <20070708130028.48061.qmail@web51601.mail.re2.yahoo.com>
References: <20070708130028.48061.qmail@web51601.mail.re2.yahoo.com>
Message-ID: <469129EC.8060404@comcast.net>

Max,

I changed 'ZT_DEFAULT_PULSEAFTERTIME' to 900, rebuilt everything, and 
rebooted. I see no change in the PPS. Also m/b has always dispalyed as 60/40

So I suspect these are in fact being overidden somewhere.

Kirt
ikjtel wrote:

>Replying on-list in the event that anyone else has
>commentary to add.............
>
>I believe there may be two separate issues here. 
>First, the parameters you mention should be tweakable
>via zaptel.h, specifically
>zaptel.h:#define        ZT_DEFAULT_PULSEMAKETIME 50   
> /* 50 ms of line closed when dial pulsing */
>zaptel.h:#define        ZT_DEFAULT_PULSEBREAKTIME 50  
> /* 50 ms of line open when dial pulsing */
>zaptel.h:#define        ZT_DEFAULT_PULSEAFTERTIME 750 
> /* 750ms between dial pulse digits */
>
>[Note, it is theoretically possible for user-space
>(ztcfg) to override these, although I don't believe
>this is currently implemented]
>
>[Note 2, I believe that these values are not correct
>for true Bell System dials, the ratio should be ~60/40
>not 50/50]
>
>The second issue is that several users have reported
>erratic pulsing, which I have noticed too, just by
>listening the problem is clearly audible.  Instead of
>hearing regularly spaced pulses you hear plainly that
>the spacing is irregular.
>
>Try connecting a butt set (in monitor mode) to the
>connection and listen to the outpulsing.   Best way to
>test this is to use a string of several 'zero'
>digits...
>
>I recently was able to grab an FXO card for my channel
>bank.  When it shows up (and when I get it set up)
>I'll try to duplicate the problem.  I have released an
>asterisk dial speed app that will also help pinpoint
>it.  See URL
>
>http://www.lightlink.com/mhp/dialtest/
>
>Max
>
>--- Kirt Stanfield <kirtley.stanfield at comcast.net>
>wrote:
>
>  
>
>>Max,
>>
>>I hope all is well with you.
>>
>>Early this year I installed a Varian T1 card in my
>>asterisk machine, and 
>>connected it to an adtran channel bank. The channel
>>bank has 5 FXS cards 
>>and 1 FXO card. The former is connected to ground
>>start 2 way trunks on 
>>my step switch. The latter is connected to the
>>incoming selectors on my 
>>step switch.
>>
>>In general the above configuration has performed
>>well, with a tremendous 
>>improvement in voice quality.
>>
>>I have however seen evidence of misdialing into the
>>switch via the FXO 
>>card. I finally wrote a php script to dial over and
>>over again, and from 
>>that I have deduced that there is about a 5-10%
>>misdial rate. I have now 
>>place a Metro-Tel TPM32  'digit grabber' in the
>>circuit. That indicates 
>>that dialing is taking place at a rate of 13pps,
>>instead of 10pps, which 
>>I suspect is the reason why I am seeing the error
>>rate that I am.
>>
>>So my question is - can you tell me any place(s) in
>>the asterisk or the 
>>drivers that I could adjust so as to slow down the
>>dial rate?
>>
>>Thanks,
>>Kirt
>>
>>
>>    
>>
>
>
>
> 
>____________________________________________________________________________________
>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 rdekema at gmail.com  Sun Jul  8 19:25:54 2007
From: rdekema at gmail.com (Rusty Dekema)
Date: Sun, 8 Jul 2007 20:25:54 -0400
Subject: [VoIP] Bogus CallerID (oops)
Message-ID: <68171c120707081725o4804de72i194d109d67639cfd@mail.gmail.com>

Greetings,

A couple hours ago, I discovered that I had been sending bogus caller
ID values with my calls onto CNET. Most of the calls I've been making
recently are to various Evan Doorbell recordings, and the values I was
sending were mostly 1000 through 1002. I thought I had taken care of
rewriting the Caller ID for calls onto CNET, but it turned out that
portion of my dialplan logic was not actually being called before
placing CNET calls.

Anyway, I've taken care of this now, and a number in office code 1-616
is now being sent where I can be reached. So if you've been wondering
who the heck's been calling your recordings 10 times a day
(exaggeration, mostly ;)), now you know.

Rusty

From hockd at dteenergy.com  Mon Jul  9 04:33:09 2007
From: hockd at dteenergy.com (Dennis D Hock)
Date: Mon, 9 Jul 2007 05:33:09 -0400
Subject: [VoIP] Bogus CallerID (oops)
Message-ID: <OFD37675FE.84D30C03-ON85257313.0034797D-85257313.00347982@dteenergy.com>

Rusty,

I noticed your choice of office code 616.  Are you in Michigan?  I am on
the eastern side between Mt Clemens and Pt Huron.

Dennis Hock 269

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


To: voip at ckts.info
From: "Rusty Dekema" <rdekema at gmail.com>
Sent by: voip-bounces at ckts.info
Date: 07/08/2007 08:25PM
Subject: [VoIP] Bogus CallerID (oops)

Greetings,

A couple hours ago, I discovered that I had been sending bogus caller
ID values with my calls onto CNET. Most of the calls I've been making
recently are to various Evan Doorbell recordings, and the values I was
sending were mostly 1000 through 1002. I thought I had taken care of
rewriting the Caller ID for calls onto CNET, but it turned out that
portion of my dialplan logic was not actually being called before
placing CNET calls.

Anyway, I've taken care of this now, and a number in office code 1-616
is now being sent where I can be reached. So if you've been wondering
who the heck's been calling your recordings 10 times a day
(exaggeration, mostly ;)), now you know.

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

From hockd at dteenergy.com  Mon Jul  9 04:40:51 2007
From: hockd at dteenergy.com (Dennis D Hock)
Date: Mon, 9 Jul 2007 05:40:51 -0400
Subject: [VoIP] New to list/subject.
Message-ID: <OFE31FF7D1.BA761818-ON85257313.00352DFF-85257313.00352E04@dteenergy.com>


Chris Welcome aboard.  This is a very diverse group with experience ranging
all over the place.  You don't have to have any electromechanical equipment
but you will probably want some in time.  As John said it is out there you
just have to dig for it and be aware of what you want so if it becomes
available you know what you need.

Asterisk seems to be a good product but still has drawbacks.  I myself am
still trying to find time to work on it.  With John Novacks help I am
online at 269 but rarely there to take calls ;-(

Just wanted to say  welcome you will see there are people scattered around
the world, well maybe not quite that far yet but in Eurpoe, England,
Scottland, Austrilia and New Zealand.

Good Luck,

Dennis Hock

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


To: voip at ckts.info
From: Chris Craft <n00dle at yahoo.com>
Sent by: voip-bounces at ckts.info
Date: 07/07/2007 09:51PM
Subject: [VoIP] New to list/subject.

I'm new to this list but have a few small PBXs that
are almost old enough to vote, wishing I had an older
switch to play with.  I work in the internet service
provider industry, but have some experience with telco
stuff, mainly from the PBX point ov view.  The first
PBX I was administrator for was an AT&T System25 back
in '94.
 Nowadays I'm building an Asterisk PBX to replace the
Lucent Partner system at work.  My current home pbx is
a Vodavi StarPlus SPD 1428, which I have
frankensteined together with my home Asterisk box via
a Sipura 2000 ATA.  I've been using linux since 1992
and Asterisk since 2003, so I can help people from
more switching experience with Asterisk, and maybe
some switchers can help me out with the history bits.

Cheers!
Chris Craft, who'll be reserving an office code soon.



____________________________________________________________________________________

Building a website is a piece of cake. Yahoo! Small Business gives you all
the tools to get online.
http://smallbusiness.yahoo.com/webhosting
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voipProject Web Page:
http://www.ckts.info/

From ikj1234i at yahoo.com  Mon Jul  9 08:50:59 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Mon, 9 Jul 2007 06:50:59 -0700 (PDT)
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <469129EC.8060404@comcast.net>
Message-ID: <217071.63147.qm@web51606.mail.re2.yahoo.com>


Kirt

The "PULSEAFTERTIME" is merely the interdigital delay
value.  You'll need to actually change the PULSEMAKE
and PULSEBREAK ones, which I believe are directly in
milliseconds.  

Also, should have asked this last time, does the
Varian card use stock drivers, or does it include its
own proprietary drivers?

If absolutely necessary to prove whether the values
are getting overridden by unknown actors, a 'printk'
could be added to print out the actual values in
effect...

Max

--- Kirt Stanfield <kirtley.stanfield at comcast.net>
wrote:

> Max,
> 
> I changed 'ZT_DEFAULT_PULSEAFTERTIME' to 900,
> rebuilt everything, and 
> rebooted. I see no change in the PPS. Also m/b has
> always dispalyed as 60/40
> 
> So I suspect these are in fact being overidden
> somewhere.
> 
> Kirt
> ikjtel wrote:
> 
> >Replying on-list in the event that anyone else has
> >commentary to add.............
> >
> >I believe there may be two separate issues here. 
> >First, the parameters you mention should be
> tweakable
> >via zaptel.h, specifically
> >zaptel.h:#define        ZT_DEFAULT_PULSEMAKETIME 50
>   
> > /* 50 ms of line closed when dial pulsing */
> >zaptel.h:#define        ZT_DEFAULT_PULSEBREAKTIME
> 50  
> > /* 50 ms of line open when dial pulsing */
> >zaptel.h:#define        ZT_DEFAULT_PULSEAFTERTIME
> 750 
> > /* 750ms between dial pulse digits */
> >
> >[Note, it is theoretically possible for user-space
> >(ztcfg) to override these, although I don't believe
> >this is currently implemented]
> >
> >[Note 2, I believe that these values are not
> correct
> >for true Bell System dials, the ratio should be
> ~60/40
> >not 50/50]
> >
> >The second issue is that several users have
> reported
> >erratic pulsing, which I have noticed too, just by
> >listening the problem is clearly audible.  Instead
> of
> >hearing regularly spaced pulses you hear plainly
> that
> >the spacing is irregular.
> >
> >Try connecting a butt set (in monitor mode) to the
> >connection and listen to the outpulsing.   Best way
> to
> >test this is to use a string of several 'zero'
> >digits...
> >
> >I recently was able to grab an FXO card for my
> channel
> >bank.  When it shows up (and when I get it set up)
> >I'll try to duplicate the problem.  I have released
> an
> >asterisk dial speed app that will also help
> pinpoint
> >it.  See URL
> >
> >http://www.lightlink.com/mhp/dialtest/
> >
> >Max
> >
> >--- Kirt Stanfield <kirtley.stanfield at comcast.net>
> >wrote:
> >
> >  
> >
> >>Max,
> >>
> >>I hope all is well with you.
> >>
> >>Early this year I installed a Varian T1 card in my
> >>asterisk machine, and 
> >>connected it to an adtran channel bank. The
> channel
> >>bank has 5 FXS cards 
> >>and 1 FXO card. The former is connected to ground
> >>start 2 way trunks on 
> >>my step switch. The latter is connected to the
> >>incoming selectors on my 
> >>step switch.
> >>
> >>In general the above configuration has performed
> >>well, with a tremendous 
> >>improvement in voice quality.
> >>
> >>I have however seen evidence of misdialing into
> the
> >>switch via the FXO 
> >>card. I finally wrote a php script to dial over
> and
> >>over again, and from 
> >>that I have deduced that there is about a 5-10%
> >>misdial rate. I have now 
> >>place a Metro-Tel TPM32  'digit grabber' in the
> >>circuit. That indicates 
> >>that dialing is taking place at a rate of 13pps,
> >>instead of 10pps, which 
> >>I suspect is the reason why I am seeing the error
> >>rate that I am.
> >>
> >>So my question is - can you tell me any place(s)
> in
> >>the asterisk or the 
> >>drivers that I could adjust so as to slow down the
> >>dial rate?
> >>
> >>Thanks,
> >>Kirt
> >>
> >>
> >>    
> >>
> >
> >
> >
> > 
>
>____________________________________________________________________________________
> >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/
> >
> >
> >  
> >
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 



 
____________________________________________________________________________________
Get your own web address.  
Have a HUGE year through Yahoo! Small Business.
http://smallbusiness.yahoo.com/domains/?p=BESTDEAL

From jnovack at stromberg-carlson.org  Mon Jul  9 09:21:27 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Mon, 09 Jul 2007 10:21:27 -0400
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <217071.63147.qm@web51606.mail.re2.yahoo.com>
References: <217071.63147.qm@web51606.mail.re2.yahoo.com>
Message-ID: <46924467.2020404@stromberg-carlson.org>



ikjtel wrote:
> Kirt
>
> The "PULSEAFTERTIME" is merely the interdigital delay
> value.  You'll need to actually change the PULSEMAKE
> and PULSEBREAK ones, which I believe are directly in
> milliseconds.  
>
> Also, should have asked this last time, does the
> Varian card use stock drivers, or does it include its
> own proprietary drivers?
>
>   
Varion makes a patch to the stock zaptel drivers for the tor3 card.
The earlier tor2 card substitutes one rbt file.
govarion.com/tor3 has all the information.
They supplied certain versions patched, and then, after mynag, a patch 
file for any of the zaptel 1.2 releases, given that Digium kept on 
making changes.

> If absolutely necessary to prove whether the values
> are getting overridden by unknown actors, a 'printk'
> could be added to print out the actual values in
> effect...
>
> Max
>   
Beyond my pay grade!

John Novack


> --- Kirt Stanfield <kirtley.stanfield at comcast.net>
> wrote:
>
>   
>> Max,
>>
>> I changed 'ZT_DEFAULT_PULSEAFTERTIME' to 900,
>> rebuilt everything, and 
>> rebooted. I see no change in the PPS. Also m/b has
>> always dispalyed as 60/40
>>
>> So I suspect these are in fact being overidden
>> somewhere.
>>
>> Kirt
>> ikjtel wrote:
>>
>>     
>>> Replying on-list in the event that anyone else has
>>> commentary to add.............
>>>
>>> I believe there may be two separate issues here. 
>>> First, the parameters you mention should be
>>>       
>> tweakable
>>     
>>> via zaptel.h, specifically
>>> zaptel.h:#define        ZT_DEFAULT_PULSEMAKETIME 50
>>>       
>>   
>>     
>>> /* 50 ms of line closed when dial pulsing */
>>> zaptel.h:#define        ZT_DEFAULT_PULSEBREAKTIME
>>>       
>> 50  
>>     
>>> /* 50 ms of line open when dial pulsing */
>>> zaptel.h:#define        ZT_DEFAULT_PULSEAFTERTIME
>>>       
>> 750 
>>     
>>> /* 750ms between dial pulse digits */
>>>
>>> [Note, it is theoretically possible for user-space
>>> (ztcfg) to override these, although I don't believe
>>> this is currently implemented]
>>>
>>> [Note 2, I believe that these values are not
>>>       
>> correct
>>     
>>> for true Bell System dials, the ratio should be
>>>       
>> ~60/40
>>     
>>> not 50/50]
>>>
>>> The second issue is that several users have
>>>       
>> reported
>>     
>>> erratic pulsing, which I have noticed too, just by
>>> listening the problem is clearly audible.  Instead
>>>       
>> of
>>     
>>> hearing regularly spaced pulses you hear plainly
>>>       
>> that
>>     
>>> the spacing is irregular.
>>>
>>> Try connecting a butt set (in monitor mode) to the
>>> connection and listen to the outpulsing.   Best way
>>>       
>> to
>>     
>>> test this is to use a string of several 'zero'
>>> digits...
>>>
>>> I recently was able to grab an FXO card for my
>>>       
>> channel
>>     
>>> bank.  When it shows up (and when I get it set up)
>>> I'll try to duplicate the problem.  I have released
>>>       
>> an
>>     
>>> asterisk dial speed app that will also help
>>>       
>> pinpoint
>>     
>>> it.  See URL
>>>
>>> http://www.lightlink.com/mhp/dialtest/
>>>
>>> Max
>>>
>>> --- Kirt Stanfield <kirtley.stanfield at comcast.net>
>>> wrote:
>>>
>>>  
>>>
>>>       
>>>> Max,
>>>>
>>>> I hope all is well with you.
>>>>
>>>> Early this year I installed a Varian T1 card in my
>>>> asterisk machine, and 
>>>> connected it to an adtran channel bank. The
>>>>         
>> channel
>>     
>>>> bank has 5 FXS cards 
>>>> and 1 FXO card. The former is connected to ground
>>>> start 2 way trunks on 
>>>> my step switch. The latter is connected to the
>>>> incoming selectors on my 
>>>> step switch.
>>>>
>>>> In general the above configuration has performed
>>>> well, with a tremendous 
>>>> improvement in voice quality.
>>>>
>>>> I have however seen evidence of misdialing into
>>>>         
>> the
>>     
>>>> switch via the FXO 
>>>> card. I finally wrote a php script to dial over
>>>>         
>> and
>>     
>>>> over again, and from 
>>>> that I have deduced that there is about a 5-10%
>>>> misdial rate. I have now 
>>>> place a Metro-Tel TPM32  'digit grabber' in the
>>>> circuit. That indicates 
>>>> that dialing is taking place at a rate of 13pps,
>>>> instead of 10pps, which 
>>>> I suspect is the reason why I am seeing the error
>>>> rate that I am.
>>>>
>>>> So my question is - can you tell me any place(s)
>>>>         
>> in
>>     
>>>> the asterisk or the 
>>>> drivers that I could adjust so as to slow down the
>>>> dial rate?
>>>>
>>>> Thanks,
>>>> Kirt
>>>>
>>>>
>>>>    
>>>>
>>>>         
>>>
>>>
>>>       
>> ____________________________________________________________________________________
>>     
>>> 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/
>>>
>>>
>>>  
>>>
>>>       
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>     
>
>
>
>  
> ____________________________________________________________________________________
> Get your own web address.  
> Have a HUGE year through Yahoo! Small Business.
> http://smallbusiness.yahoo.com/domains/?p=BESTDEAL
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

-- 
Dog is my co-pilot


From kirtley.stanfield at comcast.net  Mon Jul  9 20:53:49 2007
From: kirtley.stanfield at comcast.net (Kirt Stanfield)
Date: Mon, 09 Jul 2007 21:53:49 -0400
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <46924467.2020404@stromberg-carlson.org>
References: <217071.63147.qm@web51606.mail.re2.yahoo.com>
	<46924467.2020404@stromberg-carlson.org>
Message-ID: <4692E6AD.2010907@comcast.net>

Max,

I tried changing the PULSEMAKE and PULSEBREAK. It did seem to make some 
difference, I just have yet to home in on optimal values. IF these are 
in fact milliseconds it would seem that 60/40 would give 10 pps. It 
doesn't quite seem to work that way. In fact it seems to work better 
when I set to 40/60, the exact opposite of what I would think, but it 
still pulses at about 13pps. Gues I will try some simple math and ratio 
both up by 13/10 for the next try.

I do understand what you are saying about the PULSEAFTERTIME.
Any ideas?

Kirt

John Novack wrote:

>ikjtel wrote:
>  
>
>>Kirt
>>
>>The "PULSEAFTERTIME" is merely the interdigital delay
>>value.  You'll need to actually change the PULSEMAKE
>>and PULSEBREAK ones, which I believe are directly in
>>milliseconds.  
>>
>>Also, should have asked this last time, does the
>>Varian card use stock drivers, or does it include its
>>own proprietary drivers?
>>
>>  
>>    
>>
>Varion makes a patch to the stock zaptel drivers for the tor3 card.
>The earlier tor2 card substitutes one rbt file.
>govarion.com/tor3 has all the information.
>They supplied certain versions patched, and then, after mynag, a patch 
>file for any of the zaptel 1.2 releases, given that Digium kept on 
>making changes.
>
>  
>
>>If absolutely necessary to prove whether the values
>>are getting overridden by unknown actors, a 'printk'
>>could be added to print out the actual values in
>>effect...
>>
>>Max
>>  
>>    
>>
>Beyond my pay grade!
>
>John Novack
>
>
>  
>
>>--- Kirt Stanfield <kirtley.stanfield at comcast.net>
>>wrote:
>>
>>  
>>    
>>
>>>Max,
>>>
>>>I changed 'ZT_DEFAULT_PULSEAFTERTIME' to 900,
>>>rebuilt everything, and 
>>>rebooted. I see no change in the PPS. Also m/b has
>>>always dispalyed as 60/40
>>>
>>>So I suspect these are in fact being overidden
>>>somewhere.
>>>
>>>Kirt
>>>ikjtel wrote:
>>>
>>>    
>>>      
>>>
>>>>Replying on-list in the event that anyone else has
>>>>commentary to add.............
>>>>
>>>>I believe there may be two separate issues here. 
>>>>First, the parameters you mention should be
>>>>      
>>>>        
>>>>
>>>tweakable
>>>    
>>>      
>>>
>>>>via zaptel.h, specifically
>>>>zaptel.h:#define        ZT_DEFAULT_PULSEMAKETIME 50
>>>>      
>>>>        
>>>>
>>>  
>>>    
>>>      
>>>
>>>>/* 50 ms of line closed when dial pulsing */
>>>>zaptel.h:#define        ZT_DEFAULT_PULSEBREAKTIME
>>>>      
>>>>        
>>>>
>>>50  
>>>    
>>>      
>>>
>>>>/* 50 ms of line open when dial pulsing */
>>>>zaptel.h:#define        ZT_DEFAULT_PULSEAFTERTIME
>>>>      
>>>>        
>>>>
>>>750 
>>>    
>>>      
>>>
>>>>/* 750ms between dial pulse digits */
>>>>
>>>>[Note, it is theoretically possible for user-space
>>>>(ztcfg) to override these, although I don't believe
>>>>this is currently implemented]
>>>>
>>>>[Note 2, I believe that these values are not
>>>>      
>>>>        
>>>>
>>>correct
>>>    
>>>      
>>>
>>>>for true Bell System dials, the ratio should be
>>>>      
>>>>        
>>>>
>>>~60/40
>>>    
>>>      
>>>
>>>>not 50/50]
>>>>
>>>>The second issue is that several users have
>>>>      
>>>>        
>>>>
>>>reported
>>>    
>>>      
>>>
>>>>erratic pulsing, which I have noticed too, just by
>>>>listening the problem is clearly audible.  Instead
>>>>      
>>>>        
>>>>
>>>of
>>>    
>>>      
>>>
>>>>hearing regularly spaced pulses you hear plainly
>>>>      
>>>>        
>>>>
>>>that
>>>    
>>>      
>>>
>>>>the spacing is irregular.
>>>>
>>>>Try connecting a butt set (in monitor mode) to the
>>>>connection and listen to the outpulsing.   Best way
>>>>      
>>>>        
>>>>
>>>to
>>>    
>>>      
>>>
>>>>test this is to use a string of several 'zero'
>>>>digits...
>>>>
>>>>I recently was able to grab an FXO card for my
>>>>      
>>>>        
>>>>
>>>channel
>>>    
>>>      
>>>
>>>>bank.  When it shows up (and when I get it set up)
>>>>I'll try to duplicate the problem.  I have released
>>>>      
>>>>        
>>>>
>>>an
>>>    
>>>      
>>>
>>>>asterisk dial speed app that will also help
>>>>      
>>>>        
>>>>
>>>pinpoint
>>>    
>>>      
>>>
>>>>it.  See URL
>>>>
>>>>http://www.lightlink.com/mhp/dialtest/
>>>>
>>>>Max
>>>>
>>>>--- Kirt Stanfield <kirtley.stanfield at comcast.net>
>>>>wrote:
>>>>
>>>> 
>>>>
>>>>      
>>>>        
>>>>
>>>>>Max,
>>>>>
>>>>>I hope all is well with you.
>>>>>
>>>>>Early this year I installed a Varian T1 card in my
>>>>>asterisk machine, and 
>>>>>connected it to an adtran channel bank. The
>>>>>        
>>>>>          
>>>>>
>>>channel
>>>    
>>>      
>>>
>>>>>bank has 5 FXS cards 
>>>>>and 1 FXO card. The former is connected to ground
>>>>>start 2 way trunks on 
>>>>>my step switch. The latter is connected to the
>>>>>incoming selectors on my 
>>>>>step switch.
>>>>>
>>>>>In general the above configuration has performed
>>>>>well, with a tremendous 
>>>>>improvement in voice quality.
>>>>>
>>>>>I have however seen evidence of misdialing into
>>>>>        
>>>>>          
>>>>>
>>>the
>>>    
>>>      
>>>
>>>>>switch via the FXO 
>>>>>card. I finally wrote a php script to dial over
>>>>>        
>>>>>          
>>>>>
>>>and
>>>    
>>>      
>>>
>>>>>over again, and from 
>>>>>that I have deduced that there is about a 5-10%
>>>>>misdial rate. I have now 
>>>>>place a Metro-Tel TPM32  'digit grabber' in the
>>>>>circuit. That indicates 
>>>>>that dialing is taking place at a rate of 13pps,
>>>>>instead of 10pps, which 
>>>>>I suspect is the reason why I am seeing the error
>>>>>rate that I am.
>>>>>
>>>>>So my question is - can you tell me any place(s)
>>>>>        
>>>>>          
>>>>>
>>>in
>>>    
>>>      
>>>
>>>>>the asterisk or the 
>>>>>drivers that I could adjust so as to slow down the
>>>>>dial rate?
>>>>>
>>>>>Thanks,
>>>>>Kirt
>>>>>
>>>>>
>>>>>   
>>>>>
>>>>>        
>>>>>          
>>>>>
>>>>      
>>>>        
>>>>
>>>____________________________________________________________________________________
>>>    
>>>      
>>>
>>>>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/
>>>>
>>>>
>>>> 
>>>>
>>>>      
>>>>        
>>>>
>>>_______________________________________________
>>>VoIP mailing list
>>>VoIP at ckts.info
>>>http://lists.ckts.info/mailman/listinfo/voip
>>>Project Web Page: http://www.ckts.info/
>>>
>>>    
>>>      
>>>
>>
>> 
>>____________________________________________________________________________________
>>Get your own web address.  
>>Have a HUGE year through Yahoo! Small Business.
>>http://smallbusiness.yahoo.com/domains/?p=BESTDEAL
>>_______________________________________________
>>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  Mon Jul  9 21:50:15 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Mon, 9 Jul 2007 19:50:15 -0700 (PDT)
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <4692E6AD.2010907@comcast.net>
Message-ID: <860441.28903.qm@web51604.mail.re2.yahoo.com>


--- Kirt Stanfield <kirtley.stanfield at comcast.net>
wrote:

> Max,
> 
> I tried changing the PULSEMAKE and PULSEBREAK. It
> did seem to make some 
> difference, I just have yet to home in on optimal
> values. IF these are 
> in fact milliseconds it would seem that 60/40 would
> give 10 pps. It 
> doesn't quite seem to work that way. In fact it
> seems to work better 
> when I set to 40/60, the exact opposite of what I
> would think, but it 
> still pulses at about 13pps. Gues I will try some
> simple math and ratio 
> both up by 13/10 for the next try.

OK, the above is ambiguous.  Can't tell which one
(MAKE or BREAK) you're setting to 60 and which to 40.

I would say that proper values for Bell System dials
would be
PULSEBREAK=60
PULSEMAKE=40

(FWIW my ITT/Kellogg TIMM-2 recommends 62/38 for the
WE-clone dials used in their line of 500-series sets)

Also, still waiting for you to answer what the pulsing
sounds like (regular or erratic) from a monitor set...

Also, how much do you trust your 13pps figure???  The
amount of error (30%) is suspicious.  You should be
able to compare the speed against a known dial...

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

From stfkerman at jps.net  Mon Jul  9 22:34:37 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Mon, 09 Jul 2007 23:34:37 -0400
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <860441.28903.qm@web51604.mail.re2.yahoo.com>
References: <860441.28903.qm@web51604.mail.re2.yahoo.com>
Message-ID: <4692FE4D.1020502@jps.net>

FWIW, 62/38 is the spec I always saw stated by AE.  Bell docs I saw 
almost always stated 60/40, though I think on rare occasions I have seen 
59/41 too. 

Perhaps ITT was trying to meet the AE spec based where they expected 
their phones to be used, notwithstanding the cosmetic resemblance to a 
WECo 500 set.

SK

ikjtel wrote:
> (FWIW my ITT/Kellogg TIMM-2 recommends 62/38 for the
> WE-clone dials used in their line of 500-series sets)
>
>
>   
>

From kirtley.stanfield at comcast.net  Tue Jul 10 07:20:06 2007
From: kirtley.stanfield at comcast.net (Kirt Stanfield)
Date: Tue, 10 Jul 2007 08:20:06 -0400
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <860441.28903.qm@web51604.mail.re2.yahoo.com>
References: <860441.28903.qm@web51604.mail.re2.yahoo.com>
Message-ID: <46937976.4080903@comcast.net>

Max,

Sorry if I wasn't clear - I will try to be more careful in the future-

The good news is that I experimentally verified what you said. The 
following settings seemed to improve the dialing accuracy a bit:

PULSEBREAK=60
PULSEMAKE=40

Now one odd thing - the Digit Grabber still reports 13 PPS.

So I tried another variant - I multiplied each of the above by 13/10 on 
the assumption that if they are if fact absolute timings that doing so 
would get me closer to 10 PPS. That did NOT work, or at least the digit 
grabber still reported 13 PPS. Two possible conclusions:

1. The digit grabber is wrong.
2. The above values are some sort of % instead of absolute millisecond 
timings.


Kirt


ikjtel wrote:

>--- Kirt Stanfield <kirtley.stanfield at comcast.net>
>wrote:
>
>  
>
>>Max,
>>
>>I tried changing the PULSEMAKE and PULSEBREAK. It
>>did seem to make some 
>>difference, I just have yet to home in on optimal
>>values. IF these are 
>>in fact milliseconds it would seem that 60/40 would
>>give 10 pps. It 
>>doesn't quite seem to work that way. In fact it
>>seems to work better 
>>when I set to 40/60, the exact opposite of what I
>>would think, but it 
>>still pulses at about 13pps. Gues I will try some
>>simple math and ratio 
>>both up by 13/10 for the next try.
>>    
>>
>
>OK, the above is ambiguous.  Can't tell which one
>(MAKE or BREAK) you're setting to 60 and which to 40.
>
>I would say that proper values for Bell System dials
>would be
>PULSEBREAK=60
>PULSEMAKE=40
>
>(FWIW my ITT/Kellogg TIMM-2 recommends 62/38 for the
>WE-clone dials used in their line of 500-series sets)
>
>Also, still waiting for you to answer what the pulsing
>sounds like (regular or erratic) from a monitor set...
>
>Also, how much do you trust your 13pps figure???  The
>amount of error (30%) is suspicious.  You should be
>able to compare the speed against a known dial...
>
>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
>
>
>  
>

From kxt at fubegra.net  Tue Jul 10 07:18:56 2007
From: kxt at fubegra.net (Russ Price)
Date: Tue, 10 Jul 2007 07:18:56 -0500
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <860441.28903.qm@web51604.mail.re2.yahoo.com>
References: <860441.28903.qm@web51604.mail.re2.yahoo.com>
Message-ID: <46937930.2010100@fubegra.net>

ikjtel wrote:
> 
> I would say that proper values for Bell System dials
> would be
> PULSEBREAK=60
> PULSEMAKE=40
> 
> Also, still waiting for you to answer what the pulsing
> sounds like (regular or erratic) from a monitor set...

Back in April, I tried (and posted) my results for changing 
PULSEMAKETIME and PULSEBREAKTIME - using 60:40 or 67:33 (UK standard) 
made things even more erratic to my ear.

I've monitored FXS and FXO channels using ZapBarge (I don't have a 
buttinsky yet), and if I monitor a 500 set connected to an Adtran FXS, 
the pulses are nice and even as they should be.  Only the channel bank 
FXO ports have a problem.

	Russ

From ikj1234i at yahoo.com  Tue Jul 10 13:23:34 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Tue, 10 Jul 2007 11:23:34 -0700 (PDT)
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <46937976.4080903@comcast.net>
Message-ID: <20070710182334.57010.qmail@web51603.mail.re2.yahoo.com>


--- Kirt Stanfield <kirtley.stanfield at comcast.net>
wrote:
> grabber still reported 13 PPS. Two possible
> conclusions:
> 
> 1. The digit grabber is wrong.
> 2. The above values are some sort of % instead of
> absolute millisecond 
> timings.

OK.   These values are indeed absolute quantities in
milliseconds - they're NOT relative nor percent etc.

However, I still want to know if you can listen to the
pulsing - it seems likely that you'll also experience
the erratic pulsing - in which case the monitor may be
trying its best to decode a bad pulse stream...

Max


 
____________________________________________________________________________________
Don't pick lemons.
See all the new 2007 cars at Yahoo! Autos.
http://autos.yahoo.com/new_cars.html 

From martin at Princeton.EDU  Tue Jul 10 14:47:04 2007
From: martin at Princeton.EDU (Martin Harriss)
Date: Tue, 10 Jul 2007 15:47:04 -0400
Subject: [VoIP] Asterisk compatable boards
Message-ID: <4693E238.2070206@Princeton.EDU>

Gang,

Came across this company recently: http://www.zapmicro.com/

Don't know anything about them, so this is no recommendation, but their 
prices sure look good.

Scuttlebutt is that more companies in China will soon be doing likewise...


Martin


From lee at spenadel.com  Tue Jul 10 15:59:57 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Tue, 10 Jul 2007 16:59:57 -0400
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <46937976.4080903@comcast.net>
References: <860441.28903.qm@web51604.mail.re2.yahoo.com>
	<46937976.4080903@comcast.net>
Message-ID: <031401c7c335$46053d60$d20fb820$@com>

BTW, the Kerman message has been updated at 349-0666.

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Kirt Stanfield
Sent: Tuesday, July 10, 2007 8:20 AM
To: ikjtel
Cc: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Asterisk-Adtran dial rate

Max,

Sorry if I wasn't clear - I will try to be more careful in the future-

The good news is that I experimentally verified what you said. The 
following settings seemed to improve the dialing accuracy a bit:

PULSEBREAK=60
PULSEMAKE=40

Now one odd thing - the Digit Grabber still reports 13 PPS.

So I tried another variant - I multiplied each of the above by 13/10 on 
the assumption that if they are if fact absolute timings that doing so 
would get me closer to 10 PPS. That did NOT work, or at least the digit 
grabber still reported 13 PPS. Two possible conclusions:

1. The digit grabber is wrong.
2. The above values are some sort of % instead of absolute millisecond 
timings.


Kirt


ikjtel wrote:

>--- Kirt Stanfield <kirtley.stanfield at comcast.net>
>wrote:
>
>  
>
>>Max,
>>
>>I tried changing the PULSEMAKE and PULSEBREAK. It
>>did seem to make some 
>>difference, I just have yet to home in on optimal
>>values. IF these are 
>>in fact milliseconds it would seem that 60/40 would
>>give 10 pps. It 
>>doesn't quite seem to work that way. In fact it
>>seems to work better 
>>when I set to 40/60, the exact opposite of what I
>>would think, but it 
>>still pulses at about 13pps. Gues I will try some
>>simple math and ratio 
>>both up by 13/10 for the next try.
>>    
>>
>
>OK, the above is ambiguous.  Can't tell which one
>(MAKE or BREAK) you're setting to 60 and which to 40.
>
>I would say that proper values for Bell System dials
>would be
>PULSEBREAK=60
>PULSEMAKE=40
>
>(FWIW my ITT/Kellogg TIMM-2 recommends 62/38 for the
>WE-clone dials used in their line of 500-series sets)
>
>Also, still waiting for you to answer what the pulsing
>sounds like (regular or erratic) from a monitor set...
>
>Also, how much do you trust your 13pps figure???  The
>amount of error (30%) is suspicious.  You should be
>able to compare the speed against a known dial...
>
>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 lee at spenadel.com  Tue Jul 10 16:01:26 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Tue, 10 Jul 2007 17:01:26 -0400
Subject: [VoIP] Asterisk-Adtran dial rate
In-Reply-To: <031401c7c335$46053d60$d20fb820$@com>
References: <860441.28903.qm@web51604.mail.re2.yahoo.com>	<46937976.4080903@comcast.net>
	<031401c7c335$46053d60$d20fb820$@com>
Message-ID: <031501c7c335$7b1fe7c0$715fb740$@com>

Never mind, a mistake.

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Lee Spenadel
Sent: Tuesday, July 10, 2007 5:00 PM
To: 'Voice Over IP Tandem for Analog Switches'
Subject: Re: [VoIP] Asterisk-Adtran dial rate

BTW, the Kerman message has been updated at 349-0666.

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Kirt Stanfield
Sent: Tuesday, July 10, 2007 8:20 AM
To: ikjtel
Cc: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] Asterisk-Adtran dial rate

Max,

Sorry if I wasn't clear - I will try to be more careful in the future-

The good news is that I experimentally verified what you said. The 
following settings seemed to improve the dialing accuracy a bit:

PULSEBREAK=60
PULSEMAKE=40

Now one odd thing - the Digit Grabber still reports 13 PPS.

So I tried another variant - I multiplied each of the above by 13/10 on 
the assumption that if they are if fact absolute timings that doing so 
would get me closer to 10 PPS. That did NOT work, or at least the digit 
grabber still reported 13 PPS. Two possible conclusions:

1. The digit grabber is wrong.
2. The above values are some sort of % instead of absolute millisecond 
timings.


Kirt


ikjtel wrote:

>--- Kirt Stanfield <kirtley.stanfield at comcast.net>
>wrote:
>
>  
>
>>Max,
>>
>>I tried changing the PULSEMAKE and PULSEBREAK. It
>>did seem to make some 
>>difference, I just have yet to home in on optimal
>>values. IF these are 
>>in fact milliseconds it would seem that 60/40 would
>>give 10 pps. It 
>>doesn't quite seem to work that way. In fact it
>>seems to work better 
>>when I set to 40/60, the exact opposite of what I
>>would think, but it 
>>still pulses at about 13pps. Gues I will try some
>>simple math and ratio 
>>both up by 13/10 for the next try.
>>    
>>
>
>OK, the above is ambiguous.  Can't tell which one
>(MAKE or BREAK) you're setting to 60 and which to 40.
>
>I would say that proper values for Bell System dials
>would be
>PULSEBREAK=60
>PULSEMAKE=40
>
>(FWIW my ITT/Kellogg TIMM-2 recommends 62/38 for the
>WE-clone dials used in their line of 500-series sets)
>
>Also, still waiting for you to answer what the pulsing
>sounds like (regular or erratic) from a monitor set...
>
>Also, how much do you trust your 13pps figure???  The
>amount of error (30%) is suspicious.  You should be
>able to compare the speed against a known dial...
>
>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/

_______________________________________________
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 Jul 10 16:10:21 2007
Fr