From ka2wft at arrl.net  Thu Feb  1 19:48:03 2007
From: ka2wft at arrl.net (Doug Alderdice)
Date: Thu, 01 Feb 2007 20:48:03 -0500
Subject: [VoIP] More on the VT1005 Dial Plan
Message-ID: <5.1.0.14.0.20070201203601.00b826d0@incoming.verizon.net>

I think I finally have it!

It took some studying of the remote telnet console of the VT1005, which 
wasn't easy because it can belch out reams of info at a time and it isn't 
always possible to scroll back to find what you want to see.  These are the 
dial plan lines that work for seven-digit CNET stateside dialing (my 
Asterisk extensions.conf adds the leading '1'), as well as for 011+ dialing 
for the overseas CNET folks:

BTIOPT TeleSipDialPlanEntry[0] = "DIGITMAP:011XXXXXXXXXXXX"
BTIOPT TeleSipDialPlanEntry[0] = "DIGITMAP:[2-9]XXXXXX"
BTIOPT TeleSipDialPlanEntry[0] = "DIGITMAP:*XX"

The third line comes with the sample and allows you access to the *XX 
functions like block CID, call forwarding, etc.

It is really strange the way the VT1005 starts to use a new dial plan, 
though.  It loads the .bin file from the tftp server, and if you query 
through the remote console "TELEsipConfigShow" it'll spit back what it just 
loaded, however it won't necessarily use it right away on your next 
call!  What appeared to happen is that no calls could be placed at 
first.  Dial a zero and get reorder from the ATA.  Dial a digit (not a 
zero) and get reorder, but then on the remote console the ATA then appears 
to process and load the dial plan into memory/working space/whatever.  Next 
call attempt to any number goes through.

The same thing happened when I changed the timeout setting slightly on line 
1.  Next call attempt on line 1 starting with a zero went to 
reorder.  Tried dialing a non-zero digit and got reorder.  Remote console 
belched out dial plan processing progress.  Next call attempt went through 
with the new timing.

FWIW
YMMV

Doug.


From jjones3601 at yahoo.com  Thu Feb  1 20:22:10 2007
From: jjones3601 at yahoo.com (john jones)
Date: Thu, 1 Feb 2007 18:22:10 -0800 (PST)
Subject: [VoIP] More on the VT1005 Dial Plan
In-Reply-To: <5.1.0.14.0.20070201203601.00b826d0@incoming.verizon.net>
Message-ID: <20070202022213.29219.qmail@web34301.mail.mud.yahoo.com>

That is truly bizarre.  Did you play around any more
with the digit timing or did you leave all of those
settings at 4000?


Thanks for the investigative work!

Has your impression of this device improved or is the
jury still out?


John



--- Doug Alderdice <ka2wft at arrl.net> wrote:

> I think I finally have it!
> 
> It took some studying of the remote telnet console
> of the VT1005, which 
> wasn't easy because it can belch out reams of info
> at a time and it isn't 
> always possible to scroll back to find what you want
> to see.  These are the 
> dial plan lines that work for seven-digit CNET
> stateside dialing (my 
> Asterisk extensions.conf adds the leading '1'), as
> well as for 011+ dialing 
> for the overseas CNET folks:
> 
> BTIOPT TeleSipDialPlanEntry[0] =
> "DIGITMAP:011XXXXXXXXXXXX"
> BTIOPT TeleSipDialPlanEntry[0] =
> "DIGITMAP:[2-9]XXXXXX"
> BTIOPT TeleSipDialPlanEntry[0] = "DIGITMAP:*XX"
> 
> The third line comes with the sample and allows you
> access to the *XX 
> functions like block CID, call forwarding, etc.
> 
> It is really strange the way the VT1005 starts to
> use a new dial plan, 
> though.  It loads the .bin file from the tftp
> server, and if you query 
> through the remote console "TELEsipConfigShow" it'll
> spit back what it just 
> loaded, however it won't necessarily use it right
> away on your next 
> call!  What appeared to happen is that no calls
> could be placed at 
> first.  Dial a zero and get reorder from the ATA. 
> Dial a digit (not a 
> zero) and get reorder, but then on the remote
> console the ATA then appears 
> to process and load the dial plan into
> memory/working space/whatever.  Next 
> call attempt to any number goes through.
> 
> The same thing happened when I changed the timeout
> setting slightly on line 
> 1.  Next call attempt on line 1 starting with a zero
> went to 
> reorder.  Tried dialing a non-zero digit and got
> reorder.  Remote console 
> belched out dial plan processing progress.  Next
> call attempt went through 
> with the new timing.
> 
> FWIW
> YMMV
> 
> Doug.
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From ka2wft at arrl.net  Thu Feb  1 20:53:53 2007
From: ka2wft at arrl.net (Doug Alderdice)
Date: Thu, 01 Feb 2007 21:53:53 -0500
Subject: [VoIP] More on the VT1005 Dial Plan
In-Reply-To: <20070202022213.29219.qmail@web34301.mail.mud.yahoo.com>
References: <5.1.0.14.0.20070201203601.00b826d0@incoming.verizon.net>
Message-ID: <5.1.0.14.0.20070201213337.02e74598@incoming.verizon.net>

At 06:22 PM 2/1/2007 -0800, John Jones wrote:
>That is truly bizarre.  Did you play around any more with the digit timing 
>or did you leave all of those settings at 4000?

I presently have them set:
BTIOPT TeleSipCritialDialTimeout[0] = 2000
BTIOPT TeleSipCritialDialTimeout[1] = 2000
BTIOPT TeleSipPartialDialTimeout[0] = 3000
BTIOPT TeleSipPartialDialTimeout[1] = 3000

I may set the 3000 value back to 2000, might even try 1500 or 1750.

>Thanks for the investigative work!
>
>Has your impression of this device improved or is the jury still out?

It is much improved, at least for the rotary dialing support.  I have so 
far had little trouble with my two test phones here, the older WE500 and 
the AE 40.  Now that 011+ dialing is working, this weekend I think I will 
swap the CNET ports from my switch that presently go to the SmarT-1/Sipura 
2000 kludge to the VT1005 and then I can test from all the stations on the 
switch, which encompass a number of m'frs and vintages of rotary dial 
sets.  Oh BTW, the ringing frequency of the ATA appears to be 20 Hz because 
the AE40 has a 20 cycle frequency ringer and it rings just fine on the 
Motorola.

As for configuration, gimme the Sipura.  Set the server, dial plan, click 
"save" and you're done.  This Motorola is something else.  What I didn't 
mention earlier that was VERY strange when I originally got the 011+ 
dialing working on Line 2.  Copied the dial plan from Line 2 to the Line 1 
section.  Did the dial-a-non-zero-digit thing after the new config file 
loaded and I could dial nicely on both lines.  Changed the timing for Line 
1 (which was still at the 4000 values) and after the reload Line 1 wouldn't 
work at all, even after the dial-a-digit thing.  Got looking at the config 
file and found that I forgotten to change the line number argument when I 
copied & pasted the values from the Line 2 section to the Line 
1.  Explained why it didn't work then, but it HAD worked previous to my 
changing of the timing values!  Very bizarre.

Also, the only way I knew that a dial plan entry didn't work was to watch 
the remote console as I dialed a digit to see if the entry was parsed.  I 
was trying to use an open ended entry like "011.T" to say after the 011 
there will be any number of digits until it times out.  That looked like it 
would work from the examples of the other dial plan entries.  Watching the 
ATA parse the dial plan when you enter a digit told otherwise, and was the 
only way you could know the entry was invalid.  The manual leaves a little 
to be desired on specifics of the dial plan arguments.  It shows only 
examples, and no full syntax of what you can or can't use in each type of 
statement.

Now that the ATA is configured I am looking forward to trying it out with 
the rotary sets here.  I just hope the dial plan doesn't need tweaking any 
time soon! :-)

Doug.





From greg at vyger.net  Fri Feb  2 16:39:59 2007
From: greg at vyger.net (Greg Blakely)
Date: Fri, 2 Feb 2007 16:39:59 -0600
Subject: [VoIP] More on the VT1005 Dial Plan
Message-ID: <A691C3685A152E4AAC9608B8D8E11316686162@godzilla.hickory.vyger.net>

Okay.  You're  having way too much fun.  So, I just now ordered one for
myself.

> 
> I presently have them set:
> BTIOPT TeleSipCritialDialTimeout[0] = 2000 BTIOPT 
> TeleSipCritialDialTimeout[1] = 2000 BTIOPT 
> TeleSipPartialDialTimeout[0] = 3000 BTIOPT 
> TeleSipPartialDialTimeout[1] = 3000
> 


From ka2wft at arrl.net  Sat Feb  3 11:07:05 2007
From: ka2wft at arrl.net (Doug Alderdice)
Date: Sat, 03 Feb 2007 12:07:05 -0500
Subject: [VoIP] Alan MacDonald
Message-ID: <5.1.0.14.0.20070203120424.02715b28@incoming.verizon.net>

Sorry to use the list this way, but I don't have Alan's email address.

Alan, tried to ring you back just now (1700 UTC) but I am getting no audio 
from your system on any of your numbers.  I get msgs on the * console that 
the connections get set up, but I don't hear anything.  I am hearing audio 
from others' systems OK.

Doug.


From g4vft at btinternet.com  Sat Feb  3 13:13:40 2007
From: g4vft at btinternet.com (Jonathan Kay)
Date: Sat, 03 Feb 2007 19:13:40 +0000
Subject: [VoIP] Alan MacDonald
In-Reply-To: <5.1.0.14.0.20070203120424.02715b28@incoming.verizon.net>
References: <5.1.0.14.0.20070203120424.02715b28@incoming.verizon.net>
Message-ID: <45C4DEE4.5050808@btinternet.com>

Doug Alderdice wrote:
> Sorry to use the list this way, but I don't have Alan's email address.
>
> Alan, tried to ring you back just now (1700 UTC) but I am getting no audio 
> from your system on any of your numbers.  I get msgs on the * console that 
> the connections get set up, but I don't hear anything.  I am hearing audio 
> from others' systems OK.
>
> Doug.
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

This is a strange recurrent problem we are getting with Al's Asterisk.
He's using a x100p clone for his FXO interface.
We've set up a Cron job, to restart Asterisk at 4am every day, but might 
have to reboot the whole box with a Cron job.


Jon K

From ka2wft at arrl.net  Sat Feb  3 13:35:23 2007
From: ka2wft at arrl.net (Doug Alderdice)
Date: Sat, 03 Feb 2007 14:35:23 -0500
Subject: [VoIP] Dialing Tests w/Motorola VT1005
Message-ID: <5.1.0.14.0.20070203140507.02768198@incoming.verizon.net>

I have been playing around with the VT1005 and the phones here and I am 
impressed!  The Motorola VT1005 does remarkably well with the range of 
vintage dial sets connected to my step switch.

I connected one of the CNET ports from the switch to Line 1 of the VT1005 
last night and have been testing and trying the step switch stations 
scattered about the premises here.  The phones range in age from a pre-1930 
WE 51AL dial candlestick with #2 dial to late WE500s with #9 dials.  I have 
had only two dialing errors, one was with a call attempt on my ComKey 416 
set, but a subsequent call placed went through w/o errors.  The other was 
with my WE 460 multiline set.  Again, got an error on one call, next call 
attempt went through OK.

What was most interesting is that I had no problems dialing from a SC 1543 
set with a really pokey dial.  I need to send it to Steve Hilsz for some 
rehab, but I got the number I dialed from that set via the VT1005 on the 
first attempt.

The rotary dial sets I have tried are:
* WE51AL 'stick w/#2 dial
* Various WE500s with #7 and #9 dials
* ComKey 416 rotary dial Master set, 1 error
* WE440EC-3 set connected to a 1A key system
* WE460-series, 1 error
* Federal 802A1 desk set (302 look-alike)
* NE Contempra
* AE80
* AE40 (3 different sets)
* SC 1543 (2 different sets, both w/sluggish dials)
* SC 1243
* WE701 Princess
* Connecticut "Toaster" phone
* CTE (North 7H6 look alike) w/AE dial
* WE830 connected to a 1A2 key system
* WE302 w/#5 dial
* Leich 105
* WE302 w/#6 dial, can't dial out.  Must be pulsing faster than 10 pps or 
m/b ratio is out of whack, will not break dial tone on the VT1005, though 
will dial OK on the step switch, and the SmarT-1/Sipura setup.
* WE5302 w/20 pps dial -- won't break dial tone on VT1005, but doesn't work 
with the SmartT-1, either

All of the phones are in "as found" condition in terms of their dials.  I 
do have a 302 with a "Steved" dial, but it is currently in pieces and not 
operational awaiting the metal case from a friend who is repainting it for 
me.  In short, the VT1005 makes a good accounting of itself with the phones 
here.  It looks like a winner for those who want to avoid the SmarT-1 
pulse-to-DTMF conversion required with other ATAs.

Cheers,

Doug.


From ka2wft at arrl.net  Sat Feb  3 13:40:06 2007
From: ka2wft at arrl.net (Doug Alderdice)
Date: Sat, 03 Feb 2007 14:40:06 -0500
Subject: [VoIP] Alan MacDonald
In-Reply-To: <45C4DEE4.5050808@btinternet.com>
References: <5.1.0.14.0.20070203120424.02715b28@incoming.verizon.net>
	<5.1.0.14.0.20070203120424.02715b28@incoming.verizon.net>
Message-ID: <5.1.0.14.0.20070203143610.0275c398@incoming.verizon.net>

At 07:13 PM 2/3/2007 +0000, Jon Kay wrote:
>Doug Alderdice wrote:
> > Sorry to use the list this way, but I don't have Alan's email address.
> >
> > Alan, tried to ring you back just now (1700 UTC) but I am getting no audio
> > from your system on any of your numbers.  I get msgs on the * console that
> > the connections get set up, but I don't hear anything.  I am hearing audio
> > from others' systems OK.
> >
> > Doug.
>
>This is a strange recurrent problem we are getting with Al's Asterisk.
>He's using a x100p clone for his FXO interface.
>We've set up a Cron job, to restart Asterisk at 4am every day, but might
>have to reboot the whole box with a Cron job.

I've noticed the no audio problem on Alan's system occurs a lot, but have 
also encountered it from time to time on others' systems.  As it turns out, 
he rang me back just as I was hitting "send" on my previous message, and we 
had a nice chat.

As a number of us are running the X100P clone cards without the no audio 
problem occurring that much, if at all, I wonder if Alan's problem is 
related to the version of Asterisk and/or Zaptel driver he is running?

Doug.



From g4vft at btinternet.com  Sat Feb  3 13:52:34 2007
From: g4vft at btinternet.com (Jonathan Kay)
Date: Sat, 03 Feb 2007 19:52:34 +0000
Subject: [VoIP] Alan MacDonald
In-Reply-To: <5.1.0.14.0.20070203143610.0275c398@incoming.verizon.net>
References: <5.1.0.14.0.20070203120424.02715b28@incoming.verizon.net>	<5.1.0.14.0.20070203120424.02715b28@incoming.verizon.net>
	<5.1.0.14.0.20070203143610.0275c398@incoming.verizon.net>
Message-ID: <45C4E802.8060407@btinternet.com>


>
> I've noticed the no audio problem on Alan's system occurs a lot, but have 
> also encountered it from time to time on others' systems.  As it turns out, 
> he rang me back just as I was hitting "send" on my previous message, and we 
> had a nice chat.
>
> As a number of us are running the X100P clone cards without the no audio 
> problem occurring that much, if at all, I wonder if Alan's problem is 
> related to the version of Asterisk and/or Zaptel driver he is running?
>
> Doug.
>
>
> _

It's an idea. The box was built about 16 months ago, and hasn't been 
changed since.
Hopefully the FXS patch still works on the later zaptel relaeses. 
Because he's using a patched X100 for his FXS interface.

Jon K

From ian at uax.org.uk  Sun Feb  4 06:49:13 2007
From: ian at uax.org.uk (Ian Jolly)
Date: Sun, 4 Feb 2007 12:49:13 -0000
Subject: [VoIP] Anyone any ideas??
Message-ID: <01df01c7485a$dfe6f460$0a01a8c0@acer1dd0bbc6d0>

I'm trying to set another Asterisk box up but have run into a problem.

The PC is an Acer Power F6 
a.. Intel Pentium 4 524 3.06GHz 
a.. 256MB DDR2 400 PC3200 
a.. 80GB-7200 SATA 

The problem comes when I try to load Linux - it can't see the hard drive. It would appear that the newer S-ATA drives are not recognised by Linux. 

Anyone had any experience with this?

Ian Jolly


+44 (0)352 82 26 (via a 1929 GPO Rural Automatic eXchange!) 
CNET - the Heritage Telephone Network

From bbj at innismir.net  Sun Feb  4 06:55:01 2007
From: bbj at innismir.net (Ben Jackson)
Date: Sun, 04 Feb 2007 07:55:01 -0500
Subject: [VoIP] Anyone any ideas??
In-Reply-To: <01df01c7485a$dfe6f460$0a01a8c0@acer1dd0bbc6d0>
References: <01df01c7485a$dfe6f460$0a01a8c0@acer1dd0bbc6d0>
Message-ID: <45C5D7A5.8000909@innismir.net>

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

Ian Jolly wrote:

> a.. 80GB-7200 SATA 
> 
> The problem comes when I try to load Linux - it can't see the hard drive. It would appear that the newer S-ATA drives are not recognised by Linux. 
> 
> Anyone had any experience with this?

What Linux are you installing? SATA support is there, it's just only
available in the newer kernel versions.

				~Ben


- --
/"\  Ben Jackson - N1WBV - New Bedford, MA
\ /  bbj <at> innismir.net - http://www.innismir.net/
 X   Member of the ASCII Ribbon Campaign Against HTML Mail
/ \
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From ian at uax.org.uk  Sun Feb  4 07:07:33 2007
From: ian at uax.org.uk (Ian Jolly)
Date: Sun, 4 Feb 2007 13:07:33 -0000
Subject: [VoIP] +64 Country Code
Message-ID: <01f801c7485d$6f8fec50$0a01a8c0@acer1dd0bbc6d0>

An update

John in Auckland on the north island of New Zealand has been busy over the last week playing with an IAX softphone (DIAX) set up on his PC and hosted off my Asterisk box in the UK. 

I've seen him trying different recorded services on switches in the UK and the 'colonies' :-)  I've hear my old EM switch burst into life at odd hours and when I've checked it has been John in NZ.

Last night I had my first telephone conversation with John.  It seemed odd as I was chatting to him on Saturday evening whilst it was early Sunday afternoon with him - they are 13 hours ahead of the UK with their daylight saving time.

The conversation dropped out once. But once re-established lasted for a couple of hours. Latency made the call a bit difficult - almost needed to use 'over' as in two-way radio!.  Little bit of echo about two seconds later but at quite a low level. 

John is now going to get an ATA so that he can be permanently on without the need for leaving his PC on.

He has a Strowger PABX at home but it is not up and running. It sounds like a GPO type PABX No 1 as we knew them in the UK.

John started his working life with New Zealand Post Office Telephones and still works for one of their descendants. He is also involved at a preserved steam railway where he has a former NZ PO Telephones UAX13 strowger switch similar to those we have on CNET in the UK. Not only did he acquire the switch but he got the complete building as well !!  The exchange needs cabling - any volunteers ??  :-)

He also thinks that some other telephone folk in NZ will be interested in connecting to CNET. - maybe an NZ /411 page one day ? :-)

Ian Jolly

+44 (0)352 82 26 (via a 1929 GPO Rural Automatic eXchange!) 
CNET - the Heritage Telephone Network

From bbj at innismir.net  Sun Feb  4 10:06:12 2007
From: bbj at innismir.net (Ben Jackson)
Date: Sun, 04 Feb 2007 11:06:12 -0500
Subject: [VoIP] Anyone any ideas??
In-Reply-To: <020b01c74862$519ee070$0a01a8c0@acer1dd0bbc6d0>
References: <E1HDhZ9-0004vw-Gt@telstar.blackratchet.org>
	<020b01c74862$519ee070$0a01a8c0@acer1dd0bbc6d0>
Message-ID: <45C60474.5080507@innismir.net>

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

Ian Jolly wrote:
> Fedora Core
> 
> I think most of us in the UK were/are using Fedora Core 4  - I'll look
> and see what the latest version is and give that a try
> 
> I only wondered if you had tried one with a S-ATA hard drive.
> 
> I'll see how it goes and let you know
> 
Apparently I forgot to CC the list. Whoops

I think the current Fedora Core version is 6. A few quick googles shows
that it does support SATA. The only issue might be if it doesn't support
your exact SATA controller. However, I would think that wouldn't be the
case.

				~Ben

- --
/"\  Ben Jackson - N1WBV - New Bedford, MA
\ /  bbj <at> innismir.net - http://www.innismir.net/
 X   Member of the ASCII Ribbon Campaign Against HTML Mail
/ \
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From kirtley.stanfield at comcast.net  Sun Feb  4 10:10:50 2007
From: kirtley.stanfield at comcast.net (Kirt Stanfield)
Date: Sun, 04 Feb 2007 11:10:50 -0500
Subject: [VoIP] T1 related Problem
Message-ID: <45C6058A.5040401@comcast.net>

An interesting new wrinkle-

As some of you folks know I intsalled a T1 between my Asterisk box and 
my step switch a little over a month ago. This has been working fine, 
and in particular solved some bad echo problems I had been experiencing 
with the Digium card.

Just to confirm:

I can call out from my step switch over my ground start trunk just fine 
(uses T1 FXS).
CNET folks can call into  my switch just fine (uses T1 FXO).
I can call between my SIP phone and the step switch in both directions 
with no problems.

Last night i tried dialing 9 (goes out on T1 FXS)on my switch and 
calling back into it via the T1 FXO card on my T1. I heard some funny 
pulsing (like the first pulse was showing up somewhere in the middle of 
the 7, then a 1) from the switch room and then nothing. I examined the 
asterisk logs and saw the following:

-- Goto (internal,571,1)
-- Executing Macro("Zap/25-1", "dialswitch|571") in new stack
-- Executing Dial("Zap/25-1", "ZAP/45/571|360") in new stack -- Called 
45/571
-- Zap/45-1 answered Zap/25-1
-- Attempting native bridge of Zap/25-1 and Zap/45-1

The first few lines are what I have always seen go out to the switch, It 
is the last line that is new. I would guess that asterisk is trying to 
take advantage of some feature of the T1 card and this is screwing up 
the outpulsing.

Anyone have any ideas? Can I disable the 'native bridge' feature?

Kirt Stanfield



From voiptandem at shaneyoung.com  Sun Feb  4 10:52:08 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Sun, 04 Feb 2007 10:52:08 -0600
Subject: [VoIP] T1 related Problem
Message-ID: <20070204105208.fobduruf4g8w08ws@mail.shaneyoung.com>

Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:

> Anyone have any ideas? Can I disable the 'native bridge' feature?

The native bridge happens after the answer, so it's not causing any
problems, it is the result of everything else happening before it.

When two calls are IP through asterisk and the far end answers,
Asterisk will normally attempt to re-route the audio between the
devices directly unless it needs to be in the audio path.

I suppose this is the same for the ZAP channels, where it will attempt
to connect the two channels directly through the pseudo-TDM bus.

--Shane





From kxt at fubegra.net  Sun Feb  4 10:54:26 2007
From: kxt at fubegra.net (Russ Price)
Date: Sun, 04 Feb 2007 10:54:26 -0600
Subject: [VoIP] Anyone any ideas??
In-Reply-To: <45C60474.5080507@innismir.net>
References: <E1HDhZ9-0004vw-Gt@telstar.blackratchet.org>	<020b01c74862$519ee070$0a01a8c0@acer1dd0bbc6d0>
	<45C60474.5080507@innismir.net>
Message-ID: <45C60FC2.1020904@fubegra.net>

Ben Jackson wrote:

> I think the current Fedora Core version is 6. A few quick googles shows
> that it does support SATA. The only issue might be if it doesn't support
> your exact SATA controller. However, I would think that wouldn't be the
> case.

Also, CentOS 4.4 supports SATA; I have two systems here running it on 
SATA drives.

	Russ

From ian at uax.org.uk  Sun Feb  4 11:20:05 2007
From: ian at uax.org.uk (Ian Jolly)
Date: Sun, 4 Feb 2007 17:20:05 -0000
Subject: [VoIP] Anyone any ideas??
References: <E1HDhZ9-0004vw-Gt@telstar.blackratchet.org>	<020b01c74862$519ee070$0a01a8c0@acer1dd0bbc6d0><45C60474.5080507@innismir.net>
	<45C60FC2.1020904@fubegra.net>
Message-ID: <026301c74880$b6bf1ba0$0a01a8c0@acer1dd0bbc6d0>

Tnx Russ - I'll try both :-)

Ian

+44 (0)352 82 26 (via a 1929 GPO Rural Automatic eXchange!)
CNET - the Heritage Telephone Network

----- Original Message ----- 
From: "Russ Price" <kxt at fubegra.net>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Sunday, February 04, 2007 4:54 PM
Subject: Re: [VoIP] Anyone any ideas??


> Ben Jackson wrote:
>
>> I think the current Fedora Core version is 6. A few quick googles shows
>> that it does support SATA. The only issue might be if it doesn't support
>> your exact SATA controller. However, I would think that wouldn't be the
>> case.
>
> Also, CentOS 4.4 supports SATA; I have two systems here running it on
> SATA drives.
>
> Russ
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
> --
> This email has been verified as Virus free
> Virus Protection and more available at http://www.plus.net
>
>
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.5.432 / Virus Database: 268.17.24/668 - Release Date: 
> 04/02/2007 01:30
>
> 


From kirtley.stanfield at comcast.net  Sun Feb  4 13:27:03 2007
From: kirtley.stanfield at comcast.net (Kirt Stanfield)
Date: Sun, 04 Feb 2007 14:27:03 -0500
Subject: [VoIP] T1 related Problem
In-Reply-To: <20070204105208.fobduruf4g8w08ws@mail.shaneyoung.com>
References: <20070204105208.fobduruf4g8w08ws@mail.shaneyoung.com>
Message-ID: <45C63387.6030600@comcast.net>

Let me clarify oen thing.

When I was using the TDM 400 card I never saw the message about the 
native bridge,a nd the same thing worked. The problem started when I 
went to T1 and stoppe dusing the tDM 400 (although it is still in the  
machine).

It almost appears to be a timing problem - perhaps asterisk is trying 
the native bridging before it outpulses to the switch.

Kirt

Shane Young wrote:

>Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>
>  
>
>>Anyone have any ideas? Can I disable the 'native bridge' feature?
>>    
>>
>
>The native bridge happens after the answer, so it's not causing any
>problems, it is the result of everything else happening before it.
>
>When two calls are IP through asterisk and the far end answers,
>Asterisk will normally attempt to re-route the audio between the
>devices directly unless it needs to be in the audio path.
>
>I suppose this is the same for the ZAP channels, where it will attempt
>to connect the two channels directly through the pseudo-TDM bus.
>
>--Shane
>
>
>
>
>_______________________________________________
>VoIP mailing list
>VoIP at ckts.info
>http://lists.ckts.info/mailman/listinfo/voip
>Project Web Page: http://www.ckts.info/
>
>
>  
>

From voiptandem at shaneyoung.com  Sun Feb  4 16:50:24 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Sun, 04 Feb 2007 16:50:24 -0600
Subject: [VoIP] T1 related Problem
In-Reply-To: <45C63387.6030600@comcast.net>
References: <20070204105208.fobduruf4g8w08ws@mail.shaneyoung.com>
	<45C63387.6030600@comcast.net>
Message-ID: <20070204165024.evolsg2uecws44kg@mail.shaneyoung.com>

Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:

> Let me clarify oen thing.
>
> When I was using the TDM 400 card I never saw the message about the
> native bridge,a nd the same thing worked. The problem started when I
> went to T1 and stoppe dusing the tDM 400 (although it is still in the
> machine).
>
> It almost appears to be a timing problem - perhaps asterisk is trying
> the native bridging before it outpulses to the switch.

I've never seen it either for a zap channel, however I have features  
enabled which would always prevent it.

Again, it waits until it get's the answer back from the channel before  
it attempts this (as you show in your output) so I don't think the  
native bridge is the problem.

You could try putting in a wait in your dial string before dialing to  
see if that helps.

--Shane



From kirtley.stanfield at comcast.net  Sun Feb  4 19:50:10 2007
From: kirtley.stanfield at comcast.net (Kirt Stanfield)
Date: Sun, 04 Feb 2007 20:50:10 -0500
Subject: [VoIP] T1 related Problem
In-Reply-To: <20070204165024.evolsg2uecws44kg@mail.shaneyoung.com>
References: <20070204105208.fobduruf4g8w08ws@mail.shaneyoung.com>	<45C63387.6030600@comcast.net>
	<20070204165024.evolsg2uecws44kg@mail.shaneyoung.com>
Message-ID: <45C68D52.2020507@comcast.net>

Shane,

I have to think the native bridge must be involved. This problem only 
happens when the call originates and terminates on the T1 bank. Calls 
coming in from either my SIP phone or CNET go out just fine to my switch.

Kirt

Shane Young wrote:

>Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>
>  
>
>>Let me clarify oen thing.
>>
>>When I was using the TDM 400 card I never saw the message about the
>>native bridge,a nd the same thing worked. The problem started when I
>>went to T1 and stoppe dusing the tDM 400 (although it is still in the
>>machine).
>>
>>It almost appears to be a timing problem - perhaps asterisk is trying
>>the native bridging before it outpulses to the switch.
>>    
>>
>
>I've never seen it either for a zap channel, however I have features  
>enabled which would always prevent it.
>
>Again, it waits until it get's the answer back from the channel before  
>it attempts this (as you show in your output) so I don't think the  
>native bridge is the problem.
>
>You could try putting in a wait in your dial string before dialing to  
>see if that helps.
>
>--Shane
>
>
>_______________________________________________
>VoIP mailing list
>VoIP at ckts.info
>http://lists.ckts.info/mailman/listinfo/voip
>Project Web Page: http://www.ckts.info/
>
>
>  
>

From kirtley.stanfield at comcast.net  Sun Feb  4 20:21:21 2007
From: kirtley.stanfield at comcast.net (Kirt Stanfield)
Date: Sun, 04 Feb 2007 21:21:21 -0500
Subject: [VoIP] T1 related Problem
In-Reply-To: <45C68D52.2020507@comcast.net>
References: <20070204105208.fobduruf4g8w08ws@mail.shaneyoung.com>	<45C63387.6030600@comcast.net>	<20070204165024.evolsg2uecws44kg@mail.shaneyoung.com>
	<45C68D52.2020507@comcast.net>
Message-ID: <45C694A1.2040301@comcast.net>

I tried up to a 5 second delay before the start of dialing (using 10 w's 
in the dialstring). No change. It seems that no matter what only the 
last of my 3 digits is getting out to the switch.

Kirt

Kirt Stanfield wrote:

>Shane,
>
>I have to think the native bridge must be involved. This problem only 
>happens when the call originates and terminates on the T1 bank. Calls 
>coming in from either my SIP phone or CNET go out just fine to my switch.
>
>Kirt
>
>Shane Young wrote:
>
>  
>
>>Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>>
>> 
>>
>>    
>>
>>>Let me clarify oen thing.
>>>
>>>When I was using the TDM 400 card I never saw the message about the
>>>native bridge,a nd the same thing worked. The problem started when I
>>>went to T1 and stoppe dusing the tDM 400 (although it is still in the
>>>machine).
>>>
>>>It almost appears to be a timing problem - perhaps asterisk is trying
>>>the native bridging before it outpulses to the switch.
>>>   
>>>
>>>      
>>>
>>I've never seen it either for a zap channel, however I have features  
>>enabled which would always prevent it.
>>
>>Again, it waits until it get's the answer back from the channel before  
>>it attempts this (as you show in your output) so I don't think the  
>>native bridge is the problem.
>>
>>You could try putting in a wait in your dial string before dialing to  
>>see if that helps.
>>
>>--Shane
>>
>>
>>_______________________________________________
>>VoIP mailing list
>>VoIP at ckts.info
>>http://lists.ckts.info/mailman/listinfo/voip
>>Project Web Page: http://www.ckts.info/
>>
>>
>> 
>>
>>    
>>
>_______________________________________________
>VoIP mailing list
>VoIP at ckts.info
>http://lists.ckts.info/mailman/listinfo/voip
>Project Web Page: http://www.ckts.info/
>
>
>  
>

From voiptandem at shaneyoung.com  Sun Feb  4 21:43:54 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Sun, 04 Feb 2007 21:43:54 -0600
Subject: [VoIP] T1 related Problem
In-Reply-To: <45C68D52.2020507@comcast.net>
References: <20070204105208.fobduruf4g8w08ws@mail.shaneyoung.com>
	<45C63387.6030600@comcast.net>
	<20070204165024.evolsg2uecws44kg@mail.shaneyoung.com>
	<45C68D52.2020507@comcast.net>
Message-ID: <20070204214354.ijn9yfe6hww8kw8g@mail.shaneyoung.com>

Well, as I said, the bridge doesn't happen until after it's done  
dialing and has received an answer.

I have to go back and look to see exactly what the problem was, but,  
in the meantime, we should be able to disable the native bridging by  
requesting Asterisk stay in the audio path all the time.

Try adding a "t" to the end of the dial command like this:
Dial(zap/xx||t)






Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:

> Shane,
>
> I have to think the native bridge must be involved. This problem only
> happens when the call originates and terminates on the T1 bank. Calls
> coming in from either my SIP phone or CNET go out just fine to my switch.
>
> Kirt
>
> Shane Young wrote:
>
>> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>>
>>
>>
>>> Let me clarify oen thing.
>>>
>>> When I was using the TDM 400 card I never saw the message about the
>>> native bridge,a nd the same thing worked. The problem started when I
>>> went to T1 and stoppe dusing the tDM 400 (although it is still in the
>>> machine).
>>>
>>> It almost appears to be a timing problem - perhaps asterisk is trying
>>> the native bridging before it outpulses to the switch.
>>>
>>>
>>
>> I've never seen it either for a zap channel, however I have features
>> enabled which would always prevent it.
>>
>> Again, it waits until it get's the answer back from the channel before
>> it attempts this (as you show in your output) so I don't think the
>> native bridge is the problem.
>>
>> You could try putting in a wait in your dial string before dialing to
>> see if that helps.
>>
>> --Shane
>>
>>
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>
>>
>>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>





From voiptandem at shaneyoung.com  Sun Feb  4 21:46:35 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Sun, 04 Feb 2007 21:46:35 -0600
Subject: [VoIP] T1 related Problem
In-Reply-To: <45C6058A.5040401@comcast.net>
References: <45C6058A.5040401@comcast.net>
Message-ID: <20070204214635.hqw9as0xwgoksoco@mail.shaneyoung.com>

What digits should be going to the switch and what digits are making  
it to the switch?

Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:

> -- Goto (internal,571,1)
> -- Executing Macro("Zap/25-1", "dialswitch|571") in new stack
> -- Executing Dial("Zap/25-1", "ZAP/45/571|360") in new stack -- Called
> 45/571
> -- Zap/45-1 answered Zap/25-1
> -- Attempting native bridge of Zap/25-1 and Zap/45-1





From ikj1234i at yahoo.com  Sun Feb  4 22:38:56 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Sun, 4 Feb 2007 20:38:56 -0800 (PST)
Subject: [VoIP] T1 related Problem
In-Reply-To: <45C6058A.5040401@comcast.net>
Message-ID: <461145.84146.qm@web51603.mail.yahoo.com>


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

> ...Can I disable the 'native bridge' feature?

Yes.  For testing purposes of course :)

This small hack will disable native bridging entirely,
there may be other ways to do so but this one goes
right to the source...

In file asterisk/channels/chan_zap.c, add a line 
        return -2;
Add the above 'return' statement into function
zt_bridge(), it should be placed right before this
comment string :
/* For now, don't attempt to native bridge if either
channel needs DTMF detection. There is code below to
handle it properly until DTMF is actually seen, but
due to currently unresolved issues it's ignored... */

Max


 
____________________________________________________________________________________
The fish are biting. 
Get more visitors on your site using Yahoo! Search Marketing.
http://searchmarketing.yahoo.com/arp/sponsoredsearch_v2.php

From lee at spenadel.com  Sun Feb  4 22:41:52 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Sun, 4 Feb 2007 23:41:52 -0500
Subject: [VoIP] T1 related Problem
In-Reply-To: <20070204214354.ijn9yfe6hww8kw8g@mail.shaneyoung.com>
Message-ID: <001901c748df$f5d93c30$0202fea9@Clarabell>

What does that "t" at the end do?  I just cut over from a Linksys ATA to my
Adtran 850.  I couldn't get Asterisk to pulse out properly to the Step
switch until I put that "t" in my statement:

Before: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/ww${EXTEN:4})

After: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/t/ww${EXTEN:4})

It now works!  I've spent the last 2 hours trying to figure out these very
strange happenings:

When there's no battery connected to my FXO channel (any one for that
matter) I can see the pulsing on the FXO led.  The pulsing is correct for
the extension that I dialed.  However, when I connect battery to it, such as
an extension from my Stepper, I get consistent but strange dialing.  No
matter what extension I try to dial in Asterisk, the Stepper sees digit 2
and whatever the last digit of the called extension was.  For example, I
dial 349-8630, the Stepper sees 20.  That's it.  If I dial 349-8692, Step
sees 22.  It would always see the first digit as 2 followed by the last
digit of the called number.  I was pulling my hair out of my head until I
came upstairs and saw this email from Shane.  I was following this thread
between Shane and Kirt as I'm also getting the bridge messages.

  -- Attempting native bridge of Zap/1-1 and Zap/21-1

But it doesn't seem to be a problem.

Thanks for inadvertently helping me solve my problem guys.

Lee

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Shane Young
Sent: Sunday, February 04, 2007 10:44 PM
To: voip at ckts.info
Subject: Re: [VoIP] T1 related Problem

Well, as I said, the bridge doesn't happen until after it's done dialing and
has received an answer.

I have to go back and look to see exactly what the problem was, but, in the
meantime, we should be able to disable the native bridging by requesting
Asterisk stay in the audio path all the time.

Try adding a "t" to the end of the dial command like this:
Dial(zap/xx||t)






Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:

> Shane,
>
> I have to think the native bridge must be involved. This problem only 
> happens when the call originates and terminates on the T1 bank. Calls 
> coming in from either my SIP phone or CNET go out just fine to my switch.
>
> Kirt
>
> Shane Young wrote:
>
>> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>>
>>
>>
>>> Let me clarify oen thing.
>>>
>>> When I was using the TDM 400 card I never saw the message about the 
>>> native bridge,a nd the same thing worked. The problem started when I 
>>> went to T1 and stoppe dusing the tDM 400 (although it is still in 
>>> the machine).
>>>
>>> It almost appears to be a timing problem - perhaps asterisk is 
>>> trying the native bridging before it outpulses to the switch.
>>>
>>>
>>
>> I've never seen it either for a zap channel, however I have features 
>> enabled which would always prevent it.
>>
>> Again, it waits until it get's the answer back from the channel 
>> before it attempts this (as you show in your output) so I don't think 
>> the native bridge is the problem.
>>
>> You could try putting in a wait in your dial string before dialing to 
>> see if that helps.
>>
>> --Shane
>>
>>
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>
>>
>>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>




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


From voiptandem at shaneyoung.com  Mon Feb  5 00:01:53 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Mon, 05 Feb 2007 00:01:53 -0600
Subject: [VoIP] T1 related Problem
In-Reply-To: <001901c748df$f5d93c30$0202fea9@Clarabell>
References: <001901c748df$f5d93c30$0202fea9@Clarabell>
Message-ID: <20070205000153.ph0lh0t2qsg0g4k0@mail.shaneyoung.com>

The "t" that I suggested allows the called party to initiate a  
transfer by hitting the # key on their phone, however you have it in  
the wrong spot :)

The dial string just looks plain wrong to me, not even sure how it works :)

Any chance you could put a butt set on the line and see (hear) what's  
happening different.


Quoting Lee Spenadel <lee at spenadel.com>:

> What does that "t" at the end do?  I just cut over from a Linksys ATA to my
> Adtran 850.  I couldn't get Asterisk to pulse out properly to the Step
> switch until I put that "t" in my statement:
>
> Before: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/ww${EXTEN:4})
>
> After: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/t/ww${EXTEN:4})
>
> It now works!  I've spent the last 2 hours trying to figure out these very
> strange happenings:
>
> When there's no battery connected to my FXO channel (any one for that
> matter) I can see the pulsing on the FXO led.  The pulsing is correct for
> the extension that I dialed.  However, when I connect battery to it, such as
> an extension from my Stepper, I get consistent but strange dialing.  No
> matter what extension I try to dial in Asterisk, the Stepper sees digit 2
> and whatever the last digit of the called extension was.  For example, I
> dial 349-8630, the Stepper sees 20.  That's it.  If I dial 349-8692, Step
> sees 22.  It would always see the first digit as 2 followed by the last
> digit of the called number.  I was pulling my hair out of my head until I
> came upstairs and saw this email from Shane.  I was following this thread
> between Shane and Kirt as I'm also getting the bridge messages.
>
>   -- Attempting native bridge of Zap/1-1 and Zap/21-1
>
> But it doesn't seem to be a problem.
>
> Thanks for inadvertently helping me solve my problem guys.
>
> Lee
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Shane Young
> Sent: Sunday, February 04, 2007 10:44 PM
> To: voip at ckts.info
> Subject: Re: [VoIP] T1 related Problem
>
> Well, as I said, the bridge doesn't happen until after it's done dialing and
> has received an answer.
>
> I have to go back and look to see exactly what the problem was, but, in the
> meantime, we should be able to disable the native bridging by requesting
> Asterisk stay in the audio path all the time.
>
> Try adding a "t" to the end of the dial command like this:
> Dial(zap/xx||t)
>
>
>
>
>
>
> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>
>> Shane,
>>
>> I have to think the native bridge must be involved. This problem only
>> happens when the call originates and terminates on the T1 bank. Calls
>> coming in from either my SIP phone or CNET go out just fine to my switch.
>>
>> Kirt
>>
>> Shane Young wrote:
>>
>>> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>>>
>>>
>>>
>>>> Let me clarify oen thing.
>>>>
>>>> When I was using the TDM 400 card I never saw the message about the
>>>> native bridge,a nd the same thing worked. The problem started when I
>>>> went to T1 and stoppe dusing the tDM 400 (although it is still in
>>>> the machine).
>>>>
>>>> It almost appears to be a timing problem - perhaps asterisk is
>>>> trying the native bridging before it outpulses to the switch.
>>>>
>>>>
>>>
>>> I've never seen it either for a zap channel, however I have features
>>> enabled which would always prevent it.
>>>
>>> Again, it waits until it get's the answer back from the channel
>>> before it attempts this (as you show in your output) so I don't think
>>> the native bridge is the problem.
>>>
>>> You could try putting in a wait in your dial string before dialing to
>>> see if that helps.
>>>
>>> --Shane
>>>
>>>
>>> _______________________________________________
>>> VoIP mailing list
>>> VoIP at ckts.info
>>> http://lists.ckts.info/mailman/listinfo/voip
>>> Project Web Page: http://www.ckts.info/
>>>
>>>
>>>
>>>
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>
>
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>





From hockd at dteenergy.com  Mon Feb  5 05:08:23 2007
From: hockd at dteenergy.com (Dennis D Hock)
Date: Mon, 5 Feb 2007 06:08:23 -0500
Subject: [VoIP] +64 Country Code
Message-ID: <OF0164FB9D.1E00D25F-ON85257279.003D313F-85257279.003D3145@dteenergy.com>


Ian,

Congrats on etablishing that link with John.  Sounds like a new country
code will be added very soon.

Dennis Hock

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


To: "CNET VoIP" <voip at ckts.info>
From: "Ian Jolly" <ian at uax.org.uk>
Sent by: voip-bounces at ckts.info
Date: 02/04/2007 08:07AM
Subject: [VoIP] +64 Country Code

An update

John in Auckland on the north island of New Zealand has been busy over the
last week playing with an IAX softphone (DIAX) set up on his PC and hosted
off my Asterisk box in the UK.

I've seen him trying different recorded services on switches in the UK and
the 'colonies' :-)  I've hear my old EM switch burst into life at odd hours
and when I've checked it has been John in NZ.

Last night I had my first telephone conversation with John.  It seemed odd
as I was chatting to him on Saturday evening whilst it was early Sunday
afternoon with him - they are 13 hours ahead of the UK with their daylight
saving time.

The conversation dropped out once. But once re-established lasted for a
couple of hours. Latency made the call a bit difficult - almost needed to
use 'over' as in two-way radio!.  Little bit of echo about two seconds
later but at quite a low level.

John is now going to get an ATA so that he can be permanently on without
the need for leaving his PC on.

He has a Strowger PABX at home but it is not up and running. It sounds like
a GPO type PABX No 1 as we knew them in the UK.

John started his working life with New Zealand Post Office Telephones and
still works for one of their descendants. He is also involved at a
preserved steam railway where he has a former NZ PO Telephones UAX13
strowger switch similar to those we have on CNET in the UK. Not only did he
acquire the switch but he got the complete building as well !!  The
exchange needs cabling - any volunteers ??  :-)

He also thinks that some other telephone folk in NZ will be interested in
connecting to CNET. - maybe an NZ /411 page one day ? :-)

Ian Jolly

+44 (0)352 82 26 (via a 1929 GPO Rural Automatic eXchange!)
CNET - the Heritage Telephone Network
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VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voipProject Web Page:
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From lee at spenadel.com  Mon Feb  5 06:34:51 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Mon, 5 Feb 2007 07:34:51 -0500
Subject: [VoIP] T1 related Problem
In-Reply-To: <20070205000153.ph0lh0t2qsg0g4k0@mail.shaneyoung.com>
Message-ID: <002c01c74922$089cd4c0$0202fea9@Clarabell>

It may be wrong, but it works just fine :P 

I don't see why this shouldn't work.

Yes, I did put a butt set on the line and listened.  It sounded just like
what was being pulsed, the number 2 + the last digit dialed. 

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Shane Young
Sent: Monday, February 05, 2007 1:02 AM
To: voip at ckts.info
Subject: Re: [VoIP] T1 related Problem

The "t" that I suggested allows the called party to initiate a transfer by
hitting the # key on their phone, however you have it in the wrong spot :)

The dial string just looks plain wrong to me, not even sure how it works :)

Any chance you could put a butt set on the line and see (hear) what's
happening different.


Quoting Lee Spenadel <lee at spenadel.com>:

> What does that "t" at the end do?  I just cut over from a Linksys ATA 
> to my Adtran 850.  I couldn't get Asterisk to pulse out properly to 
> the Step switch until I put that "t" in my statement:
>
> Before: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/ww${EXTEN:4})
>
> After: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/t/ww${EXTEN:4})
>
> It now works!  I've spent the last 2 hours trying to figure out these 
> very strange happenings:
>
> When there's no battery connected to my FXO channel (any one for that
> matter) I can see the pulsing on the FXO led.  The pulsing is correct 
> for the extension that I dialed.  However, when I connect battery to 
> it, such as an extension from my Stepper, I get consistent but strange 
> dialing.  No matter what extension I try to dial in Asterisk, the 
> Stepper sees digit 2 and whatever the last digit of the called 
> extension was.  For example, I dial 349-8630, the Stepper sees 20.  
> That's it.  If I dial 349-8692, Step sees 22.  It would always see the 
> first digit as 2 followed by the last digit of the called number.  I 
> was pulling my hair out of my head until I came upstairs and saw this 
> email from Shane.  I was following this thread between Shane and Kirt as
I'm also getting the bridge messages.
>
>   -- Attempting native bridge of Zap/1-1 and Zap/21-1
>
> But it doesn't seem to be a problem.
>
> Thanks for inadvertently helping me solve my problem guys.
>
> Lee
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf 
> Of Shane Young
> Sent: Sunday, February 04, 2007 10:44 PM
> To: voip at ckts.info
> Subject: Re: [VoIP] T1 related Problem
>
> Well, as I said, the bridge doesn't happen until after it's done 
> dialing and has received an answer.
>
> I have to go back and look to see exactly what the problem was, but, 
> in the meantime, we should be able to disable the native bridging by 
> requesting Asterisk stay in the audio path all the time.
>
> Try adding a "t" to the end of the dial command like this:
> Dial(zap/xx||t)
>
>
>
>
>
>
> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>
>> Shane,
>>
>> I have to think the native bridge must be involved. This problem only 
>> happens when the call originates and terminates on the T1 bank. Calls 
>> coming in from either my SIP phone or CNET go out just fine to my switch.
>>
>> Kirt
>>
>> Shane Young wrote:
>>
>>> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>>>
>>>
>>>
>>>> Let me clarify oen thing.
>>>>
>>>> When I was using the TDM 400 card I never saw the message about the 
>>>> native bridge,a nd the same thing worked. The problem started when 
>>>> I went to T1 and stoppe dusing the tDM 400 (although it is still in 
>>>> the machine).
>>>>
>>>> It almost appears to be a timing problem - perhaps asterisk is 
>>>> trying the native bridging before it outpulses to the switch.
>>>>
>>>>
>>>
>>> I've never seen it either for a zap channel, however I have features 
>>> enabled which would always prevent it.
>>>
>>> Again, it waits until it get's the answer back from the channel 
>>> before it attempts this (as you show in your output) so I don't 
>>> think the native bridge is the problem.
>>>
>>> You could try putting in a wait in your dial string before dialing 
>>> to see if that helps.
>>>
>>> --Shane
>>>
>>>
>>> _______________________________________________
>>> VoIP mailing list
>>> VoIP at ckts.info
>>> http://lists.ckts.info/mailman/listinfo/voip
>>> Project Web Page: http://www.ckts.info/
>>>
>>>
>>>
>>>
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>
>
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>




_______________________________________________
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Project Web Page: http://www.ckts.info/


From kirtley.stanfield at comcast.net  Mon Feb  5 06:40:48 2007
From: kirtley.stanfield at comcast.net (Kirt Stanfield)
Date: Mon, 05 Feb 2007 07:40:48 -0500
Subject: [VoIP] T1 related Problem
In-Reply-To: <20070204214635.hqw9as0xwgoksoco@mail.shaneyoung.com>
References: <45C6058A.5040401@comcast.net>
	<20070204214635.hqw9as0xwgoksoco@mail.shaneyoung.com>
Message-ID: <45C725D0.7080603@comcast.net>

Well if i am calling 554 then the digitd 5 5 4 should be going out.  All 
that seems to be getting there is maybe part of the second 5 and all of 
the 4.

Kirt

Shane Young wrote:

>What digits should be going to the switch and what digits are making  
>it to the switch?
>
>Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>
>  
>
>>-- Goto (internal,571,1)
>>-- Executing Macro("Zap/25-1", "dialswitch|571") in new stack
>>-- Executing Dial("Zap/25-1", "ZAP/45/571|360") in new stack -- Called
>>45/571
>>-- Zap/45-1 answered Zap/25-1
>>-- Attempting native bridge of Zap/25-1 and Zap/45-1
>>    
>>
>
>
>
>
>_______________________________________________
>VoIP mailing list
>VoIP at ckts.info
>http://lists.ckts.info/mailman/listinfo/voip
>Project Web Page: http://www.ckts.info/
>
>
>  
>

From kirtley.stanfield at comcast.net  Mon Feb  5 07:02:29 2007
From: kirtley.stanfield at comcast.net (Kirt Stanfield)
Date: Mon, 05 Feb 2007 08:02:29 -0500
Subject: [VoIP] T1 related Problem
In-Reply-To: <461145.84146.qm@web51603.mail.yahoo.com>
References: <461145.84146.qm@web51603.mail.yahoo.com>
Message-ID: <45C72AE5.2010805@comcast.net>

Max,

Your 'hack' fixed the problem the T1 problem. I can now call 
successfully between two extensions on the same T1.

You made a comment about DTMF not working correctly - I seem to be able 
to call out with DTMF OK with the hack in place. Did you mean something 
like # sign detection or the like?

Obviously in the long term we want bridging - I suspect it takes a lot 
of load of the CPU, so a proper fix is needed. I will leave the hack in 
palce for a few days to see if I find other tings broken.

Thanks,
Kirt

ikjtel wrote:

>--- Kirt Stanfield <kirtley.stanfield at comcast.net>
>wrote:
>
>  
>
>>...Can I disable the 'native bridge' feature?
>>    
>>
>
>Yes.  For testing purposes of course :)
>
>This small hack will disable native bridging entirely,
>there may be other ways to do so but this one goes
>right to the source...
>
>In file asterisk/channels/chan_zap.c, add a line 
>        return -2;
>Add the above 'return' statement into function
>zt_bridge(), it should be placed right before this
>comment string :
>/* For now, don't attempt to native bridge if either
>channel needs DTMF detection. There is code below to
>handle it properly until DTMF is actually seen, but
>due to currently unresolved issues it's ignored... */
>
>Max
>
>
> 
>____________________________________________________________________________________
>The fish are biting. 
>Get more visitors on your site using Yahoo! Search Marketing.
>http://searchmarketing.yahoo.com/arp/sponsoredsearch_v2.php
>
>
>  
>

From voiptandem at shaneyoung.com  Mon Feb  5 07:33:01 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Mon, 05 Feb 2007 07:33:01 -0600
Subject: [VoIP] T1 related Problem
In-Reply-To: <002c01c74922$089cd4c0$0202fea9@Clarabell>
References: <002c01c74922$089cd4c0$0202fea9@Clarabell>
Message-ID: <20070205073301.m7f56olcg848cw4s@mail.shaneyoung.com>

Quoting Lee Spenadel <lee at spenadel.com>:

> It may be wrong, but it works just fine :P
>
> I don't see why this shouldn't work.

Cool, maybe the t is "working" a different way :)

To do what I described, the dial statement should look like this:
_${OFFICECODE}863X,4,Dial(ZAP/21/ww${EXTEN:4}||t)

When the called phone answeres, can they hit # and get a prompt back  
from Asterisk? If so, then it's working as I described and is some  
hidden feature that allows you to use it where you put it rather than  
where the documentation states to put it (*CLI> show application dial)



> Yes, I did put a butt set on the line and listened.  It sounded just like
> what was being pulsed, the number 2 + the last digit dialed.
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Shane Young
> Sent: Monday, February 05, 2007 1:02 AM
> To: voip at ckts.info
> Subject: Re: [VoIP] T1 related Problem
>
> The "t" that I suggested allows the called party to initiate a transfer by
> hitting the # key on their phone, however you have it in the wrong spot :)
>
> The dial string just looks plain wrong to me, not even sure how it works :)
>
> Any chance you could put a butt set on the line and see (hear) what's
> happening different.
>
>
> Quoting Lee Spenadel <lee at spenadel.com>:
>
>> What does that "t" at the end do?  I just cut over from a Linksys ATA
>> to my Adtran 850.  I couldn't get Asterisk to pulse out properly to
>> the Step switch until I put that "t" in my statement:
>>
>> Before: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/ww${EXTEN:4})
>>
>> After: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/t/ww${EXTEN:4})
>>
>> It now works!  I've spent the last 2 hours trying to figure out these
>> very strange happenings:
>>
>> When there's no battery connected to my FXO channel (any one for that
>> matter) I can see the pulsing on the FXO led.  The pulsing is correct
>> for the extension that I dialed.  However, when I connect battery to
>> it, such as an extension from my Stepper, I get consistent but strange
>> dialing.  No matter what extension I try to dial in Asterisk, the
>> Stepper sees digit 2 and whatever the last digit of the called
>> extension was.  For example, I dial 349-8630, the Stepper sees 20.
>> That's it.  If I dial 349-8692, Step sees 22.  It would always see the
>> first digit as 2 followed by the last digit of the called number.  I
>> was pulling my hair out of my head until I came upstairs and saw this
>> email from Shane.  I was following this thread between Shane and Kirt as
> I'm also getting the bridge messages.
>>
>>   -- Attempting native bridge of Zap/1-1 and Zap/21-1
>>
>> But it doesn't seem to be a problem.
>>
>> Thanks for inadvertently helping me solve my problem guys.
>>
>> Lee
>>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf
>> Of Shane Young
>> Sent: Sunday, February 04, 2007 10:44 PM
>> To: voip at ckts.info
>> Subject: Re: [VoIP] T1 related Problem
>>
>> Well, as I said, the bridge doesn't happen until after it's done
>> dialing and has received an answer.
>>
>> I have to go back and look to see exactly what the problem was, but,
>> in the meantime, we should be able to disable the native bridging by
>> requesting Asterisk stay in the audio path all the time.
>>
>> Try adding a "t" to the end of the dial command like this:
>> Dial(zap/xx||t)
>>
>>
>>
>>
>>
>>
>> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>>
>>> Shane,
>>>
>>> I have to think the native bridge must be involved. This problem only
>>> happens when the call originates and terminates on the T1 bank. Calls
>>> coming in from either my SIP phone or CNET go out just fine to my switch.
>>>
>>> Kirt
>>>
>>> Shane Young wrote:
>>>
>>>> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>>>>
>>>>
>>>>
>>>>> Let me clarify oen thing.
>>>>>
>>>>> When I was using the TDM 400 card I never saw the message about the
>>>>> native bridge,a nd the same thing worked. The problem started when
>>>>> I went to T1 and stoppe dusing the tDM 400 (although it is still in
>>>>> the machine).
>>>>>
>>>>> It almost appears to be a timing problem - perhaps asterisk is
>>>>> trying the native bridging before it outpulses to the switch.
>>>>>
>>>>>
>>>>
>>>> I've never seen it either for a zap channel, however I have features
>>>> enabled which would always prevent it.
>>>>
>>>> Again, it waits until it get's the answer back from the channel
>>>> before it attempts this (as you show in your output) so I don't
>>>> think the native bridge is the problem.
>>>>
>>>> You could try putting in a wait in your dial string before dialing
>>>> to see if that helps.
>>>>
>>>> --Shane




From lee at spenadel.com  Mon Feb  5 07:37:54 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Mon, 5 Feb 2007 08:37:54 -0500
Subject: [VoIP] T1 related Problem
In-Reply-To: <20070205073301.m7f56olcg848cw4s@mail.shaneyoung.com>
Message-ID: <004801c7492a$d7d43050$0202fea9@Clarabell>

I removed my "t" again this AM.  It broke dialing to the step switch.  Put
it back and it worked again.   

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Shane Young
Sent: Monday, February 05, 2007 8:33 AM
To: voip at ckts.info
Subject: Re: [VoIP] T1 related Problem

Quoting Lee Spenadel <lee at spenadel.com>:

> It may be wrong, but it works just fine :P
>
> I don't see why this shouldn't work.

Cool, maybe the t is "working" a different way :)

To do what I described, the dial statement should look like this:
_${OFFICECODE}863X,4,Dial(ZAP/21/ww${EXTEN:4}||t)

When the called phone answeres, can they hit # and get a prompt back from
Asterisk? If so, then it's working as I described and is some hidden feature
that allows you to use it where you put it rather than where the
documentation states to put it (*CLI> show application dial)



> Yes, I did put a butt set on the line and listened.  It sounded just 
> like what was being pulsed, the number 2 + the last digit dialed.
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf 
> Of Shane Young
> Sent: Monday, February 05, 2007 1:02 AM
> To: voip at ckts.info
> Subject: Re: [VoIP] T1 related Problem
>
> The "t" that I suggested allows the called party to initiate a 
> transfer by hitting the # key on their phone, however you have it in 
> the wrong spot :)
>
> The dial string just looks plain wrong to me, not even sure how it 
> works :)
>
> Any chance you could put a butt set on the line and see (hear) what's 
> happening different.
>
>
> Quoting Lee Spenadel <lee at spenadel.com>:
>
>> What does that "t" at the end do?  I just cut over from a Linksys ATA 
>> to my Adtran 850.  I couldn't get Asterisk to pulse out properly to 
>> the Step switch until I put that "t" in my statement:
>>
>> Before: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/ww${EXTEN:4})
>>
>> After: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/t/ww${EXTEN:4})
>>
>> It now works!  I've spent the last 2 hours trying to figure out these 
>> very strange happenings:
>>
>> When there's no battery connected to my FXO channel (any one for that
>> matter) I can see the pulsing on the FXO led.  The pulsing is correct 
>> for the extension that I dialed.  However, when I connect battery to 
>> it, such as an extension from my Stepper, I get consistent but 
>> strange dialing.  No matter what extension I try to dial in Asterisk, 
>> the Stepper sees digit 2 and whatever the last digit of the called 
>> extension was.  For example, I dial 349-8630, the Stepper sees 20.
>> That's it.  If I dial 349-8692, Step sees 22.  It would always see 
>> the first digit as 2 followed by the last digit of the called number.  
>> I was pulling my hair out of my head until I came upstairs and saw 
>> this email from Shane.  I was following this thread between Shane and 
>> Kirt as
> I'm also getting the bridge messages.
>>
>>   -- Attempting native bridge of Zap/1-1 and Zap/21-1
>>
>> But it doesn't seem to be a problem.
>>
>> Thanks for inadvertently helping me solve my problem guys.
>>
>> Lee
>>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On 
>> Behalf Of Shane Young
>> Sent: Sunday, February 04, 2007 10:44 PM
>> To: voip at ckts.info
>> Subject: Re: [VoIP] T1 related Problem
>>
>> Well, as I said, the bridge doesn't happen until after it's done 
>> dialing and has received an answer.
>>
>> I have to go back and look to see exactly what the problem was, but, 
>> in the meantime, we should be able to disable the native bridging by 
>> requesting Asterisk stay in the audio path all the time.
>>
>> Try adding a "t" to the end of the dial command like this:
>> Dial(zap/xx||t)
>>
>>
>>
>>
>>
>>
>> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>>
>>> Shane,
>>>
>>> I have to think the native bridge must be involved. This problem 
>>> only happens when the call originates and terminates on the T1 bank. 
>>> Calls coming in from either my SIP phone or CNET go out just fine to my
switch.
>>>
>>> Kirt
>>>
>>> Shane Young wrote:
>>>
>>>> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>>>>
>>>>
>>>>
>>>>> Let me clarify oen thing.
>>>>>
>>>>> When I was using the TDM 400 card I never saw the message about 
>>>>> the native bridge,a nd the same thing worked. The problem started 
>>>>> when I went to T1 and stoppe dusing the tDM 400 (although it is 
>>>>> still in the machine).
>>>>>
>>>>> It almost appears to be a timing problem - perhaps asterisk is 
>>>>> trying the native bridging before it outpulses to the switch.
>>>>>
>>>>>
>>>>
>>>> I've never seen it either for a zap channel, however I have 
>>>> features enabled which would always prevent it.
>>>>
>>>> Again, it waits until it get's the answer back from the channel 
>>>> before it attempts this (as you show in your output) so I don't 
>>>> think the native bridge is the problem.
>>>>
>>>> You could try putting in a wait in your dial string before dialing 
>>>> to see if that helps.
>>>>
>>>> --Shane



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


From voiptandem at shaneyoung.com  Mon Feb  5 07:41:12 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Mon, 05 Feb 2007 07:41:12 -0600
Subject: [VoIP] T1 related Problem
In-Reply-To: <004801c7492a$d7d43050$0202fea9@Clarabell>
References: <004801c7492a$d7d43050$0202fea9@Clarabell>
Message-ID: <20070205074112.i95pn3g64g0g80cg@mail.shaneyoung.com>

I got that.

What I'm wondering is *what* is it doing to make it work :)

If you have the "t" there, can you hit # on the called phone and get  
something out of asterisk?  If not, then I want to dig deeper to see  
what the "t" does when it's in the position you put it.


Quoting Lee Spenadel <lee at spenadel.com>:

> I removed my "t" again this AM.  It broke dialing to the step switch.  Put
> it back and it worked again.
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Shane Young
> Sent: Monday, February 05, 2007 8:33 AM
> To: voip at ckts.info
> Subject: Re: [VoIP] T1 related Problem
>
> Quoting Lee Spenadel <lee at spenadel.com>:
>
>> It may be wrong, but it works just fine :P
>>
>> I don't see why this shouldn't work.
>
> Cool, maybe the t is "working" a different way :)
>
> To do what I described, the dial statement should look like this:
> _${OFFICECODE}863X,4,Dial(ZAP/21/ww${EXTEN:4}||t)
>
> When the called phone answeres, can they hit # and get a prompt back from
> Asterisk? If so, then it's working as I described and is some hidden feature
> that allows you to use it where you put it rather than where the
> documentation states to put it (*CLI> show application dial)
>
>
>
>> Yes, I did put a butt set on the line and listened.  It sounded just
>> like what was being pulsed, the number 2 + the last digit dialed.
>>
>> -----Original Message-----
>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf
>> Of Shane Young
>> Sent: Monday, February 05, 2007 1:02 AM
>> To: voip at ckts.info
>> Subject: Re: [VoIP] T1 related Problem
>>
>> The "t" that I suggested allows the called party to initiate a
>> transfer by hitting the # key on their phone, however you have it in
>> the wrong spot :)
>>
>> The dial string just looks plain wrong to me, not even sure how it
>> works :)
>>
>> Any chance you could put a butt set on the line and see (hear) what's
>> happening different.
>>
>>
>> Quoting Lee Spenadel <lee at spenadel.com>:
>>
>>> What does that "t" at the end do?  I just cut over from a Linksys ATA
>>> to my Adtran 850.  I couldn't get Asterisk to pulse out properly to
>>> the Step switch until I put that "t" in my statement:
>>>
>>> Before: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/ww${EXTEN:4})
>>>
>>> After: exten => _${OFFICECODE}863X,4,Dial(ZAP/21/t/ww${EXTEN:4})
>>>
>>> It now works!  I've spent the last 2 hours trying to figure out these
>>> very strange happenings:
>>>
>>> When there's no battery connected to my FXO channel (any one for that
>>> matter) I can see the pulsing on the FXO led.  The pulsing is correct
>>> for the extension that I dialed.  However, when I connect battery to
>>> it, such as an extension from my Stepper, I get consistent but
>>> strange dialing.  No matter what extension I try to dial in Asterisk,
>>> the Stepper sees digit 2 and whatever the last digit of the called
>>> extension was.  For example, I dial 349-8630, the Stepper sees 20.
>>> That's it.  If I dial 349-8692, Step sees 22.  It would always see
>>> the first digit as 2 followed by the last digit of the called number.
>>> I was pulling my hair out of my head until I came upstairs and saw
>>> this email from Shane.  I was following this thread between Shane and
>>> Kirt as
>> I'm also getting the bridge messages.
>>>
>>>   -- Attempting native bridge of Zap/1-1 and Zap/21-1
>>>
>>> But it doesn't seem to be a problem.
>>>
>>> Thanks for inadvertently helping me solve my problem guys.
>>>
>>> Lee
>>>
>>> -----Original Message-----
>>> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On
>>> Behalf Of Shane Young
>>> Sent: Sunday, February 04, 2007 10:44 PM
>>> To: voip at ckts.info
>>> Subject: Re: [VoIP] T1 related Problem
>>>
>>> Well, as I said, the bridge doesn't happen until after it's done
>>> dialing and has received an answer.
>>>
>>> I have to go back and look to see exactly what the problem was, but,
>>> in the meantime, we should be able to disable the native bridging by
>>> requesting Asterisk stay in the audio path all the time.
>>>
>>> Try adding a "t" to the end of the dial command like this:
>>> Dial(zap/xx||t)
>>>
>>>
>>>
>>>
>>>
>>>
>>> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>>>
>>>> Shane,
>>>>
>>>> I have to think the native bridge must be involved. This problem
>>>> only happens when the call originates and terminates on the T1 bank.
>>>> Calls coming in from either my SIP phone or CNET go out just fine to my
> switch.
>>>>
>>>> Kirt
>>>>
>>>> Shane Young wrote:
>>>>
>>>>> Quoting Kirt Stanfield <kirtley.stanfield at comcast.net>:
>>>>>
>>>>>
>>>>>
>>>>>> Let me clarify oen thing.
>>>>>>
>>>>>> When I was using the TDM 400 card I never saw the message about
>>>>>> the native bridge,a nd the same thing worked. The problem started
>>>>>> when I went to T1 and stoppe dusing the tDM 400 (although it is
>>>>>> still in the machine).
>>>>>>
>>>>>> It almost appears to be a timing problem - perhaps asterisk is
>>>>>> trying the native bridging before it outpulses to the switch.
>>>>>>
>>>>>>
>>>>>
>>>>> I've never seen it either for a zap channel, however I have
>>>>> features enabled which would always prevent it.
>>>>>
>>>>> Again, it waits until it get's the answer back from the channel
>>>>> before it attempts this (as you show in your output) so I don't
>>>>> think the native bridge is the problem.
>>>>>
>>>>> You could try putting in a wait in your dial string before dialing
>>>>> to see if that helps.
>>>>>
>>>>> --Shane
>
>
>
> _______________________________________________
> 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 ikj1234i at yahoo.com  Mon Feb  5 09:56:34 2007
From: ikj1234i at yahoo.com (ikjtel)
Date: Mon, 5 Feb 2007 07:56:34 -0800 (PST)
Subject: [VoIP] T1 related Problem
In-Reply-To: <001b01c748e0$35ac0a90$0202fea9@Clarabell>
Message-ID: <20070205155634.26720.qmail@web51611.mail.yahoo.com>


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

>  Why would one want to disable this feature as it
> appears to be needed to
> bridge the two channels together.

Native bridging (apologies, I may not have this 100%
right) is not so much a needed feature as it is an
optimization.  When it kicks in, the voice packets can
be directly routed through the zaptel driver (in this
case) rather than being routed through * in "user
space".  The result is lower system overhead (but
shouldn't cause any change in functionality).

In this case we wanted to try to disable it for
testing purposes, to see if it would improve the
situation.  It shouldn't have made a difference, and
the fact that it did suggests (to me) that there may
be a bug in there somewhere that should be made known
to the * folks (if it isn't already)...

Max


 
____________________________________________________________________________________
Do you Yahoo!?
Everyone is raving about the all-new Yahoo! Mail beta.
http://new.mail.yahoo.com

From greg at vyger.net  Wed Feb  7 18:26:37 2007
From: greg at vyger.net (Greg Blakely)
Date: Wed, 7 Feb 2007 18:26:37 -0600
Subject: [VoIP] VT1005 on Asterisk
Message-ID: <A691C3685A152E4AAC9608B8D8E11316686190@godzilla.hickory.vyger.net>

Do you have the bticonfig executable file?  I received my VT1005 and
would like to give it a crack. 

> 
> 
> Here is the batch file to generate the VT1005 config file.  
> Note that the 12 hexadecimal (0-9 and A-F) characters 
> represnt the MAC address of the specific VT1005.  You need to 
> change that to match you MAC address.
> 
> jjbticonfig.bat
> 
> bticonfig vt1005_001225d62628.txt -o
> c:\tftp-root\motvt1000_001225d62628.bin 
> 
> This reads a text file called bticonfig
> vt1005_001225d62628.txt  in the current directory, compiles 
> it, and places a file named motvt1000_001225d62628.bin in the 
> home directory of a TFTP server  (C:\tftp\) on my system.
> 
> 
> 
> The text file (vt1005_001225d62628.txt) contents are as follows:
> 
[snip]


From jjones3601 at yahoo.com  Wed Feb  7 18:44:38 2007
From: jjones3601 at yahoo.com (john jones)
Date: Wed, 7 Feb 2007 16:44:38 -0800 (PST)
Subject: [VoIP] VT1005 on Asterisk
In-Reply-To: <A691C3685A152E4AAC9608B8D8E11316686190@godzilla.hickory.vyger.net>
Message-ID: <20070208004438.1804.qmail@web34304.mail.mud.yahoo.com>

Here you go.  I also attached a "template" that should
work, although the dial plan stuff that Doug worked
out is not in here.

John



--- Greg Blakely <greg at vyger.net> wrote:

> Do you have the bticonfig executable file?  I
> received my VT1005 and
> would like to give it a crack. 
> 
> > 
> > 
> > Here is the batch file to generate the VT1005
> config file.  
> > Note that the 12 hexadecimal (0-9 and A-F)
> characters 
> > represnt the MAC address of the specific VT1005. 
> You need to 
> > change that to match you MAC address.
> > 
> > jjbticonfig.bat
> > 
> > bticonfig vt1005_001225d62628.txt -o
> > c:\tftp-root\motvt1000_001225d62628.bin 
> > 
> > This reads a text file called bticonfig
> > vt1005_001225d62628.txt  in the current directory,
> compiles 
> > it, and places a file named
> motvt1000_001225d62628.bin in the 
> > home directory of a TFTP server  (C:\tftp\) on my
> system.
> > 
> > 
> > 
> > The text file (vt1005_001225d62628.txt) contents
> are as follows:
> > 
> [snip]
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
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From lee at spenadel.com  Wed Feb  7 18:55:10 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Wed, 7 Feb 2007 19:55:10 -0500
Subject: [VoIP] VT1005 on Asterisk
In-Reply-To: <20070208004438.1804.qmail@web34304.mail.mud.yahoo.com>
Message-ID: <010701c74b1b$cc1a62f0$0ac94da6@Clarabell>

Can someone say T1?  I'm a convert, thanks to John Novack.  Grrrrrrrrr. 

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
john jones
Sent: Wednesday, February 07, 2007 7:45 PM
To: Voice Over IP Tandem for Analog Switches
Subject: Re: [VoIP] VT1005 on Asterisk

Here you go.  I also attached a "template" that should work, although the
dial plan stuff that Doug worked out is not in here.

John



--- Greg Blakely <greg at vyger.net> wrote:

> Do you have the bticonfig executable file?  I received my VT1005 and 
> would like to give it a crack.
> 
> > 
> > 
> > Here is the batch file to generate the VT1005
> config file.  
> > Note that the 12 hexadecimal (0-9 and A-F)
> characters
> > represnt the MAC address of the specific VT1005. 
> You need to
> > change that to match you MAC address.
> > 
> > jjbticonfig.bat
> > 
> > bticonfig vt1005_001225d62628.txt -o 
> > c:\tftp-root\motvt1000_001225d62628.bin
> > 
> > This reads a text file called bticonfig vt1005_001225d62628.txt  in 
> > the current directory,
> compiles
> > it, and places a file named
> motvt1000_001225d62628.bin in the
> > home directory of a TFTP server  (C:\tftp\) on my
> system.
> > 
> > 
> > 
> > The text file (vt1005_001225d62628.txt) contents
> are as follows:
> > 
> [snip]
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From greg at vyger.net  Wed Feb  7 21:04:40 2007
From: greg at vyger.net (Greg Blakely)
Date: Wed, 7 Feb 2007 21:04:40 -0600
Subject: [VoIP] VT1005 on Asterisk
Message-ID: <A691C3685A152E4AAC9608B8D8E11316686194@godzilla.hickory.vyger.net>

John,

Could you send it to me off-list?  The executable was replaced by a text
file somewhere along the line.

Try kb0tdf at yahoo.com.

Greg 

> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] 
> On Behalf Of john jones
> Sent: Wednesday, February 07, 2007 6:45 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] VT1005 on Asterisk
> 
> Here you go.  I also attached a "template" that should work, 
> although the dial plan stuff that Doug worked out is not in here.
> 
> John
> 
> 
> 
> --- Greg Blakely <greg at vyger.net> wrote:
> 
> > Do you have the bticonfig executable file?  I received my 
> VT1005 and 
> > would like to give it a crack.
> > 
> > > 
> > > 
> > > Here is the batch file to generate the VT1005
> > config file.  
> > > Note that the 12 hexadecimal (0-9 and A-F)
> > characters
> > > represnt the MAC address of the specific VT1005. 
> > You need to
> > > change that to match you MAC address.
> > > 
> > > jjbticonfig.bat
> > > 
> > > bticonfig vt1005_001225d62628.txt -o 
> > > c:\tftp-root\motvt1000_001225d62628.bin
> > > 
> > > This reads a text file called bticonfig 
> vt1005_001225d62628.txt  in 
> > > the current directory,
> > compiles
> > > it, and places a file named
> > motvt1000_001225d62628.bin in the
> > > home directory of a TFTP server  (C:\tftp\) on my
> > system.
> > > 
> > > 
> > > 
> > > The text file (vt1005_001225d62628.txt) contents
> > are as follows:
> > > 
> > [snip]
> > 
> > _______________________________________________
> > 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 Feb  7 21:13:30 2007
From: jjones3601 at yahoo.com (john jones)
Date: Wed, 7 Feb 2007 19:13:30 -0800 (PST)
Subject: [VoIP] VT1005 on Asterisk
In-Reply-To: <A691C3685A152E4AAC9608B8D8E11316686194@godzilla.hickory.vyger.net>
Message-ID: <283101.20130.qm@web34312.mail.mud.yahoo.com>

Hope this helps.

John

--- Greg Blakely <greg at vyger.net> wrote:

> John,
> 
> Could you send it to me off-list?  The executable
> was replaced by a text
> file somewhere along the line.
> 
> Try kb0tdf at yahoo.com.
> 
> Greg 
> 
> > -----Original Message-----
> > From: voip-bounces at ckts.info
> [mailto:voip-bounces at ckts.info] 
> > On Behalf Of john jones
> > Sent: Wednesday, February 07, 2007 6:45 PM
> > To: Voice Over IP Tandem for Analog Switches
> > Subject: Re: [VoIP] VT1005 on Asterisk
> > 
> > Here you go.  I also attached a "template" that
> should work, 
> > although the dial plan stuff that Doug worked out
> is not in here.
> > 
> > John
> > 
> > 
> > 
> > --- Greg Blakely <greg at vyger.net> wrote:
> > 
> > > Do you have the bticonfig executable file?  I
> received my 
> > VT1005 and 
> > > would like to give it a crack.
> > > 
> > > > 
> > > > 
> > > > Here is the batch file to generate the VT1005
> > > config file.  
> > > > Note that the 12 hexadecimal (0-9 and A-F)
> > > characters
> > > > represnt the MAC address of the specific
> VT1005. 
> > > You need to
> > > > change that to match you MAC address.
> > > > 
> > > > jjbticonfig.bat
> > > > 
> > > > bticonfig vt1005_001225d62628.txt -o 
> > > > c:\tftp-root\motvt1000_001225d62628.bin
> > > > 
> > > > This reads a text file called bticonfig 
> > vt1005_001225d62628.txt  in 
> > > > the current directory,
> > > compiles
> > > > it, and places a file named
> > > motvt1000_001225d62628.bin in the
> > > > home directory of a TFTP server  (C:\tftp\) on
> my
> > > system.
> > > > 
> > > > 
> > > > 
> > > > The text file (vt1005_001225d62628.txt)
> contents
> > > are as follows:
> > > > 
> > > [snip]
> > > 
> > > _______________________________________________
> > > 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/
> 
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From voiptandem at shaneyoung.com  Wed Feb  7 21:31:10 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Wed, 07 Feb 2007 21:31:10 -0600
Subject: [VoIP] VT1005 on Asterisk Polarity Reversal on Answer
In-Reply-To: <A691C3685A152E4AAC9608B8D8E11316686194@godzilla.hickory.vyger.net>
References: <A691C3685A152E4AAC9608B8D8E11316686194@godzilla.hickory.vyger.net>
Message-ID: <20070207213110.gijfp81z4088wc44@mail.shaneyoung.com>

For those who are testing the VT1005:

Does it support polarity reversal on answer?

--Shane



From ka2wft at arrl.net  Wed Feb  7 21:39:34 2007
From: ka2wft at arrl.net (Doug Alderdice)
Date: Wed, 07 Feb 2007 22:39:34 -0500
Subject: [VoIP] VT1005 on Asterisk Polarity Reversal on Answer
In-Reply-To: <20070207213110.gijfp81z4088wc44@mail.shaneyoung.com>
References: <A691C3685A152E4AAC9608B8D8E11316686194@godzilla.hickory.vyger.net>
	<A691C3685A152E4AAC9608B8D8E11316686194@godzilla.hickory.vyger.net>
Message-ID: <5.1.0.14.0.20070207223758.00bb3840@incoming.verizon.net>

At 09:31 PM 2/7/2007 -0600, Shane wrote:
>For those who are testing the VT1005:
>
>Does it support polarity reversal on answer?

Not that I've seen so far, but it may be a setting that isn't mentioned in 
the manual.  The manual appears to be well less than "full disclosure" on 
all the available settings.  John J. may know more.

Doug.



From jnovack at stromberg-carlson.org  Thu Feb  8 10:56:12 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Thu, 08 Feb 2007 11:56:12 -0500
Subject: [VoIP] VT1005 on Asterisk Polarity Reversal on Answer
In-Reply-To: <20070207213110.gijfp81z4088wc44@mail.shaneyoung.com>
References: <A691C3685A152E4AAC9608B8D8E11316686194@godzilla.hickory.vyger.net>
	<20070207213110.gijfp81z4088wc44@mail.shaneyoung.com>
Message-ID: <45CB562C.40807@stromberg-carlson.org>



Shane Young wrote:

For those who are testing the VT1005:

Does it support polarity reversal on answer?

--Shane




Not quite sure what you mean by this question.
The V1005 is a dual port FXS that supports pulse dial, and interfaces to 
Asterisk via SIP.
Do you want reverse polarity when the called party answers?
The answer is no, id does not provide battery supervision except when 
the called party is busy. Then it either drops or reverses battery ( 
twice?) and the busy comes from the box
That is the way it works on Vonage, and also perhaps with Asterisk, 
though I am not sure what Sip response is sent to the ATA to cause that
An interesting feature is  if one flashes the box to place a call on 
hold, place another call through Asterisk, flash again and you are in a 
three way conference IN THE ATA. No star codes involved. If you don't 
set up the three way, after a period of time you hear two short beeps as 
a hold reminder.
I don't remember seeing that in the manual, but as Doug has mentioned, 
it isn't the greatest, and is easily missed.
I believe it was David Josephson who made the documentation available to 
me and perhaps Greg could put it on the ckts.info site as well.
Wonder if there is more to be had somewhere?

John Novack



From jjones3601 at yahoo.com  Thu Feb  8 20:31:49 2007
From: jjones3601 at yahoo.com (john jones)
Date: Thu, 8 Feb 2007 18:31:49 -0800 (PST)
Subject: [VoIP] VT1005 on Asterisk Polarity Reversal on Answer
In-Reply-To: <5.1.0.14.0.20070207223758.00bb3840@incoming.verizon.net>
Message-ID: <20070209023149.28909.qmail@web34303.mail.mud.yahoo.com>

Not That I know of either.

John




--- Doug Alderdice <ka2wft at arrl.net> wrote:

> At 09:31 PM 2/7/2007 -0600, Shane wrote:
> >For those who are testing the VT1005:
> >
> >Does it support polarity reversal on answer?
> 
> Not that I've seen so far, but it may be a setting
> that isn't mentioned in 
> the manual.  The manual appears to be well less than
> "full disclosure" on 
> all the available settings.  John J. may know more.
> 
> Doug.
> 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 


From lee at spenadel.com  Thu Feb  8 20:46:32 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Thu, 8 Feb 2007 21:46:32 -0500
Subject: [VoIP] SIP, NAT and Linksys ATA
Message-ID: <00b101c74bf4$825c29d0$87147d70$@com>

I have an non-subscribed Linksys PAP2T that I was using on my Asterisk
switch, now replaced with an Adtran T1 mux.  I'd like to redeploy the ATA at
a friend's house as a remote on my system.  I know that NAT and SIP don't
play well together, so I am wondering who has managed to get this
combination working and how the obstacles were overcome.

 

Thanks

Lee

 

 

 

 

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


From lee at spenadel.com  Sat Feb 10 19:20:30 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Sat, 10 Feb 2007 20:20:30 -0500
Subject: [VoIP] CallerID
Message-ID: <001701c74d7a$d5f39dd0$81dad970$@com>

I've set my caller ID globally in my Zapata.conf.  However, I want to be
able to have the actual CallerID of the Asterisk extension be sent with each
call.  What's the best approach to do this?  As expected, my CallerID
defined in Zapata is sent with each call, no matter the extension being
originated from.

 

Thanks

 

 

 

 

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


From pdwills at cedarknolltelephone.com  Sat Feb 10 19:37:59 2007
From: pdwills at cedarknolltelephone.com (Paul Wills)
Date: Sat, 10 Feb 2007 20:37:59 -0500
Subject: [VoIP] GoToIf
References: <001701c74d7a$d5f39dd0$81dad970$@com>
Message-ID: <004001c74d7d$445154a0$0301a8c0@Main>

Here's hopefully a quick question:

When using the GoToIf statement, is there a way to compare a numerical value 
and not a string value?

By the way, the ANI is now working using the Cisco router and a lot of weird 
code in extensions.conf.

PDW


From voiptandem at shaneyoung.com  Sat Feb 10 20:26:43 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Sat, 10 Feb 2007 20:26:43 -0600
Subject: [VoIP] CallerID
In-Reply-To: <001701c74d7a$d5f39dd0$81dad970$@com>
References: <001701c74d7a$d5f39dd0$81dad970$@com>
Message-ID: <20070210202643.a9d5hewk9ws4cko8@mail.shaneyoung.com>

Assuming all of your extensions are ZAP, you need to set it for each  
channel in zapata like this:


;Channel zap/1
;DID => 3115557318
callerid => Computer Room<3115557318>
signalling => fxo_ks
channel => 1
;--------------------------------------------------------------------------------
;Channel zap/2
;DID => 3115557319
callerid => Alarm System<3115557319>
signalling => fxo_ks
channel => 2
;--------------------------------------------------------------------------------
;Channel zap/3
;DID => 3115557333
callerid => Kids Room<3115557333>
signalling => fxo_ks
channel => 3
;--------------------------------------------------------------------------------
;Channel zap/4
;DID => 3115557321
callerid => MBR DirecTV<3115557321>
signalling => fxo_ks
channel => 4
;--------------------------------------------------------------------------------
;Channel zap/5
;DID => 3115557320
callerid => Computer Room<3115557320>
signalling => fxo_ks
channel => 5
;--------------------------------------------------------------------------------
;Channel zap/6
;DID => 3115557311
callerid => Shane's Room<3115557311>
signalling => fxo_ks
channel => 6
;--------------------------------------------------------------------------------
;Channel zap/7
;DID => 3115557324
callerid => Living Room<3115557324>
signalling => fxo_ks
channel => 7






Quoting Lee Spenadel <lee at spenadel.com>:

> I've set my caller ID globally in my Zapata.conf.  However, I want to be
> able to have the actual CallerID of the Asterisk extension be sent with each
> call.  What's the best approach to do this?  As expected, my CallerID
> defined in Zapata is sent with each call, no matter the extension being
> originated from.
>
>
>
> Thanks
>
>

>
>
>
>
>
>
>
> If your car could travel at the speed of light, would your headlights work?
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>





From lee at spenadel.com  Sat Feb 10 20:38:07 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Sat, 10 Feb 2007 21:38:07 -0500
Subject: [VoIP] CallerID
In-Reply-To: <20070210202643.a9d5hewk9ws4cko8@mail.shaneyoung.com>
References: <001701c74d7a$d5f39dd0$81dad970$@com>
	<20070210202643.a9d5hewk9ws4cko8@mail.shaneyoung.com>
Message-ID: <001f01c74d85$aa0d5ca0$fe2815e0$@com>

Ugh.  Thanks

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Shane Young
Sent: Saturday, February 10, 2007 9:27 PM
To: voip at ckts.info
Subject: Re: [VoIP] CallerID

Assuming all of your extensions are ZAP, you need to set it for each  
channel in zapata like this:


;Channel zap/1
;DID => 3115557318
callerid => Computer Room<3115557318>
signalling => fxo_ks
channel => 1
;---------------------------------------------------------------------------
-----
;Channel zap/2
;DID => 3115557319
callerid => Alarm System<3115557319>
signalling => fxo_ks
channel => 2
;---------------------------------------------------------------------------
-----
;Channel zap/3
;DID => 3115557333
callerid => Kids Room<3115557333>
signalling => fxo_ks
channel => 3
;---------------------------------------------------------------------------
-----
;Channel zap/4
;DID => 3115557321
callerid => MBR DirecTV<3115557321>
signalling => fxo_ks
channel => 4
;---------------------------------------------------------------------------
-----
;Channel zap/5
;DID => 3115557320
callerid => Computer Room<3115557320>
signalling => fxo_ks
channel => 5
;---------------------------------------------------------------------------
-----
;Channel zap/6
;DID => 3115557311
callerid => Shane's Room<3115557311>
signalling => fxo_ks
channel => 6
;---------------------------------------------------------------------------
-----
;Channel zap/7
;DID => 3115557324
callerid => Living Room<3115557324>
signalling => fxo_ks
channel => 7






Quoting Lee Spenadel <lee at spenadel.com>:

> I've set my caller ID globally in my Zapata.conf.  However, I want to be
> able to have the actual CallerID of the Asterisk extension be sent with
each
> call.  What's the best approach to do this?  As expected, my CallerID
> defined in Zapata is sent with each call, no matter the extension being
> originated from.
>
>
>
> Thanks
>
>

>
>
>
>
>
>
>
> If your car could travel at the speed of light, would your headlights
work?
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>




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


From voiptandem at shaneyoung.com  Sat Feb 10 20:51:57 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Sat, 10 Feb 2007 20:51:57 -0600
Subject: [VoIP] CallerID
In-Reply-To: <001f01c74d85$aa0d5ca0$fe2815e0$@com>
References: <001701c74d7a$d5f39dd0$81dad970$@com>
	<20070210202643.a9d5hewk9ws4cko8@mail.shaneyoung.com>
	<001f01c74d85$aa0d5ca0$fe2815e0$@com>
Message-ID: <20070210205157.f0i4o82144oggg44@mail.shaneyoung.com>

It's one of the good and bad things about some of the conf files.

In zapata.conf, every channel inherits the previous channels settings  
unless you specifically set them, which is why all of your channels  
inherited the callerid.

What I sent was just a snip of my config, what I do for every channel  
looks more like this:
;Channel zap/1
;DID => 3115557318
adsi => no
amaflags => default
callerid => Computer Room<3115557318>
callgroup => 1
callreturn => yes
callwaiting => no
callwaitingcallerid => no
context => sip-phones
echocancel => yes
echotraining => yes
group => 10
immediate => no
language => en
mailbox => 7318 at lyndale
pickupgroup => 1
relaxdtmf => no
rxgain => 0
signalling => fxo_ks
threewaycalling => yes
transfer => yes
txgain => 0
usecallerid => yes
useincomingcalleridonzaptransfer => yes
channel => 1

The last line "channel =>" is what says "inhereit the most recennt of  
every setting above.






Quoting Lee Spenadel <lee at spenadel.com>:

> Ugh.  Thanks
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Shane Young
> Sent: Saturday, February 10, 2007 9:27 PM
> To: voip at ckts.info
> Subject: Re: [VoIP] CallerID
>
> Assuming all of your extensions are ZAP, you need to set it for each
> channel in zapata like this:
>
>
> ;Channel zap/1
> ;DID => 3115557318
> callerid => Computer Room<3115557318>
> signalling => fxo_ks
> channel => 1
> ;---------------------------------------------------------------------------
> -----
> ;Channel zap/2
> ;DID => 3115557319
> callerid => Alarm System<3115557319>
> signalling => fxo_ks
> channel => 2
> ;---------------------------------------------------------------------------
> -----
> ;Channel zap/3
> ;DID => 3115557333
> callerid => Kids Room<3115557333>
> signalling => fxo_ks
> channel => 3
> ;---------------------------------------------------------------------------
> -----
> ;Channel zap/4
> ;DID => 3115557321
> callerid => MBR DirecTV<3115557321>
> signalling => fxo_ks
> channel => 4
> ;---------------------------------------------------------------------------
> -----
> ;Channel zap/5
> ;DID => 3115557320
> callerid => Computer Room<3115557320>
> signalling => fxo_ks
> channel => 5
> ;---------------------------------------------------------------------------
> -----
> ;Channel zap/6
> ;DID => 3115557311
> callerid => Shane's Room<3115557311>
> signalling => fxo_ks
> channel => 6
> ;---------------------------------------------------------------------------
> -----
> ;Channel zap/7
> ;DID => 3115557324
> callerid => Living Room<3115557324>
> signalling => fxo_ks
> channel => 7
>
>
>
>
>
>
> Quoting Lee Spenadel <lee at spenadel.com>:
>
>> I've set my caller ID globally in my Zapata.conf.  However, I want to be
>> able to have the actual CallerID of the Asterisk extension be sent with
> each
>> call.  What's the best approach to do this?  As expected, my CallerID
>> defined in Zapata is sent with each call, no matter the extension being
>> originated from.
>>
>>
>>
>> Thanks
>>
>>
>
>>
>>
>>
>>
>>
>>
>>
>> If your car could travel at the speed of light, would your headlights
> work?
>>
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>
>
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>





From martin at Princeton.EDU  Sat Feb 10 22:29:38 2007
From: martin at Princeton.EDU (Martin Harriss)
Date: Sat, 10 Feb 2007 23:29:38 -0500
Subject: [VoIP] GoToIf
In-Reply-To: <004001c74d7d$445154a0$0301a8c0@Main>
References: <001701c74d7a$d5f39dd0$81dad970$@com>
	<004001c74d7d$445154a0$0301a8c0@Main>
Message-ID: <45CE9BB2.8020804@Princeton.EDU>

Paul Wills wrote:
> Here's hopefully a quick question:
> 
> When using the GoToIf statement, is there a way to compare a numerical value 
> and not a string value?
> 
> By the way, the ANI is now working using the Cisco router and a lot of weird 
> code in extensions.conf.
> 
> PDW

The Asterisk book says that the '=' operator (and, for that matter, the 
other relational operators) perform an integer comparison if both 
arguments are integers.  Does that help?

Martin

From pdwills at cedarknolltelephone.com  Sun Feb 11 07:27:38 2007
From: pdwills at cedarknolltelephone.com (Paul Wills)
Date: Sun, 11 Feb 2007 08:27:38 -0500
Subject: [VoIP] GoToIf
References: <001701c74d7a$d5f39dd0$81dad970$@com>
	<004001c74d7d$445154a0$0301a8c0@Main> <45CE9BB2.8020804@Princeton.EDU>
Message-ID: <002901c74de0$cea28940$0301a8c0@Main>


----- Original Message ----- 
From: "Martin Harriss" <martin at Princeton.EDU>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Saturday, February 10, 2007 11:29 PM
Subject: Re: [VoIP] GoToIf


> Paul Wills wrote:
>> Here's hopefully a quick question:
>>
>> When using the GoToIf statement, is there a way to compare a numerical 
>> value
>> and not a string value?
>>
>> By the way, the ANI is now working using the Cisco router and a lot of 
>> weird
>> code in extensions.conf.
>>
>> PDW
>
> The Asterisk book says that the '=' operator (and, for that matter, the
> other relational operators) perform an integer comparison if both
> arguments are integers.  Does that help?
>

It could be that the "comparee" is not an integer.  I guess making sure it 
is will be the next step.

Thanks,

PDW


From greg at vyger.net  Sun Feb 11 21:00:14 2007
From: greg at vyger.net (Greg Blakely)
Date: Sun, 11 Feb 2007 21:00:14 -0600
Subject: [VoIP] Rare Phone
Message-ID: <A691C3685A152E4AAC9608B8D8E113166861B6@godzilla.hickory.vyger.net>

A rare phone, but is it worth $1784.00?
 
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=140082684076
 
 

From hockd at dteenergy.com  Mon Feb 12 06:12:20 2007
From: hockd at dteenergy.com (Dennis D Hock)
Date: Mon, 12 Feb 2007 07:12:20 -0500
Subject: [VoIP] SIP, NAT and Linksys ATA
Message-ID: <OFBB4BEE11.01A20124-ON85257280.00430C62-85257280.00430C6A@dteenergy.com>


Lee,

Greg was able to get me going using the SPA 2K that I have,.  I would be
more than happy to burn a copy of theos econfigs on my SPA and perhaps Greg
can fill in what he neede to do on the host.  Mine is working behind a
D-Link router and 4 portr integral switch from on the big box stores.

Also have you tried the web page seems to me Greg had posted some examples
of how this would go together.  At least I think there was some suggested
configs up there, could be wrong.

But if not there are enough us to be able to help.

Dennis Hock

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


To: <VoIP at ckts.info>
From: "Lee Spenadel" <lee at spenadel.com>
Sent by: voip-bounces at ckts.info
Date: 02/08/2007 09:46PM
Subject: [VoIP] SIP, NAT and Linksys ATA

I have an non-subscribed Linksys PAP2T that I was using on my Asterisk
switch, now replaced with an Adtran T1 mux.  I'd like to redeploy the ATA
at
a friend's house as a remote on my system.  I know that NAT and SIP don't
play well together, so I am wondering who has managed to get this
combination working and how the obstacles were overcome.



Thanks

Lee









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

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

From lee at spenadel.com  Mon Feb 12 06:25:01 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Mon, 12 Feb 2007 07:25:01 -0500
Subject: [VoIP] SIP, NAT and Linksys ATA
In-Reply-To: <OFBB4BEE11.01A20124-ON85257280.00430C62-85257280.00430C6A@dteenergy.com>
References: <OFBB4BEE11.01A20124-ON85257280.00430C62-85257280.00430C6A@dteenergy.com>
Message-ID: <002c01c74ea0$d1a2a6a0$74e7f3e0$@com>

Dennis,

That would be great.  I can hopefully adapt the config to the Linksys ATA.

Lee

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Dennis D Hock
Sent: Monday, February 12, 2007 7:12 AM
To: Voice Over IP Tandem for Analog Switches
Cc: VoIP at ckts.info
Subject: Re: [VoIP] SIP, NAT and Linksys ATA


Lee,

Greg was able to get me going using the SPA 2K that I have,.  I would be
more than happy to burn a copy of theos econfigs on my SPA and perhaps Greg
can fill in what he neede to do on the host.  Mine is working behind a
D-Link router and 4 portr integral switch from on the big box stores.

Also have you tried the web page seems to me Greg had posted some examples
of how this would go together.  At least I think there was some suggested
configs up there, could be wrong.

But if not there are enough us to be able to help.

Dennis Hock

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


To: <VoIP at ckts.info>
From: "Lee Spenadel" <lee at spenadel.com>
Sent by: voip-bounces at ckts.info
Date: 02/08/2007 09:46PM
Subject: [VoIP] SIP, NAT and Linksys ATA

I have an non-subscribed Linksys PAP2T that I was using on my Asterisk
switch, now replaced with an Adtran T1 mux.  I'd like to redeploy the ATA
at
a friend's house as a remote on my system.  I know that NAT and SIP don't
play well together, so I am wondering who has managed to get this
combination working and how the obstacles were overcome.



Thanks

Lee









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

_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voipProject 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  Mon Feb 12 10:51:21 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Mon, 12 Feb 2007 11:51:21 -0500
Subject: [VoIP] Rare Phone
In-Reply-To: <A691C3685A152E4AAC9608B8D8E113166861B6@godzilla.hickory.vyger.net>
References: <A691C3685A152E4AAC9608B8D8E113166861B6@godzilla.hickory.vyger.net>
Message-ID: <45D09B09.8000704@stromberg-carlson.org>

Mahogany Brown, with Gold trim and model 34
Pretty rare
I suppose several people think it is worth close to that, given the bidding

John Novack


Greg Blakely wrote:
> A rare phone, but is it worth $1784.00?
>  
> http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=140082684076
>  
>  
> _______________________________________________
> 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  Mon Feb 12 10:55:20 2007
From: jnovack at stromberg-carlson.org (John Novack)
Date: Mon, 12 Feb 2007 11:55:20 -0500
Subject: [VoIP] SIP, NAT and Linksys ATA
In-Reply-To: <002c01c74ea0$d1a2a6a0$74e7f3e0$@com>
References: <OFBB4BEE11.01A20124-ON85257280.00430C62-85257280.00430C6A@dteenergy.com>
	<002c01c74ea0$d1a2a6a0$74e7f3e0$@com>
Message-ID: <45D09BF8.7080505@stromberg-carlson.org>

If I am not mistaken, though, Greg's machine is NOT behind a router.
 From the little I understand about NAT, two routers, each with unknown 
NAT handling ( seems there are at least 4 types of NAT handling ) 
rapidly make the problem more difficult
Kirt and I played with it a little many months ago, and the best we 
could do was one way audio.

Nest of luck getting it to work
Let everyone know how it was done, what devices, what model routers, etc.

John Novack


Lee Spenadel wrote:
> Dennis,
>
> That would be great.  I can hopefully adapt the config to the Linksys ATA.
>
> Lee
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Dennis D Hock
> Sent: Monday, February 12, 2007 7:12 AM
> To: Voice Over IP Tandem for Analog Switches
> Cc: VoIP at ckts.info
> Subject: Re: [VoIP] SIP, NAT and Linksys ATA
>
>
> Lee,
>
> Greg was able to get me going using the SPA 2K that I have,.  I would be
> more than happy to burn a copy of theos econfigs on my SPA and perhaps Greg
> can fill in what he neede to do on the host.  Mine is working behind a
> D-Link router and 4 portr integral switch from on the big box stores.
>
> Also have you tried the web page seems to me Greg had posted some examples
> of how this would go together.  At least I think there was some suggested
> configs up there, could be wrong.
>
> But if not there are enough us to be able to help.
>
> Dennis Hock
>
> -----voip-bounces at ckts.info wrote: -----
>
>
> To: <VoIP at ckts.info>
> From: "Lee Spenadel" <lee at spenadel.com>
> Sent by: voip-bounces at ckts.info
> Date: 02/08/2007 09:46PM
> Subject: [VoIP] SIP, NAT and Linksys ATA
>
> I have an non-subscribed Linksys PAP2T that I was using on my Asterisk
> switch, now replaced with an Adtran T1 mux.  I'd like to redeploy the ATA
> at
> a friend's house as a remote on my system.  I know that NAT and SIP don't
> play well together, so I am wondering who has managed to get this
> combination working and how the obstacles were overcome.
>
>
>
> Thanks
>
> Lee
>
>
>
>
>
>
>
>
>
> If your car could travel at the speed of light, would your headlights work?
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voipProject Web Page:
> http://www.ckts.info/
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

From greg at vyger.net  Mon Feb 12 12:28:56 2007
From: greg at vyger.net (Greg Blakely)
Date: Mon, 12 Feb 2007 12:28:56 -0600
Subject: [VoIP] SIP, NAT and Linksys ATA
Message-ID: <A691C3685A152E4AAC9608B8D8E113166861B8@godzilla.hickory.vyger.net>

The one I currently use to host ATAs is not natted.  However, there was
a period of time when Dennis homed to my asterisk box at my house, which
was natted.  What I had to do then was port forward port 5060 udp and
10000-20000 to my asterisk box. 

> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] 
> On Behalf Of John Novack
> Sent: Monday, February 12, 2007 10:55 AM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] SIP, NAT and Linksys ATA
> 
> If I am not mistaken, though, Greg's machine is NOT behind a router.
>  From the little I understand about NAT, two routers, each 
> with unknown NAT handling ( seems there are at least 4 types 
> of NAT handling ) rapidly make the problem more difficult 
> Kirt and I played with it a little many months ago, and the 
> best we could do was one way audio.
> 
> Nest of luck getting it to work
> Let everyone know how it was done, what devices, what model 
> routers, etc.
> 
> John Novack
> 
> 
> Lee Spenadel wrote:
> > Dennis,
> >
> > That would be great.  I can hopefully adapt the config to 
> the Linksys ATA.
> >
> > Lee
> >
> > -----Original Message-----
> > From: voip-bounces at ckts.info 
> [mailto:voip-bounces at ckts.info] On Behalf 
> > Of Dennis D Hock
> > Sent: Monday, February 12, 2007 7:12 AM
> > To: Voice Over IP Tandem for Analog Switches
> > Cc: VoIP at ckts.info
> > Subject: Re: [VoIP] SIP, NAT and Linksys ATA
> >
> >
> > Lee,
> >
> > Greg was able to get me going using the SPA 2K that I 
> have,.  I would 
> > be more than happy to burn a copy of theos econfigs on my SPA and 
> > perhaps Greg can fill in what he neede to do on the host.  Mine is 
> > working behind a D-Link router and 4 portr integral switch 
> from on the big box stores.
> >
> > Also have you tried the web page seems to me Greg had posted some 
> > examples of how this would go together.  At least I think there was 
> > some suggested configs up there, could be wrong.
> >
> > But if not there are enough us to be able to help.
> >
> > Dennis Hock
> >
> > -----voip-bounces at ckts.info wrote: -----
> >
> >
> > To: <VoIP at ckts.info>
> > From: "Lee Spenadel" <lee at spenadel.com> Sent by: 
> > voip-bounces at ckts.info
> > Date: 02/08/2007 09:46PM
> > Subject: [VoIP] SIP, NAT and Linksys ATA
> >
> > I have an non-subscribed Linksys PAP2T that I was using on 
> my Asterisk 
> > switch, now replaced with an Adtran T1 mux.  I'd like to 
> redeploy the 
> > ATA at a friend's house as a remote on my system.  I know 
> that NAT and 
> > SIP don't play well together, so I am wondering who has 
> managed to get 
> > this combination working and how the obstacles were overcome.
> >
> >
> >
> > Thanks
> >
> > Lee
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > If your car could travel at the speed of light, would your 
> headlights work?
> >
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voipProject Web Page:
> > http://www.ckts.info/
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> >   
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 

From stfkerman at jps.net  Mon Feb 12 14:17:41 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Mon, 12 Feb 2007 15:17:41 -0500
Subject: [VoIP] Rare Phone
In-Reply-To: <45D09B09.8000704@stromberg-carlson.org>
References: <A691C3685A152E4AAC9608B8D8E113166861B6@godzilla.hickory.vyger.net>
	<45D09B09.8000704@stromberg-carlson.org>
Message-ID: <45D0CB65.8070409@jps.net>

Consider too as a base line that a completely common ordinary black one 
went for $261 around the same time.

Steph

John Novack wrote:
> Mahogany Brown, with Gold trim and model 34
> Pretty rare
> I suppose several people think it is worth close to that, given the bidding
>
> John Novack
>
>
> Greg Blakely wrote:
>   
>> A rare phone, but is it worth $1784.00?
>>  
>> http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=140082684076
>>  
>>  
>> _______________________________________________
>> 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 hockd at dteenergy.com  Mon Feb 12 14:24:40 2007
From: hockd at dteenergy.com (Dennis D Hock)
Date: Mon, 12 Feb 2007 15:24:40 -0500
Subject: [VoIP] SIP, NAT and Linksys ATA
Message-ID: <OF72AC3310.C8970054-ON85257280.00701F79-85257280.00701F7E@dteenergy.com>


John you may be right.  I don't recall what Greg did toget me up and
running butI can get a copy of my SPA configs.  Maybe it will help.

Thanks,

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


To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
From: John Novack <jnovack at stromberg-carlson.org>
Sent by: voip-bounces at ckts.info
Date: 02/12/2007 11:55AM
Subject: Re: [VoIP] SIP, NAT and Linksys ATA

If I am not mistaken, though, Greg's machine is NOT behind a router.
>From the little I understand about NAT, two routers, each with unknown
NAT handling ( seems there are at least 4 types of NAT handling )
rapidly make the problem more difficult
Kirt and I played with it a little many months ago, and the best we
could do was one way audio.

Nest of luck getting it to work
Let everyone know how it was done, what devices, what model routers, etc.

John Novack


Lee Spenadel wrote:
> Dennis,
>
> That would be great.  I can hopefully adapt the config to the Linksys
ATA.
>
> Lee
>
> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
> Dennis D Hock
> Sent: Monday, February 12, 2007 7:12 AM
> To: Voice Over IP Tandem for Analog Switches
> Cc: VoIP at ckts.info
> Subject: Re: [VoIP] SIP, NAT and Linksys ATA
>
>
> Lee,
>
> Greg was able to get me going using the SPA 2K that I have,.  I would be
> more than happy to burn a copy of theos econfigs on my SPA and perhaps
Greg
> can fill in what he neede to do on the host.  Mine is working behind a
> D-Link router and 4 portr integral switch from on the big box stores.
>
> Also have you tried the web page seems to me Greg had posted some
examples
> of how this would go together.  At least I think there was some suggested
> configs up there, could be wrong.
>
> But if not there are enough us to be able to help.
>
> Dennis Hock
>
> -----voip-bounces at ckts.info wrote: -----
>
>
> To: <VoIP at ckts.info>
> From: "Lee Spenadel" <lee at spenadel.com>
> Sent by: voip-bounces at ckts.info
> Date: 02/08/2007 09:46PM
> Subject: [VoIP] SIP, NAT and Linksys ATA
>
> I have an non-subscribed Linksys PAP2T that I was using on my Asterisk
> switch, now replaced with an Adtran T1 mux.  I'd like to redeploy the ATA
> at
> a friend's house as a remote on my system.  I know that NAT and SIP don't
> play well together, so I am wondering who has managed to get this
> combination working and how the obstacles were overcome.
>
>
>
> Thanks
>
> Lee
>
>
>
>
>
>
>
>
>
> If your car could travel at the speed of light, would your headlights
work?
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voipProjectWeb Page:
> http://www.ckts.info/
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voipProject Web Page:
http://www.ckts.info/

From ratguy at bellsouth.net  Mon Feb 12 17:29:02 2007
From: ratguy at bellsouth.net (Jayson Smith)
Date: Mon, 12 Feb 2007 18:29:02 -0500
Subject: [VoIP] Off-Topic - Daylight Saving Time
Message-ID: <001d01c74efd$9429ac80$0600a8c0@bluegrasspals.com>

Hello,
I have written a document which describes how to change the timezone data in
Windows operating systems in order to deal with the new Daylight Saving Time
rules that go into effect this year in the United States. If you live in the
United States and have not taken any action to correct this problem, you
need to read this, otherwise your computer's clock will be inaccurate for
three weeks in the spring and one week in the fall each year. This document
is written from a blind person's perspective, so I tell you how to do things
from the keyboard, rather than using the mouse. You can find this document,
and the required timezone editor program, here.
http://www.bluegrasspals.com/dst.html
Hope this helps.
Jayson


From david at josephson.com  Mon Feb 12 19:37:50 2007
From: david at josephson.com (David Josephson)
Date: Mon, 12 Feb 2007 17:37:50 -0800
Subject: [VoIP] SS400 awake
Message-ID: <45D1166E.2000207@josephson.com>

The switchroom here is beginning to take shape. We finally migrated the 
office phones (what used to be a one man shop with a couple helpers is 
now five people full time making microphones) from a KTU-less system and 
an answering machine to an AT&T Merlin Legend system with display phones 
and voicemail, which has a rather convoluted programming interface but 
works well.

Next step was the EM switch, and the WE Switching System 400 that I got 
from Larry Tighe is now rewired after its disassembly to come upstairs, 
and working. This SS400 is basically a stripped-down 756A, which 
provides (in one rack of three slides) 8 lines of 1A1 key phone service 
and a 40 line crossbar intercom with dial pulse register and junctor 
circuits like those in a 756A. There are no trunks per se, but three 
"universal line" circuits which provide single-digit links to external 
tie-line, paging or dictation trunks.

I am still looking for a good way to provide touch tone calling on the 
SS400. I have some tone-to-pulse converters but that seems like a real 
hack. The BSP's suggest either of two types of touch tone receivers. I 
have a pair of the better A-type receivers, but these require a J58847AE 
L1 relay applique to translate two-of-seven (individual tone decode 
outputs) to one-of-ten plus steering. There is an option for internal 
mounting of the "junior" C-type touchtone receivers, which are basically 
the circuit cards of a 427B or C decoder mounted on a rack plate. I have 
an external rack for mounting decoders etc. but would really like to 
keep everything inside the switch.

So ... I would like to find a precise dial tone plant (either the 
correct 404C tone generator or something like it), two J58844A touch 
tone receivers or equivalent (one rack unit, one-of-ten output), and two 
AF156 relays to make the required modification to the dial pulse 
register circuits. Anyone have these pieces in their trading pile?

--
David Josephson



From voiptandem at shaneyoung.com  Mon Feb 12 20:54:36 2007
From: voiptandem at shaneyoung.com (Shane Young)
Date: Mon, 12 Feb 2007 20:54:36 -0600
Subject: [VoIP] Off-Topic - Daylight Saving Time
In-Reply-To: <001d01c74efd$9429ac80$0600a8c0@bluegrasspals.com>
References: <001d01c74efd$9429ac80$0600a8c0@bluegrasspals.com>
Message-ID: <20070212205436.sruss82nkss0wgoc@mail.shaneyoung.com>

Isn't there a Microsoft Update that just fixes this problem?

--Shane
Quoting Jayson Smith <ratguy at bellsouth.net>:

> Hello,
> I have written a document which describes how to change the timezone data in
> Windows operating systems in order to deal with the new Daylight Saving Time
> rules that go into effect this year in the United States. If you live in the
> United States and have not taken any action to correct this problem, you
> need to read this, otherwise your computer's clock will be inaccurate for
> three weeks in the spring and one week in the fall each year. This document
> is written from a blind person's perspective, so I tell you how to do things
> from the keyboard, rather than using the mouse. You can find this document,
> and the required timezone editor program, here.
> http://www.bluegrasspals.com/dst.html
> Hope this helps.
> Jayson




From lee at spenadel.com  Mon Feb 12 21:12:03 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Mon, 12 Feb 2007 22:12:03 -0500
Subject: [VoIP] Off-Topic - Daylight Saving Time
In-Reply-To: <20070212205436.sruss82nkss0wgoc@mail.shaneyoung.com>
References: <001d01c74efd$9429ac80$0600a8c0@bluegrasspals.com>
	<20070212205436.sruss82nkss0wgoc@mail.shaneyoung.com>
Message-ID: <001501c74f1c$c29a7710$47cf6530$@com>

https://partner.microsoft.com/us/40029783

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Shane Young
Sent: Monday, February 12, 2007 9:55 PM
To: voip at ckts.info
Subject: Re: [VoIP] Off-Topic - Daylight Saving Time

Isn't there a Microsoft Update that just fixes this problem?

--Shane
Quoting Jayson Smith <ratguy at bellsouth.net>:

> Hello,
> I have written a document which describes how to change the timezone data
in
> Windows operating systems in order to deal with the new Daylight Saving
Time
> rules that go into effect this year in the United States. If you live in
the
> United States and have not taken any action to correct this problem, you
> need to read this, otherwise your computer's clock will be inaccurate for
> three weeks in the spring and one week in the fall each year. This
document
> is written from a blind person's perspective, so I tell you how to do
things
> from the keyboard, rather than using the mouse. You can find this
document,
> and the required timezone editor program, here.
> http://www.bluegrasspals.com/dst.html
> Hope this helps.
> Jayson



_______________________________________________
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  Mon Feb 12 22:30:17 2007
From: lee at spenadel.com (Lee Spenadel)
Date: Mon, 12 Feb 2007 23:30:17 -0500
Subject: [VoIP] Daylight Saving Update
Message-ID: <001601c74f27$da267220$8e735660$@com>

This is a better link:

 

http://support.microsoft.com/gp/dst_homeuser#howto

 

 

 

 

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


From stfkerman at jps.net  Tue Feb 13 02:28:01 2007
From: stfkerman at jps.net (Steph Kerman)
Date: Tue, 13 Feb 2007 03:28:01 -0500
Subject: [VoIP] Daylight Saving Update
In-Reply-To: <001601c74f27$da267220$8e735660$@com>
References: <001601c74f27$da267220$8e735660$@com>
Message-ID: <45D17691.90404@jps.net>

Well it looks like those of us who are determined to run Win2K or older 
versions rather than pay Mr. Gate's vigorish to run XP are fated to find 
another solution.  So perhaps Jayson's efforts will find a receptive 
audience.

Steph

Lee Spenadel wrote:
> This is a better link:
>
>  
>
> http://support.microsoft.com/gp/dst_homeuser#howto
>
>  
>
>  
>
>  
>
>  
>
> If your car could travel at the speed of light, would your headlights work?
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>   

From ratguy at bellsouth.net  Tue Feb 13 02:36:11 2007
From: ratguy at bellsouth.net (Jayson Smith)
Date: Tue, 13 Feb 2007 03:36:11 -0500
Subject: [VoIP] Daylight Saving Update
References: <001601c74f27$da267220$8e735660$@com> <45D17691.90404@jps.net>
Message-ID: <003f01c74f4a$03f34fc0$0600a8c0@bluegrasspals.com>

Hi,
That was my point in writing that article, this method works even if you're
running an unsupported OS for which Microsoft doesn't provide an update.
Jayson.

----- Original Message ----- 
From: "Steph Kerman" <stfkerman at jps.net>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Tuesday, February 13, 2007 3:28 AM
Subject: Re: [VoIP] Daylight Saving Update


> Well it looks like those of us who are determined to run Win2K or older
> versions rather than pay Mr. Gate's vigorish to run XP are fated to find
> another solution.  So perhaps Jayson's efforts will find a receptive
> audience.
>
> Steph
>
> Lee Spenadel wrote:
> > This is a better link:
> >
> >
> >
> > http://support.microsoft.com/gp/dst_homeuser#howto
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > If your car could travel at the speed of light, would your headlights
work?
> >
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> >
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>


From hockd at dteenergy.com  Tue Feb 13 05:04:21 2007
From: hockd at dteenergy.com (Dennis D Hock)
Date: Tue, 13 Feb 2007 06:04:21 -0500
Subject: [VoIP] My SPA and D-Link Configs  (was SIP, NAT and Linksys ATA)
Message-ID: <OF6568BA2F.01F546CE-ON85257281.003C462B-85257281.003CD2E4@dteenergy.com>


Lee,

Here is what I have from my D-Link firewall which John helped me with and
the SPA 2K which Greg helped me with.  I did not include the "Regional"
settings as it only contains all the Class Feature codes, time (which I
can't get set) and "FXS Port Impedance."  Like wise I didn't include the
configs from "line 1" as that is set up for Fwd.Pulver .com but it doesn't
seem to want to work for quite a while.  Anyway I hope these are of some
value to you in the quest.

Dennis Hock

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


To: "'Voice Over IP Tandem for Analog Switches'" <voip at ckts.info>
From: "Lee Spenadel" <lee at spenadel.com>
Sent by: voip-bounces at ckts.info
Date: 02/12/2007 07:25AM
Subject: Re: [VoIP] SIP, NAT and Linksys ATA

Dennis,

That would be great.  I can hopefully adapt the config to the Linksys ATA.

Lee

-----Original Message-----
From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf Of
Dennis D Hock
Sent: Monday, February 12, 2007 7:12 AM
To: Voice Over IP Tandem for Analog Switches
Cc: VoIP at ckts.info
Subject: Re: [VoIP] SIP, NAT and Linksys ATA


Lee,

Greg was able to get me going using the SPA 2K that I have,.  I would be
more than happy to burn a copy of theos econfigs on my SPA and perhaps Greg
can fill in what he neede to do on the host.  Mine is working behind a
D-Link router and 4 portr integral switch from on the big box stores.

Also have you tried the web page seems to me Greg had posted some examples
of how this would go together.  At least I think there was some suggested
configs up there, could be wrong.

But if not there are enough us to be able to help.

Dennis Hock

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


To: <VoIP at ckts.info>
From: "Lee Spenadel" <lee at spenadel.com>
Sent by: voip-bounces at ckts.info
Date: 02/08/2007 09:46PM
Subject: [VoIP] SIP, NAT and Linksys ATA

I have an non-subscribed Linksys PAP2T that I was using on my Asterisk
switch, now replaced with an Adtran T1 mux.  I'd like to redeploy the ATA
at
a friend's house as a remote on my system.  I know that NAT and SIP don't
play well together, so I am wondering who has managed to get this
combination working and how the obstacles were overcome.



Thanks

Lee









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

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

_______________________________________________
VoIP mailing list
VoIP at ckts.info
http://lists.ckts.info/mailman/listinfo/voip
Project Web Page: http://www.ckts.info/
(See attached file: _0213051317_001.pdf)
(See attached file: _021