From radiorapido at yahoo.co.uk  Fri Dec  1 09:21:04 2006
From: radiorapido at yahoo.co.uk (Duncan Hill)
Date: Fri, 1 Dec 2006 15:21:04 +0000 (GMT)
Subject: [VoIP] New member
Message-ID: <20061201152104.3687.qmail@web27211.mail.ukl.yahoo.com>

Hello all,

Just introducing myself to you all here, I'm currently
on the south coast of England, but in a couple of
days, I will be moving to the Hertfordshire area.

My new place of work is vast enough in size to have a
little Asterisk system of sorts running and several
extensions. A telephone line will be installed there
next Wednesday (with Featureline - a mistake?) and I'm
basically responsible for it all, including the
payphone and PBX!

I have been interested in 'phones for ages, probably
since I was a single-figure number! I hope to have
enough time to spare fiddling with it all, altho'
money is always an issue of course, but I hope to
acquire some analogue exchange equipment, once I'm
able to afford it.

I'm hoping to get my office code sorted out later and
that it'll all be connected up and running by
Christmas or the New Year...? Perhaps a bit over
optimistic!

Anyway, that was s'posed to only be a short
introduction, no doubt I'll be posting more soon!

-regards
Duncan


	
	
		
___________________________________________________________ 
All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease of use." - PC Magazine 
http://uk.docs.yahoo.com/nowyoucan.html


From ian at uax.org.uk  Fri Dec  1 09:29:58 2006
From: ian at uax.org.uk (Ian Jolly)
Date: Fri, 1 Dec 2006 15:29:58 -0000
Subject: [VoIP] [-SPAM-]  New member
References: <20061201152104.3687.qmail@web27211.mail.ukl.yahoo.com>
Message-ID: <019401c7155d$8fbe1750$0b01a8c0@acer1dd0bbc6d0>

Welcome Duncan

>From here in North Wales. Running an Asterisk box which feeds into a digital 
PABX used purely for LD/DTMF conversion and linking various exchanges to the 
Asterisk -  no BT lines connected to it. I have currently two 
electro-mechanical exchanges connected to it - a former GPO UAX 5 dating 
from 1929 that served a village in Northumberland until circa 1950 plus a 
small cross bar exchange.  Other UAXs from my collection will eventually get 
linked to it when I get them sorted out.

Good to have you on-line.  You've only got to ask for help - some of us are 
still regrowing our finger nails after the steep climb.

Regards

Ian Jolly


+44 (0)352 85 26 (via a 1929 GPO Rural Automatic eXchange!)
CNET - the Heritage Telephone Network


----- Original Message ----- 
From: "Duncan Hill" <radiorapido at yahoo.co.uk>
To: <voip at ckts.info>
Sent: Friday, December 01, 2006 3:21 PM
Subject: [-SPAM-] [VoIP] New member


> Hello all,
>
> Just introducing myself to you all here, I'm currently
> on the south coast of England, but in a couple of
> days, I will be moving to the Hertfordshire area.
>
> My new place of work is vast enough in size to have a
> little Asterisk system of sorts running and several
> extensions. A telephone line will be installed there
> next Wednesday (with Featureline - a mistake?) and I'm
> basically responsible for it all, including the
> payphone and PBX!
>
> I have been interested in 'phones for ages, probably
> since I was a single-figure number! I hope to have
> enough time to spare fiddling with it all, altho'
> money is always an issue of course, but I hope to
> acquire some analogue exchange equipment, once I'm
> able to afford it.
>
> I'm hoping to get my office code sorted out later and
> that it'll all be connected up and running by
> Christmas or the New Year...? Perhaps a bit over
> optimistic!
>
> Anyway, that was s'posed to only be a short
> introduction, no doubt I'll be posting more soon!
>
> -regards
> Duncan
>
>
>
>
>
> ___________________________________________________________
> All new Yahoo! Mail "The new Interface is stunning in its simplicity and 
> ease of use." - PC Magazine
> http://uk.docs.yahoo.com/nowyoucan.html
>
> _______________________________________________
> 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: 7.1.409 / Virus Database: 268.15.3/561 - Release Date: 01/12/2006
> 



From pdwills at verizon.net  Fri Dec  1 09:38:35 2006
From: pdwills at verizon.net (Paul Wills)
Date: Fri, 01 Dec 2006 09:38:35 -0600 (CST)
Subject: [VoIP] New Question
Message-ID: <16157932.7901481164987519936.JavaMail.root@vms061.mailsrvcs.net>

Here's a question:

One of my incoming CNET lines goes to an Automatic Electric code call.  The "confirmation" that a page is taking place consists of an audible pattern of buzzes being fed back to the caller.  Unfortunately, the Asterisk seems to clip out any sudden tone or sound (It also happens when one calls my Northeast Electronics milliwatt source).  This causes the caller to only hear a very short burst of tone.  

Is this the echo cancelling at work?

Is there a way to remove the tone break without screwing up the echo cancelling too much?

Thanks in advance,

PDW


From g4vft at btinternet.com  Fri Dec  1 10:10:44 2006
From: g4vft at btinternet.com (Jonathan Kay)
Date: Fri, 01 Dec 2006 16:10:44 +0000
Subject: [VoIP] New Question
In-Reply-To: <16157932.7901481164987519936.JavaMail.root@vms061.mailsrvcs.net>
References: <16157932.7901481164987519936.JavaMail.root@vms061.mailsrvcs.net>
Message-ID: <45705404.6000309@btinternet.com>

Paul Wills wrote:
> Here's a question:
>
> One of my incoming CNET lines goes to an Automatic Electric code call.  The "confirmation" that a page is taking place consists of an audible pattern of buzzes being fed back to the caller.  Unfortunately, the Asterisk seems to clip out any sudden tone or sound (It also happens when one calls my Northeast Electronics milliwatt source).  This causes the caller to only hear a very short burst of tone.  
>
> Is this the echo cancelling at work?
>
> Is there a way to remove the tone break without screwing up the echo cancelling too much?
>   
Hi Paul,
You could try *echotraining=no*
in your zapata.conf, just against this relevant zap channel entry.
The Echo training, mutes the channel, while it sends a tone pulse to 
train the echo canceller.
The echo canceller should still train itself, but just take longer.


Jon

From jnovack at stromberg-carlson.org  Fri Dec  1 10:08:14 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Fri, 01 Dec 2006 11:08:14 -0500
Subject: [VoIP] New Question
In-Reply-To: <16157932.7901481164987519936.JavaMail.root@vms061.mailsrvcs.net>
References: <16157932.7901481164987519936.JavaMail.root@vms061.mailsrvcs.net>
Message-ID: <4570536E.1000501@stromberg-carlson.org>

That is curious
My NE electronics milliwatt doesn't ( didn't ) do that. I just 
discovered it isn't answering at all, something else to fix!
MY NE is connected to two station lines, but right now all I get is RNA!

Do you access your code call in a similar manner?
Asterisk => station line => LF/selector/connector => NE electronics
Don't think that is a byproduct of the EC though
Tx and Rcv levels?


John Novack


Paul Wills wrote:
> Here's a question:
>
> One of my incoming CNET lines goes to an Automatic Electric code call.  The "confirmation" that a page is taking place consists of an audible pattern of buzzes being fed back to the caller.  Unfortunately, the Asterisk seems to clip out any sudden tone or sound (It also happens when one calls my Northeast Electronics milliwatt source).  This causes the caller to only hear a very short burst of tone.  
>
> Is this the echo cancelling at work?
>
> Is there a way to remove the tone break without screwing up the echo cancelling too much?
>
> Thanks in advance,
>
> PDW
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
>   

From pdwills at verizon.net  Fri Dec  1 10:17:13 2006
From: pdwills at verizon.net (Paul Wills)
Date: Fri, 01 Dec 2006 10:17:13 -0600 (CST)
Subject: [VoIP] New Question
Message-ID: <2835787.7915451164989833448.JavaMail.root@vms061.mailsrvcs.net>

It's physically connected as you describe.  The NE electronics unit interrupts the tone every 10 seconds or so.  Each time the tone restarts, there is a short break inserted by the Asterisk so you get a ee--------eeeeeeeeeeee sort of sound (eeee being tone and ----- being silence) at the beginning of each tone burst.  On the code call, the silence interval is just about as long as the answerback tone so all the caller hears is a very short burst of tone and then nothing.

PDW


>From: John Novack <jnovack at stromberg-carlson.org>
>Date: 2006/12/01 Fri AM 10:08:14 CST
>To: pdwills at cedarknolltelephone.com, 
	Voice Over IP Tandem for Analog Switches <voip at ckts.info>
>Subject: Re: [VoIP] New Question

>  That is curious
>My NE electronics milliwatt doesn't ( didn't ) do that. I justdiscovered it isn't answering at all, something else to fix!
>MY NE is connected to two station lines, but right now all I get is RNA!
>
>Do you access your code call in a similar manner?
>Asterisk => station line => LF/selector/connector => NEelectronics
>Don't think that is a byproduct of the EC though
>Tx and Rcv levels?
>
>
>John Novack
>
>
>Paul Wills wrote:  Here's a question:One of my incoming CNET lines goes to an Automatic Electric code call.  The "confirmation" that a page is taking place consists of an audible pattern of buzzes being fed back to the caller.  Unfortunately, the Asterisk seems to clip out any sudden tone or sound (It also happens when one calls my Northeast Electronics milliwatt source).  This causes the caller to only hear a very short burst of tone.  Is this the echo cancelling at work?Is there a way to remove the tone break without screwing up the echo cancelling too much?Thanks in advance,PDW_______________________________________________VoIP mailing listVoIP at ckts.infohttp://lists.ckts.info/mailman/listinfo/voipProject Web Page: http://www.ckts.info/  



From ikj1234i at yahoo.com  Fri Dec  1 15:43:02 2006
From: ikj1234i at yahoo.com (ikjtel)
Date: Fri, 1 Dec 2006 13:43:02 -0800 (PST)
Subject: [VoIP] New Question
In-Reply-To: <16157932.7901481164987519936.JavaMail.root@vms061.mailsrvcs.net>
Message-ID: <20061201214302.76206.qmail@web51609.mail.yahoo.com>


Hey Paul

Suggest that, for testing purposes, you disable the
echo cancelling stuff entirely.  I think echocancel=no
(in zapata.conf) should do this (but haven't
confirmed).  If the problem persists after you've
properly disabled the echo cancellation features, you
will have eliminated that as a variable. 

What exact FXO hardware are you using?  Is it a
TDM400P or a X100P or what?  Reason for asking is that
a lot of these cards will mute the audio (generally in
the range of 150-200 msec. or so) in certain
situations.   In particular if the answering unit is
reversing battery as part of its cycling, that would
cause the symptom you mention.  You might also monitor
the loop DC current to see if it's steady or if it
fluctuates...

Max

--- Paul Wills <pdwills at verizon.net> wrote:

> Here's a question:
> 
> One of my incoming CNET lines goes to an Automatic
> Electric code call.  The "confirmation" that a page
> is taking place consists of an audible pattern of
> buzzes being fed back to the caller.  Unfortunately,
> the Asterisk seems to clip out any sudden tone or
> sound (It also happens when one calls my Northeast
> Electronics milliwatt source).  This causes the
> caller to only hear a very short burst of tone.  
> 
> Is this the echo cancelling at work?
> 
> Is there a way to remove the tone break without
> screwing up the echo cancelling too much?
> 
> Thanks in advance,
> 
> PDW
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 



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


From g4vft at btinternet.com  Fri Dec  1 16:02:12 2006
From: g4vft at btinternet.com (Jonathan Kay)
Date: Fri, 01 Dec 2006 22:02:12 +0000
Subject: [VoIP] Atcom AG188 Problems
Message-ID: <4570A664.8090402@btinternet.com>



Guys,
I'm having trouble with calls to Steve's ATA, which is registered to my *.
Call from an IAX phone, off the same * box work fine, but anything from 
the ZAP interface or Sipgate Portal, get rejected by the ATA.
The caller gets a congestion announcement, but the phone on the ATA 
rings until picked up.
Any suggestions please?
Jon

Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX     Subclass: 
REGREQ
  Timestamp: 00003ms  SCall: 00032  DCall: 00005 [90.195.139.24:4569]
  USERNAME        : stevebarlow
  REFRESH         : 60
  MD5 RESULT      : 799eb0120db4ba8a6b5d37197e279df9

Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass: 
REGACK
  Timestamp: 00049ms  SCall: 00005  DCall: 00032 [90.195.139.24:4569]
  USERNAME        : stevebarlow
  DATE TIME       : 2006-12-01  21:46:52
  REFRESH         : 60
  APPARENT ADDRES : IPV4 90.195.139.24:4569
  CALLING NUMBER  :
  CALLING NAME    : 44(0)706 Steve Barlow

   -- Executing Dial("Zap/1-1", "IAX2/stevebarlow|30") in new stack
   -- Called stevebarlow
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass: NEW
  Timestamp: 00011ms  SCall: 00009  DCall: 00000 [90.195.139.24:4569]
  VERSION         : 2
  CALLED NUMBER   : s
  CODEC_PREFS     : (ulaw|alaw|gsm|g726|lpc10|speex|ilbc|adpcm)
  CALLING PRESNTN : 67
  CALLING TYPEOFN : 0
  CALLING TRANSIT : 0
  LANGUAGE        : en
  USERNAME        : stevebarlow
  FORMAT          : 4
  CAPABILITY      : 65214
  ADSICPE         : 2
  DATE TIME       : 2006-12-01  21:46:52

Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass: ACK
  Timestamp: 00049ms  SCall: 00032  DCall: 00005 [90.195.139.24:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: ACK
  Timestamp: 00011ms  SCall: 00033  DCall: 00009 [90.195.139.24:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: 
ACCEPT
  Timestamp: 01000ms  SCall: 00033  DCall: 00009 [90.195.139.24:4569]
  FORMAT          : 4

   -- Call accepted by 90.195.139.24 (format ulaw)
   -- Format for call is ulaw
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX     Subclass: ACK
  Timestamp: 01000ms  SCall: 00009  DCall: 00033 [90.195.139.24:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass: 
RINGING
  Timestamp: 01000ms  SCall: 00033  DCall: 00009 [90.195.139.24:4569]
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass: ACK
  Timestamp: 01000ms  SCall: 00009  DCall: 00033 [90.195.139.24:4569]
   -- IAX2/stevebarlow-9 is ringing
Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: VOICE   Subclass: 4
  Timestamp: 01080ms  SCall: 00009  DCall: 00033 [90.195.139.24:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass: ACK
  Timestamp: 01080ms  SCall: 00033  DCall: 00009 [90.195.139.24:4569]
Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass: 
REJECT
  Timestamp: 01003ms  SCall: 00033  DCall: 00009 [90.195.139.24:4569]
  CAUSE           : Call rejected manually.

Dec  1 21:46:53 WARNING[2135]: chan_iax2.c:7100 socket_read: Call 
rejected by 90.195.139.24: Call rejected manually.
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX     Subclass: ACK
  Timestamp: 01003ms  SCall: 00009  DCall: 00033 [90.195.139.24:4569]
   -- Hungup 'IAX2/stevebarlow-9'
 == Everyone is busy/congested at this time (1:0/0/1)
   -- Executing Macro("Zap/1-1", "fastbusy") in new stack
   -- Executing Answer("Zap/1-1", "") in new stack
   -- Executing Wait("Zap/1-1", "1") in new stack
   -- Executing PlayTones("Zap/1-1", "!950/330|!1400/330|!1800/330|0") 
in new stack
   -- Executing Wait("Zap/1-1", "1") in new stack
   -- Executing Playback("Zap/1-1", "im-sorry") in new stack
   -- Playing 'im-sorry' (language 'en')
   -- Executing Playback("Zap/1-1", "all-circuits-busy-now") in new stack
   -- Playing 'all-circuits-busy-now' (language 'en')
   -- Executing Playback("Zap/1-1", "pls-try-call-later") in new 
stack                                                 




From ian at uax.org.uk  Fri Dec  1 16:06:43 2006
From: ian at uax.org.uk (Ian Jolly)
Date: Fri, 1 Dec 2006 22:06:43 -0000
Subject: [VoIP] Atcom AG188 Problems
References: <4570A664.8090402@btinternet.com>
Message-ID: <025b01c71594$fd4c7d20$0b01a8c0@acer1dd0bbc6d0>

Exactly the same problem is happening with an identical AG188 registered to 
my Asterisk box.

Three times I made calls from an extension on a PBX off my *box.  Each call 
was set up and just as the phone started to ring at the other end (we had a 
separate voice link between the locations open) - my CLI showed - call 
manually rejected and I dropped back to busy tone. Nothing had been touched 
at the AG188 end but the phone carried on ringing until answered.

I then tried from an IAX IP phone on my LAN and first attempt got the same 
result but clearing down at my end and then instantly pressing 'redial' on 
the IP phone and I got straight through. Out going calls from the ATA are 
fine.

Ian J


+44 (0)352 85 26 (via a 1929 GPO Rural Automatic eXchange!)
CNET - the Heritage Telephone Network

----- Original Message ----- 
From: "Jonathan Kay" <g4vft at btinternet.com>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Friday, December 01, 2006 10:02 PM
Subject: [-SPAM-] [VoIP] Atcom AG188 Problems


>
>
> Guys,
> I'm having trouble with calls to Steve's ATA, which is registered to my *.
> Call from an IAX phone, off the same * box work fine, but anything from
> the ZAP interface or Sipgate Portal, get rejected by the ATA.
> The caller gets a congestion announcement, but the phone on the ATA
> rings until picked up.
> Any suggestions please?
> Jon
>
> Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX     Subclass:
> REGREQ
>  Timestamp: 00003ms  SCall: 00032  DCall: 00005 [90.195.139.24:4569]
>  USERNAME        : stevebarlow
>  REFRESH         : 60
>  MD5 RESULT      : 799eb0120db4ba8a6b5d37197e279df9
>
> Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass:
> REGACK
>  Timestamp: 00049ms  SCall: 00005  DCall: 00032 [90.195.139.24:4569]
>  USERNAME        : stevebarlow
>  DATE TIME       : 2006-12-01  21:46:52
>  REFRESH         : 60
>  APPARENT ADDRES : IPV4 90.195.139.24:4569
>  CALLING NUMBER  :
>  CALLING NAME    : 44(0)706 Steve Barlow
>
>   -- Executing Dial("Zap/1-1", "IAX2/stevebarlow|30") in new stack
>   -- Called stevebarlow
> Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass: NEW
>  Timestamp: 00011ms  SCall: 00009  DCall: 00000 [90.195.139.24:4569]
>  VERSION         : 2
>  CALLED NUMBER   : s
>  CODEC_PREFS     : (ulaw|alaw|gsm|g726|lpc10|speex|ilbc|adpcm)
>  CALLING PRESNTN : 67
>  CALLING TYPEOFN : 0
>  CALLING TRANSIT : 0
>  LANGUAGE        : en
>  USERNAME        : stevebarlow
>  FORMAT          : 4
>  CAPABILITY      : 65214
>  ADSICPE         : 2
>  DATE TIME       : 2006-12-01  21:46:52
>
> Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass: ACK
>  Timestamp: 00049ms  SCall: 00032  DCall: 00005 [90.195.139.24:4569]
> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: ACK
>  Timestamp: 00011ms  SCall: 00033  DCall: 00009 [90.195.139.24:4569]
> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass:
> ACCEPT
>  Timestamp: 01000ms  SCall: 00033  DCall: 00009 [90.195.139.24:4569]
>  FORMAT          : 4
>
>   -- Call accepted by 90.195.139.24 (format ulaw)
>   -- Format for call is ulaw
> Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX     Subclass: ACK
>  Timestamp: 01000ms  SCall: 00009  DCall: 00033 [90.195.139.24:4569]
> Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: CONTROL Subclass:
> RINGING
>  Timestamp: 01000ms  SCall: 00033  DCall: 00009 [90.195.139.24:4569]
> Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX     Subclass: ACK
>  Timestamp: 01000ms  SCall: 00009  DCall: 00033 [90.195.139.24:4569]
>   -- IAX2/stevebarlow-9 is ringing
> Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 002 Type: VOICE   Subclass: 4
>  Timestamp: 01080ms  SCall: 00009  DCall: 00033 [90.195.139.24:4569]
> Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass: ACK
>  Timestamp: 01080ms  SCall: 00033  DCall: 00009 [90.195.139.24:4569]
> Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 002 Type: IAX     Subclass:
> REJECT
>  Timestamp: 01003ms  SCall: 00033  DCall: 00009 [90.195.139.24:4569]
>  CAUSE           : Call rejected manually.
>
> Dec  1 21:46:53 WARNING[2135]: chan_iax2.c:7100 socket_read: Call
> rejected by 90.195.139.24: Call rejected manually.
> Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 003 Type: IAX     Subclass: ACK
>  Timestamp: 01003ms  SCall: 00009  DCall: 00033 [90.195.139.24:4569]
>   -- Hungup 'IAX2/stevebarlow-9'
> == Everyone is busy/congested at this time (1:0/0/1)
>   -- Executing Macro("Zap/1-1", "fastbusy") in new stack
>   -- Executing Answer("Zap/1-1", "") in new stack
>   -- Executing Wait("Zap/1-1", "1") in new stack
>   -- Executing PlayTones("Zap/1-1", "!950/330|!1400/330|!1800/330|0")
> in new stack
>   -- Executing Wait("Zap/1-1", "1") in new stack
>   -- Executing Playback("Zap/1-1", "im-sorry") in new stack
>   -- Playing 'im-sorry' (language 'en')
>   -- Executing Playback("Zap/1-1", "all-circuits-busy-now") in new stack
>   -- Playing 'all-circuits-busy-now' (language 'en')
>   -- Executing Playback("Zap/1-1", "pls-try-call-later") in new
> stack
>
>
>
> _______________________________________________
> 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: 7.1.409 / Virus Database: 268.15.3/561 - Release Date: 01/12/2006
>
> 



From pdwills at cedarknolltelephone.com  Fri Dec  1 17:20:09 2006
From: pdwills at cedarknolltelephone.com (Paul Wills)
Date: Fri, 01 Dec 2006 18:20:09 -0500
Subject: [VoIP] New Question
References: <16157932.7901481164987519936.JavaMail.root@vms061.mailsrvcs.net>
	<45705404.6000309@btinternet.com>
Message-ID: <004801c7159f$418c0ff0$0301a8c0@Main>

I just tried turning the echotraining off (along with all the echo 
"treatments.")

Does anyone know if an Asterisk "reload" will grab the settings or should I 
reboot the computer?

PDW


----- Original Message ----- 
From: "Jonathan Kay" <g4vft at btinternet.com>
To: <pdwills at cedarknolltelephone.com>; "Voice Over IP Tandem for Analog 
Switches" <voip at ckts.info>
Sent: Friday, December 01, 2006 11:10 AM
Subject: Re: [VoIP] New Question


> Paul Wills wrote:
>> Here's a question:
>>
>> One of my incoming CNET lines goes to an Automatic Electric code call. 
>> The "confirmation" that a page is taking place consists of an audible 
>> pattern of buzzes being fed back to the caller.  Unfortunately, the 
>> Asterisk seems to clip out any sudden tone or sound (It also happens when 
>> one calls my Northeast Electronics milliwatt source).  This causes the 
>> caller to only hear a very short burst of tone.
>>
>> Is this the echo cancelling at work?
>>
>> Is there a way to remove the tone break without screwing up the echo 
>> cancelling too much?
>>
> Hi Paul,
> You could try *echotraining=no*
> in your zapata.conf, just against this relevant zap channel entry.
> The Echo training, mutes the channel, while it sends a tone pulse to
> train the echo canceller.
> The echo canceller should still train itself, but just take longer.
>
>
> Jon
> 



From ka2wft at arrl.net  Fri Dec  1 17:23:19 2006
From: ka2wft at arrl.net (Doug Alderdice)
Date: Fri, 01 Dec 2006 18:23:19 -0500
Subject: [VoIP] New Question
In-Reply-To: <004801c7159f$418c0ff0$0301a8c0@Main>
References: <16157932.7901481164987519936.JavaMail.root@vms061.mailsrvcs.net>
	<45705404.6000309@btinternet.com>
Message-ID: <5.1.0.14.0.20061201182152.02754d28@incoming.verizon.net>

At 06:20 PM 12/1/2006 -0500, Paul Wills wrote:
>I just tried turning the echotraining off (along with all the echo
>"treatments.")
>
>Does anyone know if an Asterisk "reload" will grab the settings or should I
>reboot the computer?

Reload gets some settings, but not all, and I am never which.  A more sure 
fire way is to do a "restart now" in Asterisk... providing you have no 
calls currently up, as a restart will terminate anything in progress.

Doug.




>----- Original Message -----
>From: "Jonathan Kay" <g4vft at btinternet.com>
>To: <pdwills at cedarknolltelephone.com>; "Voice Over IP Tandem for Analog
>Switches" <voip at ckts.info>
>Sent: Friday, December 01, 2006 11:10 AM
>Subject: Re: [VoIP] New Question
>
>
> > Paul Wills wrote:
> >> Here's a question:
> >>
> >> One of my incoming CNET lines goes to an Automatic Electric code call.
> >> The "confirmation" that a page is taking place consists of an audible
> >> pattern of buzzes being fed back to the caller.  Unfortunately, the
> >> Asterisk seems to clip out any sudden tone or sound (It also happens when
> >> one calls my Northeast Electronics milliwatt source).  This causes the
> >> caller to only hear a very short burst of tone.
> >>
> >> Is this the echo cancelling at work?
> >>
> >> Is there a way to remove the tone break without screwing up the echo
> >> cancelling too much?
> >>
> > Hi Paul,
> > You could try *echotraining=no*
> > in your zapata.conf, just against this relevant zap channel entry.
> > The Echo training, mutes the channel, while it sends a tone pulse to
> > train the echo canceller.
> > The echo canceller should still train itself, but just take longer.
> >
> >
> > Jon




From wepbx at sbcglobal.net  Fri Dec  1 17:40:41 2006
From: wepbx at sbcglobal.net (Richard Walsh)
Date: Fri, 1 Dec 2006 15:40:41 -0800 (PST)
Subject: [VoIP] * directory
In-Reply-To: <44DB8F73.8020609@stromberg-carlson.org>
Message-ID: <20061201234041.31778.qmail@web82012.mail.mud.yahoo.com>



Hi John -
   
  I was able to finally get registered with Doug's help - 
  I have now  put up my two working "267" exchange lines.
   
  Question - 
   
  Do you know how to insert the blank lines (rows) between listings?
   
  I wanted to separate the start end end of the 267 entries with a blank.
   
  Thanks
   
  Rick
   
  Obviously I need to spend time and wire up more lines!!!
   

From andrew.e.green at gmail.com  Fri Dec  1 17:46:18 2006
From: andrew.e.green at gmail.com (Andrew Green)
Date: Fri, 1 Dec 2006 20:16:18 -0330
Subject: [VoIP] New Question
In-Reply-To: <5.1.0.14.0.20061201182152.02754d28@incoming.verizon.net>
References: <16157932.7901481164987519936.JavaMail.root@vms061.mailsrvcs.net>
	<45705404.6000309@btinternet.com>
	<004801c7159f$418c0ff0$0301a8c0@Main>
	<5.1.0.14.0.20061201182152.02754d28@incoming.verizon.net>
Message-ID: <c30e256a0612011546x260bead6xd2d99f04a6395909@mail.gmail.com>

You can always do 'restart gracefully' to get Asterisk to wait for any
calls to finish before restarting.


On 12/1/06, Doug Alderdice <ka2wft at arrl.net> wrote:
> At 06:20 PM 12/1/2006 -0500, Paul Wills wrote:
> >I just tried turning the echotraining off (along with all the echo
> >"treatments.")
> >
> >Does anyone know if an Asterisk "reload" will grab the settings or should I
> >reboot the computer?
>
> Reload gets some settings, but not all, and I am never which.  A more sure
> fire way is to do a "restart now" in Asterisk... providing you have no
> calls currently up, as a restart will terminate anything in progress.
>
> Doug.
>
>
>
>
> >----- Original Message -----
> >From: "Jonathan Kay" <g4vft at btinternet.com>
> >To: <pdwills at cedarknolltelephone.com>; "Voice Over IP Tandem for Analog
> >Switches" <voip at ckts.info>
> >Sent: Friday, December 01, 2006 11:10 AM
> >Subject: Re: [VoIP] New Question
> >
> >
> > > Paul Wills wrote:
> > >> Here's a question:
> > >>
> > >> One of my incoming CNET lines goes to an Automatic Electric code call.
> > >> The "confirmation" that a page is taking place consists of an audible
> > >> pattern of buzzes being fed back to the caller.  Unfortunately, the
> > >> Asterisk seems to clip out any sudden tone or sound (It also happens when
> > >> one calls my Northeast Electronics milliwatt source).  This causes the
> > >> caller to only hear a very short burst of tone.
> > >>
> > >> Is this the echo cancelling at work?
> > >>
> > >> Is there a way to remove the tone break without screwing up the echo
> > >> cancelling too much?
> > >>
> > > Hi Paul,
> > > You could try *echotraining=no*
> > > in your zapata.conf, just against this relevant zap channel entry.
> > > The Echo training, mutes the channel, while it sends a tone pulse to
> > > train the echo canceller.
> > > The echo canceller should still train itself, but just take longer.
> > >
> > >
> > > Jon
>
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>


From ratguy at bellsouth.net  Fri Dec  1 18:08:40 2006
From: ratguy at bellsouth.net (Jayson Smith)
Date: Fri, 1 Dec 2006 19:08:40 -0500
Subject: [VoIP] FWD SIP question
Message-ID: <000601c715a6$05a36fe0$0600a8c0@bluegrasspals.com>

Hi,
For a while now I have been having trouble with Ipkall.  Calls would never
reach me.  Now I just signed up with Sipnumber, thinking Ipkall was at
fault, and set my forwarding to XXXXXX at fwd.pulver.com, where XXXXXX is my
FWD number.  Sipnumber can't reach me either, or am I doing something wrong?
Thanks for any help.
Jayson.



From jnovack at stromberg-carlson.org  Fri Dec  1 19:29:19 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Fri, 01 Dec 2006 20:29:19 -0500
Subject: [VoIP] New Question
In-Reply-To: <004801c7159f$418c0ff0$0301a8c0@Main>
References: <16157932.7901481164987519936.JavaMail.root@vms061.mailsrvcs.net>	<45705404.6000309@btinternet.com>
	<004801c7159f$418c0ff0$0301a8c0@Main>
Message-ID: <4570D6EF.3020706@stromberg-carlson.org>

If you made a change in the Zaptel.conf, you need to remove then reload 
modprobe and rerun ztcfg
All other changes, AFAIK, in the late Asterisk,  1.2.13, reload will 
work, except for Musiconhold and perhaps a few others.
Long ago I resorted to "stop now" and asterisk -c, given I don't have a 
high volume of traffic.

Did you examine the battery reversal issue mentioned previously?

John Novack
 

Paul Wills wrote:
> I just tried turning the echotraining off (along with all the echo 
> "treatments.")
>
> Does anyone know if an Asterisk "reload" will grab the settings or should I 
> reboot the computer?
>
> PDW
>
>
> ----- Original Message ----- 
> From: "Jonathan Kay" <g4vft at btinternet.com>
> To: <pdwills at cedarknolltelephone.com>; "Voice Over IP Tandem for Analog 
> Switches" <voip at ckts.info>
> Sent: Friday, December 01, 2006 11:10 AM
> Subject: Re: [VoIP] New Question
>
>
>   
>> Paul Wills wrote:
>>     
>>> Here's a question:
>>>
>>> One of my incoming CNET lines goes to an Automatic Electric code call. 
>>> The "confirmation" that a page is taking place consists of an audible 
>>> pattern of buzzes being fed back to the caller.  Unfortunately, the 
>>> Asterisk seems to clip out any sudden tone or sound (It also happens when 
>>> one calls my Northeast Electronics milliwatt source).  This causes the 
>>> caller to only hear a very short burst of tone.
>>>
>>> Is this the echo cancelling at work?
>>>
>>> Is there a way to remove the tone break without screwing up the echo 
>>> cancelling too much?
>>>
>>>       
>> Hi Paul,
>> You could try *echotraining=no*
>> in your zapata.conf, just against this relevant zap channel entry.
>> The Echo training, mutes the channel, while it sends a tone pulse to
>> train the echo canceller.
>> The echo canceller should still train itself, but just take longer.
>>
>>
>> Jon
>>
>>     
>
>
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>
>   

From pdwills at cedarknolltelephone.com  Fri Dec  1 20:11:51 2006
From: pdwills at cedarknolltelephone.com (Paul Wills)
Date: Fri, 01 Dec 2006 21:11:51 -0500
Subject: [VoIP] New Question
References: <16157932.7901481164987519936.JavaMail.root@vms061.mailsrvcs.net>
	<45705404.6000309@btinternet.com>	<004801c7159f$418c0ff0$0301a8c0@Main>
	<4570D6EF.3020706@stromberg-carlson.org>
Message-ID: <00a401c715b7$3d993e00$0301a8c0@Main>

That sounds reasonable.  I'll give the modprobe and ztcfg a try.

To answer the second question, there is definitely no battery reversal or 
break on the code call unit.  It's strictly coupled tone.

PDW

----- Original Message ----- 
From: "John Novack" <jnovack at stromberg-carlson.org>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Friday, December 01, 2006 8:29 PM
Subject: Re: [VoIP] New Question


> If you made a change in the Zaptel.conf, you need to remove then reload
> modprobe and rerun ztcfg
> All other changes, AFAIK, in the late Asterisk,  1.2.13, reload will
> work, except for Musiconhold and perhaps a few others.
> Long ago I resorted to "stop now" and asterisk -c, given I don't have a
> high volume of traffic.
>
> Did you examine the battery reversal issue mentioned previously?
>
> John Novack
>
>
> Paul Wills wrote:
>> I just tried turning the echotraining off (along with all the echo
>> "treatments.")
>>
>> Does anyone know if an Asterisk "reload" will grab the settings or should 
>> I
>> reboot the computer?
>>
>> PDW
>>
>> 



From ikj1234i at yahoo.com  Fri Dec  1 21:45:58 2006
From: ikj1234i at yahoo.com (ikjtel)
Date: Fri, 1 Dec 2006 19:45:58 -0800 (PST)
Subject: [VoIP] New Question
In-Reply-To: <004801c7159f$418c0ff0$0301a8c0@Main>
Message-ID: <706471.95118.qm@web51603.mail.yahoo.com>


Paul

I'm still skeptical that the echo cancellation stuff
has anything to do with the audio problem...

First, the audio in question wouldn't get suppressed
in the first place, because it's flowing in the
opposite direction from which the canceller is in
effect!

Second, cancellation is rarely perfect, yet you
reported "silence", not "attenuated"...

Third, cancellation would be all-or-nothing and
wouldn't selectively affect only the "first syllable"
like that...

How hard would it be to try lowering the tone level?

Max


 
____________________________________________________________________________________
Want to start your own business?
Learn how on Yahoo! Small Business.
http://smallbusiness.yahoo.com/r-index


From radiorapido at yahoo.co.uk  Sat Dec  2 07:08:26 2006
From: radiorapido at yahoo.co.uk (Duncan Hill)
Date: Sat, 2 Dec 2006 13:08:26 +0000 (GMT)
Subject: [VoIP] New member
In-Reply-To: <019401c7155d$8fbe1750$0b01a8c0@acer1dd0bbc6d0>
Message-ID: <663214.93301.qm@web27201.mail.ukl.yahoo.com>

Hi Ian,

many thanks for your warm welcome!

I've already got myself a couple of Mitel Euroroutes
with connection boxes (for LD to DTMF and v.v.), and
I'm sure I'll be struggling to work it all out once
I'm faced with the task of connecting it all up!

Just reserved my office code 0992, notice that Peter
Walker (from Oftel?) just down the road in Cuffley
also seems to be on CNET!

Anyway, Ian, I hope to be able to dial into your (and
everyone else's) analogue exchanges without using the
gateway number in a couple of weeks time!

-regards
Duncan Hill


--- Ian Jolly <ian at uax.org.uk> wrote:

> Welcome Duncan
> 
> >From here in North Wales. Running an Asterisk box
> which feeds into a digital 
> PABX used purely for LD/DTMF conversion and linking
> various exchanges to the 
> Asterisk -  no BT lines connected to it. I have
> currently two 
> electro-mechanical exchanges connected to it - a
> former GPO UAX 5 dating 
> from 1929 that served a village in Northumberland
> until circa 1950 plus a 
> small cross bar exchange.  Other UAXs from my
> collection will eventually get 
> linked to it when I get them sorted out.
> 
> Good to have you on-line.  You've only got to ask
> for help - some of us are 
> still regrowing our finger nails after the steep
> climb.
> 
> Regards
> 
> Ian Jolly
> 
> 
> +44 (0)352 85 26 (via a 1929 GPO Rural Automatic
> eXchange!)
> CNET - the Heritage Telephone Network

Send instant messages to your online friends http://uk.messenger.yahoo.com 


From pdwills at cedarknolltelephone.com  Sat Dec  2 07:03:11 2006
From: pdwills at cedarknolltelephone.com (Paul Wills)
Date: Sat, 02 Dec 2006 08:03:11 -0500
Subject: [VoIP] New Question
References: <706471.95118.qm@web51603.mail.yahoo.com>
Message-ID: <004601c71614$9675d490$0301a8c0@Main>

Max, et al;

The tone level *sounds* low but I'm starting to suspect that there are some 
high frequency harmonics that are messing things up.  (After all, the tone 
source is a buzzer.)  I am going to play with some filtering later today to 
see if that helps.

PDW


----- Original Message ----- 
From: "ikjtel" <ikj1234i at yahoo.com>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>; "Paul 
Wills" <pdwills at cedarknolltelephone.com>
Sent: Friday, December 01, 2006 10:45 PM
Subject: Re: [VoIP] New Question


>
> Paul
>
> I'm still skeptical that the echo cancellation stuff
> has anything to do with the audio problem...
>
> First, the audio in question wouldn't get suppressed
> in the first place, because it's flowing in the
> opposite direction from which the canceller is in
> effect!
>
> Second, cancellation is rarely perfect, yet you
> reported "silence", not "attenuated"...
>
> Third, cancellation would be all-or-nothing and
> wouldn't selectively affect only the "first syllable"
> like that...
>
> How hard would it be to try lowering the tone level?
>
> Max
>
>
>
> ____________________________________________________________________________________
> Want to start your own business?
> Learn how on Yahoo! Small Business.
> http://smallbusiness.yahoo.com/r-index
> 



From pdwills at cedarknolltelephone.com  Sat Dec  2 13:04:49 2006
From: pdwills at cedarknolltelephone.com (Paul Wills)
Date: Sat, 02 Dec 2006 14:04:49 -0500
Subject: [VoIP] New Question - Problem Solved
References: <706471.95118.qm@web51603.mail.yahoo.com>
	<004601c71614$9675d490$0301a8c0@Main>
Message-ID: <004601c71644$c0a89790$0301a8c0@Main>

I didn't make the "fix" permanent yet but I found two problems.

Putting an inductor in series with the tone source filtered out the high 
frequency stuff that was preventing the FXO card from working properly.  I 
was still getting a "break" in the middle of the tone burst though.

That turned out to be caused by an inductive "kick" caused by the chime 
relay opening.  Although there is no direct path to the audio circuit, I can 
see it coupling through the cable plant or power feed supplying the code 
call unit.  A little Western Electric RC network did the job there.  All is 
well.

I will permanently install the components tonight.  In the mean time, 
there's always clip leads.

Sheesh!  You would think that Automatic Electric would anticipate such 
problems when they designed their stuff back in the 1920's.  ;-)

PDW

----- Original Message ----- 
From: "Paul Wills" <pdwills at cedarknolltelephone.com>
To: "ikjtel" <ikj1234i at yahoo.com>; "Voice Over IP Tandem for Analog 
Switches" <voip at ckts.info>
Sent: Saturday, December 02, 2006 8:03 AM
Subject: Re: [VoIP] New Question


> Max, et al;
>
> The tone level *sounds* low but I'm starting to suspect that there are 
> some
> high frequency harmonics that are messing things up.  (After all, the tone
> source is a buzzer.)  I am going to play with some filtering later today 
> to
> see if that helps.
>
> PDW
>
>
> ----- Original Message ----- 
> From: "ikjtel" <ikj1234i at yahoo.com>
> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>; "Paul
> Wills" <pdwills at cedarknolltelephone.com>
> Sent: Friday, December 01, 2006 10:45 PM
> Subject: Re: [VoIP] New Question
>
>
>>
>> Paul
>>
>> I'm still skeptical that the echo cancellation stuff
>> has anything to do with the audio problem...
>>
>> First, the audio in question wouldn't get suppressed
>> in the first place, because it's flowing in the
>> opposite direction from which the canceller is in
>> effect!
>>
>> Second, cancellation is rarely perfect, yet you
>> reported "silence", not "attenuated"...
>>
>> Third, cancellation would be all-or-nothing and
>> wouldn't selectively affect only the "first syllable"
>> like that...
>>
>> How hard would it be to try lowering the tone level?
>>
>> Max
>>
>>



From ikj1234i at yahoo.com  Sun Dec  3 10:36:35 2006
From: ikj1234i at yahoo.com (ikjtel)
Date: Sun, 3 Dec 2006 08:36:35 -0800 (PST)
Subject: [VoIP] New Question - Problem Solved
In-Reply-To: <004601c71644$c0a89790$0301a8c0@Main>
Message-ID: <20061203163635.72662.qmail@web51612.mail.yahoo.com>


Paul

As you of all people would know, this isn't the first
instance where this class of FXO hardware has failed
to handle signals that it should handle properly... 
Probably not even the second...

Max

--- Paul Wills <pdwills at cedarknolltelephone.com>
wrote:

> I didn't make the "fix" permanent yet but I found
> two problems.
> 
> Putting an inductor in series with the tone source
> filtered out the high 
> frequency stuff that was preventing the FXO card
> from working properly.  I 
> was still getting a "break" in the middle of the
> tone burst though.
> 
> That turned out to be caused by an inductive "kick"
> caused by the chime 
> relay opening.  Although there is no direct path to
> the audio circuit, I can 
> see it coupling through the cable plant or power
> feed supplying the code 
> call unit.  A little Western Electric RC network did
> the job there.  All is 
> well.
> 
> I will permanently install the components tonight. 
> In the mean time, 
> there's always clip leads.
> 
> Sheesh!  You would think that Automatic Electric
> would anticipate such 
> problems when they designed their stuff back in the
> 1920's.  ;-)
> 
> PDW
> 
> ----- Original Message ----- 
> From: "Paul Wills" <pdwills at cedarknolltelephone.com>
> To: "ikjtel" <ikj1234i at yahoo.com>; "Voice Over IP
> Tandem for Analog 
> Switches" <voip at ckts.info>
> Sent: Saturday, December 02, 2006 8:03 AM
> Subject: Re: [VoIP] New Question
> 
> 
> > Max, et al;
> >
> > The tone level *sounds* low but I'm starting to
> suspect that there are 
> > some
> > high frequency harmonics that are messing things
> up.  (After all, the tone
> > source is a buzzer.)  I am going to play with some
> filtering later today 
> > to
> > see if that helps.
> >
> > PDW
> >
> >
> > ----- Original Message ----- 
> > From: "ikjtel" <ikj1234i at yahoo.com>
> > To: "Voice Over IP Tandem for Analog Switches"
> <voip at ckts.info>; "Paul
> > Wills" <pdwills at cedarknolltelephone.com>
> > Sent: Friday, December 01, 2006 10:45 PM
> > Subject: Re: [VoIP] New Question
> >
> >
> >>
> >> Paul
> >>
> >> I'm still skeptical that the echo cancellation
> stuff
> >> has anything to do with the audio problem...
> >>
> >> First, the audio in question wouldn't get
> suppressed
> >> in the first place, because it's flowing in the
> >> opposite direction from which the canceller is in
> >> effect!
> >>
> >> Second, cancellation is rarely perfect, yet you
> >> reported "silence", not "attenuated"...
> >>
> >> Third, cancellation would be all-or-nothing and
> >> wouldn't selectively affect only the "first
> syllable"
> >> like that...
> >>
> >> How hard would it be to try lowering the tone
> level?
> >>
> >> Max
> >>
> >>
> 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 



 
____________________________________________________________________________________
Want to start your own business?
Learn how on Yahoo! Small Business.
http://smallbusiness.yahoo.com/r-index


From ikj1234i at yahoo.com  Sun Dec  3 10:41:55 2006
From: ikj1234i at yahoo.com (ikjtel)
Date: Sun, 3 Dec 2006 08:41:55 -0800 (PST)
Subject: [VoIP] New Question - Problem Solved
In-Reply-To: <004601c71644$c0a89790$0301a8c0@Main>
Message-ID: <586794.4105.qm@web51606.mail.yahoo.com>


Hey Paul

What EXACT type of FXO hardware are you using? 
X100P/clone/TDM400P or what ?

Max


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


From pdwills at cedarknolltelephone.com  Sun Dec  3 12:16:00 2006
From: pdwills at cedarknolltelephone.com (Paul Wills)
Date: Sun, 03 Dec 2006 13:16:00 -0500
Subject: [VoIP] New Question - Problem Solved
References: <586794.4105.qm@web51606.mail.yahoo.com>
Message-ID: <001f01c71707$19d444a0$0301a8c0@Main>

I'm using an X100P "clone."

I also have a TDM400 but the FXO card was taken out by an inductive kick and 
has been returned to Digium.  I don't remember trying the FXO module with 
the code call and am a bit afraid to try until I provide some hefty surge 
protection.

My hope is to maintain two paths into the EM switch from the Asterisk box. 
The clone card certainly holds up to any "abuse" that it receives from the 
inductive components.  A Google search indicates that the Digium modules are 
a lot more fragile.

PDW

----- Original Message ----- 
From: "ikjtel" <ikj1234i at yahoo.com>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>; "Paul 
Wills" <pdwills at cedarknolltelephone.com>
Sent: Sunday, December 03, 2006 11:41 AM
Subject: Re: [VoIP] New Question - Problem Solved


>
> Hey Paul
>
> What EXACT type of FXO hardware are you using?
> X100P/clone/TDM400P or what ?
>
> Max
>
> 


From wepbx at sbcglobal.net  Sun Dec  3 14:39:32 2006
From: wepbx at sbcglobal.net (Richard Walsh)
Date: Sun, 3 Dec 2006 12:39:32 -0800 (PST)
Subject: [VoIP] WE inductor needed
In-Reply-To: <001f01c71707$19d444a0$0301a8c0@Main>
Message-ID: <162558.63729.qm@web82003.mail.mud.yahoo.com>


One of the electromechanical switch terminations I am working on for my SXS switch is a WE 24B loop check generator (1000 to 3000 Hz sweep).  
  The unit is complete except for one adjustable inductor - the part is L6 on the unit.
   
  Does anyone have a schematic that they could identify the part and the inductive range? 
  The 24B LP CHK GEN is SD99707-01    J94024B-1
   
  The needed inductor is like a small black hockey puck - perhaps 1 " high and  1-1/2" in diameter.  I believe the part number is 1059x     (x= letter code).  The 24B unit I have dates from 1962.
   
  Might someone have the part, or suggest an alternative?
   
  Many thanks,
   
  Rick Walsh

From stfkerman at jps.net  Sun Dec  3 14:43:31 2006
From: stfkerman at jps.net (Steph Kerman)
Date: Sun, 03 Dec 2006 15:43:31 -0500
Subject: [VoIP] WE inductor needed
In-Reply-To: <162558.63729.qm@web82003.mail.mud.yahoo.com>
References: <162558.63729.qm@web82003.mail.mud.yahoo.com>
Message-ID: <457336F3.7010000@jps.net>

I definitely have the schematic.  I'm not 100% sure where.

Steph

Richard Walsh wrote:
> One of the electromechanical switch terminations I am working on for my SXS switch is a WE 24B loop check generator (1000 to 3000 Hz sweep).  
>   The unit is complete except for one adjustable inductor - the part is L6 on the unit.
>    
>   Does anyone have a schematic that they could identify the part and the inductive range? 
>   The 24B LP CHK GEN is SD99707-01    J94024B-1
>    
>   The needed inductor is like a small black hockey puck - perhaps 1 " high and  1-1/2" in diameter.  I believe the part number is 1059x     (x= letter code).  The 24B unit I have dates from 1962.
>    
>   Might someone have the part, or suggest an alternative?
>    
>   Many thanks,
>    
>   Rick Walsh
>   


From pdwills at cedarknolltelephone.com  Sun Dec  3 16:02:26 2006
From: pdwills at cedarknolltelephone.com (Paul Wills)
Date: Sun, 03 Dec 2006 17:02:26 -0500
Subject: [VoIP] WE inductor needed
References: <162558.63729.qm@web82003.mail.mud.yahoo.com>
Message-ID: <001f01c71726$bad9a3d0$0301a8c0@Main>

Rick,

Do you need the SD and CD.  They are of a size that I can scan to PDF quite 
easily.

The info for L6 is 15098E  or 1509BE  20 mH  Looking at the others, I 
suspect that "BE" is correct.

PDW

----- Original Message ----- 
From: "Richard Walsh" <wepbx at sbcglobal.net>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Sunday, December 03, 2006 3:39 PM
Subject: [VoIP] WE inductor needed


>
> One of the electromechanical switch terminations I am working on for my 
> SXS switch is a WE 24B loop check generator (1000 to 3000 Hz sweep).
>  The unit is complete except for one adjustable inductor - the part is L6 
> on the unit.
>
>  Does anyone have a schematic that they could identify the part and the 
> inductive range?
>  The 24B LP CHK GEN is SD99707-01    J94024B-1
>
>  The needed inductor is like a small black hockey puck - perhaps 1 " high 
> and  1-1/2" in diameter.  I believe the part number is 1059x     (x= 
> letter code).  The 24B unit I have dates from 1962.
>
>  Might someone have the part, or suggest an alternative?
>
>  Many thanks,
>
>  Rick Walsh
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
> 



From wepbx at sbcglobal.net  Sun Dec  3 18:20:46 2006
From: wepbx at sbcglobal.net (Richard Walsh)
Date: Sun, 3 Dec 2006 16:20:46 -0800 (PST)
Subject: [VoIP] WE inductor needed
In-Reply-To: <162558.63729.qm@web82003.mail.mud.yahoo.com>
Message-ID: <20061204002046.21583.qmail@web82012.mail.mud.yahoo.com>

Replying to my own message to report that Paul Wills  has identified the needed L6 inductor for the 24B loop check generator as a WE part 1059BE with a value of 20MH (adjustable).
   
  If anyone has such a part I would be apprecitive.
   
  Rick Walsh

Richard Walsh <wepbx at sbcglobal.net> wrote:
  
One of the electromechanical switch terminations I am working on for my SXS switch is a WE 24B loop check generator (1000 to 3000 Hz sweep). 
The unit is complete except for one adjustable inductor - the part is L6 on the unit.

Does anyone have a schematic that they could identify the part and the inductive range? 
The 24B LP CHK GEN is SD99707-01 J94024B-1

The needed inductor is like a small black hockey puck - perhaps 1 " high and 1-1/2" in diameter. I believe the part number is 1059x (x= letter code). The 24B unit I have dates from 1962.

Might someone have the part, or suggest an alternative?

Many thanks,

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



From wepbx at sbcglobal.net  Sun Dec  3 18:14:47 2006
From: wepbx at sbcglobal.net (Richard Walsh)
Date: Sun, 3 Dec 2006 16:14:47 -0800 (PST)
Subject: [VoIP] WE inductor needed
In-Reply-To: <001f01c71726$bad9a3d0$0301a8c0@Main>
Message-ID: <135445.58062.qm@web82005.mail.mud.yahoo.com>

Paul -
  Thank you very much for identifying the part -  I appreciate the value and P/N.
   
  I do not need the print at the present time, in fact, I may have it in my collection but am unable to presently get my hands on it.
   
  Thanks,
   
  Rick

Paul Wills <pdwills at cedarknolltelephone.com> wrote:
  Rick,

Do you need the SD and CD. They are of a size that I can scan to PDF quite 
easily.

The info for L6 is 15098E or 1509BE 20 mH Looking at the others, I 
suspect that "BE" is correct.

PDW

----- Original Message ----- 
From: "Richard Walsh" 
To: "Voice Over IP Tandem for Analog Switches" 
Sent: Sunday, December 03, 2006 3:39 PM
Subject: [VoIP] WE inductor needed


>
> One of the electromechanical switch terminations I am working on for my 
> SXS switch is a WE 24B loop check generator (1000 to 3000 Hz sweep).
> The unit is complete except for one adjustable inductor - the part is L6 
> on the unit.
>
> Does anyone have a schematic that they could identify the part and the 
> inductive range?
> The 24B LP CHK GEN is SD99707-01 J94024B-1
>
> The needed inductor is like a small black hockey puck - perhaps 1 " high 
> and 1-1/2" in diameter. I believe the part number is 1059x (x= 
> letter code). The 24B unit I have dates from 1962.
>
> Might someone have the part, or suggest an alternative?
>
> Many thanks,
>
> Rick Walsh
> _______________________________________________
> 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 wepbx at sbcglobal.net  Sun Dec  3 18:46:08 2006
From: wepbx at sbcglobal.net (Richard Walsh)
Date: Sun, 3 Dec 2006 16:46:08 -0800 (PST)
Subject: [VoIP] WE inductor needed
In-Reply-To: <20061204002046.21583.qmail@web82012.mail.mud.yahoo.com>
Message-ID: <193627.51536.qm@web82011.mail.mud.yahoo.com>

Make that We P/N 1509BE  Ferrite inductor.
   
  Rick

Richard Walsh <wepbx at sbcglobal.net> wrote:
  Replying to my own message to report that Paul Wills has identified the needed L6 inductor for the 24B loop check generator as a WE part 1059BE with a value of 20MH (adjustable).

If anyone has such a part I would be apprecitive.

Rick Walsh

Richard Walsh wrote:

One of the electromechanical switch terminations I am working on for my SXS switch is a WE 24B loop check generator (1000 to 3000 Hz sweep). 
The unit is complete except for one adjustable inductor - the part is L6 on the unit.

Does anyone have a schematic that they could identify the part and the inductive range? 
The 24B LP CHK GEN is SD99707-01 J94024B-1

The needed inductor is like a small black hockey puck - perhaps 1 " high and 1-1/2" in diameter. I believe the part number is 1059x (x= letter code). The 24B unit I have dates from 1962.

Might someone have the part, or suggest an alternative?

Many thanks,

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


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



From stfkerman at jps.net  Sun Dec  3 20:11:53 2006
From: stfkerman at jps.net (Steph Kerman)
Date: Sun, 03 Dec 2006 21:11:53 -0500
Subject: [VoIP] Submitted for your consideration:  Fire in telephone cord
Message-ID: <457383E9.70604@jps.net>

Here's something to consider that someone slipped in over my email transom:

http://www.yomiuri.co.jp/dy/national/20061203TDY02004.htm

It reminds me of a thread that came through some of these lists a year or two ago about someone 
using a textile insulated tinsel line cord to provide 120VAC power to a lamp or something.  The 
exact details escape me.



From chad at maine.maine.edu  Mon Dec  4 17:34:39 2006
From: chad at maine.maine.edu (Chad Perkins)
Date: Mon, 04 Dec 2006 18:34:39 -0500
Subject: [VoIP] Question
In-Reply-To: <20061128040339.56309.qmail@web82007.mail.mud.yahoo.com>
Message-ID: <45746A3F.12585.2514C4@localhost>

There is something fubar with www.ckts.info/secure/myswitches.php too.  I am 
unable to create a tandem (anything but an End Office).

Chad

> I have tried repeatedly to add my exchange to the "Member Listings"
> directory but am unable to edit the list because I am unable to log
> in.  I have twice registered, but I am apparently not able to set up a
> new user.
> 
>   Rick Walsh
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 
> 
> 
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.1.409 / Virus Database: 268.15.6/568 - Release Date:
> 12/4/06
> 




From ian at uax.org.uk  Mon Dec  4 17:42:56 2006
From: ian at uax.org.uk (Ian Jolly)
Date: Mon, 4 Dec 2006 23:42:56 -0000
Subject: [VoIP] [-SPAM-] Re:  Question
References: <45746A3F.12585.2514C4@localhost>
Message-ID: <000901c717fd$ecb3f590$0b01a8c0@acer1dd0bbc6d0>

I can get into Members Listings, make the amendments/additions, then enter 
them. It give 'accepted'  but nothing shows

Ian Jolly
+44 (0)352 82 26 (via a 1929 GPO Rural Automatic eXchange!)
CNET


----- Original Message ----- 
From: "Chad Perkins" <chad at maine.maine.edu>
To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
Sent: Monday, December 04, 2006 11:34 PM
Subject: [-SPAM-] Re: [VoIP] Question


> There is something fubar with www.ckts.info/secure/myswitches.php too.  I 
> am
> unable to create a tandem (anything but an End Office).
>
> Chad
>
>> I have tried repeatedly to add my exchange to the "Member Listings"
>> directory but am unable to edit the list because I am unable to log
>> in.  I have twice registered, but I am apparently not able to set up a
>> new user.
>>
>>   Rick Walsh
>> _______________________________________________
>> VoIP mailing list
>> VoIP at ckts.info
>> http://lists.ckts.info/mailman/listinfo/voip
>> Project Web Page: http://www.ckts.info/
>>
>>
>>
>> -- 
>> No virus found in this incoming message.
>> Checked by AVG Free Edition.
>> Version: 7.1.409 / Virus Database: 268.15.6/568 - Release Date:
>> 12/4/06
>>
>
>
>
> _______________________________________________
> 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: 7.1.409 / Virus Database: 268.15.6/567 - Release Date: 04/12/2006
> 



From greg at vyger.net  Mon Dec  4 21:36:43 2006
From: greg at vyger.net (Greg Blakely)
Date: Mon, 4 Dec 2006 21:36:43 -0600
Subject: [VoIP] [-SPAM-] Re:  Question
Message-ID: <A691C3685A152E4AAC9608B8D8E113163EB1AC@godzilla.hickory.vyger.net>

Ian,

Would it be possible for you to send me a screen shot (off list) of what
it is you were doing when you entered the information?  I was able to
add a tandem under my own login, but I also have administrative
privileges, so that;s not really an apples-to-apples comparison.

I readily admit that the whole thing is a kludge, put together by
someone with limited mySQL and PHP expertise.  So, one screen at a time,
we'll iron out the bugs.

I appreciate the heads up.

Greg 

> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] 
> On Behalf Of Ian Jolly
> Sent: Monday, December 04, 2006 5:43 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] [-SPAM-] Re: Question
> 
> I can get into Members Listings, make the 
> amendments/additions, then enter them. It give 'accepted'  
> but nothing shows
> 
> Ian Jolly
> +44 (0)352 82 26 (via a 1929 GPO Rural Automatic eXchange!)
> CNET
> 
> 
> ----- Original Message -----
> From: "Chad Perkins" <chad at maine.maine.edu>
> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> Sent: Monday, December 04, 2006 11:34 PM
> Subject: [-SPAM-] Re: [VoIP] Question
> 
> 
> > There is something fubar with 
> www.ckts.info/secure/myswitches.php too.  I 
> > am
> > unable to create a tandem (anything but an End Office).
> >
> > Chad
> >
> >> I have tried repeatedly to add my exchange to the "Member Listings"
> >> directory but am unable to edit the list because I am unable to log
> >> in.  I have twice registered, but I am apparently not able 
> to set up a
> >> new user.
> >>
> >>   Rick Walsh
> >> _______________________________________________
> >> VoIP mailing list
> >> VoIP at ckts.info
> >> http://lists.ckts.info/mailman/listinfo/voip
> >> Project Web Page: http://www.ckts.info/
> >>
> >>
> >>
> >> -- 
> >> No virus found in this incoming message.
> >> Checked by AVG Free Edition.
> >> Version: 7.1.409 / Virus Database: 268.15.6/568 - Release Date:
> >> 12/4/06
> >>
> >
> >
> >
> > _______________________________________________
> > 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: 7.1.409 / Virus Database: 268.15.6/567 - Release 
> Date: 04/12/2006
> > 
> 
> 
> _______________________________________________
> 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  Mon Dec  4 22:10:39 2006
From: ratguy at bellsouth.net (Jayson Smith)
Date: Mon, 4 Dec 2006 23:10:39 -0500
Subject: [VoIP] [-SPAM-] Re:  Question
References: <A691C3685A152E4AAC9608B8D8E113163EB1AC@godzilla.hickory.vyger.net>
Message-ID: <000e01c71823$52c7db60$0600a8c0@bluegrasspals.com>

Hi,
I know this is barely on-topic, but for any system like that, I always find
it useful to have a test user who doesn't have admin rights, so you can test
out stuff from a real user's perspective.  That is because, as an admin
user, you are God and can do anything and possibly mess up everything, so if
there's a permissions problem, the Admin user is likely to be able to get by
with stuffing the rules into a thin garbage bag, then putting a pickup truck
on top of said bag, the pickup truck being held in the palm of one hand by
the way, whereas a real user or, in this case, a test dummy user has to play
by the rules just like everybody else does.
Jayson.

----- Original Message ----- 
From: "Greg Blakely" <greg at vyger.net>
To: "Ian Jolly" <ian at uax.org.uk>; "Voice Over IP Tandem for Analog Switches"
<voip at ckts.info>
Sent: Monday, December 04, 2006 10:36 PM
Subject: Re: [VoIP] [-SPAM-] Re: Question


> Ian,
>
> Would it be possible for you to send me a screen shot (off list) of what
> it is you were doing when you entered the information?  I was able to
> add a tandem under my own login, but I also have administrative
> privileges, so that;s not really an apples-to-apples comparison.
>
> I readily admit that the whole thing is a kludge, put together by
> someone with limited mySQL and PHP expertise.  So, one screen at a time,
> we'll iron out the bugs.
>
> I appreciate the heads up.
>
> Greg
>
> > -----Original Message-----
> > From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info]
> > On Behalf Of Ian Jolly
> > Sent: Monday, December 04, 2006 5:43 PM
> > To: Voice Over IP Tandem for Analog Switches
> > Subject: Re: [VoIP] [-SPAM-] Re: Question
> >
> > I can get into Members Listings, make the
> > amendments/additions, then enter them. It give 'accepted'
> > but nothing shows
> >
> > Ian Jolly
> > +44 (0)352 82 26 (via a 1929 GPO Rural Automatic eXchange!)
> > CNET
> >
> >
> > ----- Original Message -----
> > From: "Chad Perkins" <chad at maine.maine.edu>
> > To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> > Sent: Monday, December 04, 2006 11:34 PM
> > Subject: [-SPAM-] Re: [VoIP] Question
> >
> >
> > > There is something fubar with
> > www.ckts.info/secure/myswitches.php too.  I
> > > am
> > > unable to create a tandem (anything but an End Office).
> > >
> > > Chad
> > >
> > >> I have tried repeatedly to add my exchange to the "Member Listings"
> > >> directory but am unable to edit the list because I am unable to log
> > >> in.  I have twice registered, but I am apparently not able
> > to set up a
> > >> new user.
> > >>
> > >>   Rick Walsh
> > >> _______________________________________________
> > >> VoIP mailing list
> > >> VoIP at ckts.info
> > >> http://lists.ckts.info/mailman/listinfo/voip
> > >> Project Web Page: http://www.ckts.info/
> > >>
> > >>
> > >>
> > >> -- 
> > >> No virus found in this incoming message.
> > >> Checked by AVG Free Edition.
> > >> Version: 7.1.409 / Virus Database: 268.15.6/568 - Release Date:
> > >> 12/4/06
> > >>
> > >
> > >
> > >
> > > _______________________________________________
> > > 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: 7.1.409 / Virus Database: 268.15.6/567 - Release
> > Date: 04/12/2006
> > >
> >
> >
> > _______________________________________________
> > 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 chad at maine.maine.edu  Mon Dec  4 23:10:33 2006
From: chad at maine.maine.edu (Chad Perkins)
Date: Tue, 05 Dec 2006 00:10:33 -0500
Subject: [VoIP] CLLI Technology and Function Switch Types
Message-ID: <4574B8F9.11545.158A64F@localhost>

If you have any additions, suggestions or comments feel free to contact me 
either on or off list (or via CNET 955-9924).
Chad.

http://maine.maine.edu/~chad/cllitfst.html




From jnovack at stromberg-carlson.org  Wed Dec  6 21:41:58 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Wed, 06 Dec 2006 22:41:58 -0500
Subject: [VoIP] T1 cards.
Message-ID: <45778D86.9030304@stromberg-carlson.org>

Anyone trying to get the Govarion T1 card to co-exist in a machine with 
the TDM400, be advised the tor3 driver needs to be loaded FIRST.
Not sure if the tor2 driver works the same, but I spent many hours today 
going down dead ends until I got it to work

My replacement card also is really sensitive to the brand of plug in 
port #1. I went through several cables until one was solid

John Novack


From Jmda02 at aol.com  Sun Dec 10 23:47:32 2006
From: Jmda02 at aol.com (Jmda02@aol.com)
Date: Mon, 11 Dec 2006 00:47:32 EST
Subject: [VoIP] broken sip
Message-ID: <ccc.5612ea6.32ae4af4@aol.com>

hi greg can you check my ata  . please give me a  number  that i can contact 
you on. 
 

From stfkerman at jps.net  Sun Dec 10 23:55:46 2006
From: stfkerman at jps.net (Steph Kerman)
Date: Mon, 11 Dec 2006 00:55:46 -0500
Subject: [VoIP] Naive question
Message-ID: <457CF2E2.30700@jps.net>

Is there any practical mechanism by which people who subscribe to 
commercial Voip services could be called on a direct Voip basis by CNET 
users, without going out into the PSTN and NANP?

Steph


From bbj at innismir.net  Mon Dec 11 06:44:03 2006
From: bbj at innismir.net (Ben Jackson)
Date: Mon, 11 Dec 2006 07:44:03 -0500
Subject: [VoIP] Naive question
Message-ID: <E1GtkVW-0007bT-FR@telstar.blackratchet.org>

Short answer: No.

Unless they have their own ATA with a spare port that could be configured for someone on CNET (Which would be unlikely if they are using a consumer grade VoIP service, I presume), or if we work out some kind of peering arangement with Vonage/Broadvoice/etc (About as likely as having pigs fly) then the PSTN is a required hop.

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

-----Original Message-----
From: "Steph Kerman" <stfkerman at jps.net>
To: "VoIP" <voip at ckts.info>
Sent: 12/11/06 12:55 AM
Subject: [VoIP] Naive question

Is there any practical mechanism by which people who subscribe to 
commercial Voip services could be called on a direct Voip basis by CNET 
users, without going out into the PSTN and NANP?

Steph

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





From g4vft at btinternet.com  Mon Dec 11 07:39:07 2006
From: g4vft at btinternet.com (Jonathan Kay)
Date: Mon, 11 Dec 2006 13:39:07 +0000
Subject: [VoIP] Naive question
In-Reply-To: <457CF2E2.30700@jps.net>
References: <457CF2E2.30700@jps.net>
Message-ID: <457D5F7B.2060205@btinternet.com>

Steph Kerman wrote:
> Is there any practical mechanism by which people who subscribe to 
> commercial Voip services could be called on a direct Voip basis by CNET 
> users, without going out into the PSTN and NANP?
>
> Steph
>
>   
My CNET asterisk box, is registered to Free World Dialup and Sipgate. 
Mainly for incoming DISA portal use.
I have set up fixed destination routes in the dial plan for testing 
purposes. So that a CNET dialled number, hosted on my box diverts to a 
users Sipgate account number.


For example (this is a fictitious number btw) This will give you an 
incoming service from the CNET. 
Which is what you asked for.
Outgoing, would probably need DISA access to a CNET Asterisk Box.

exten => 447052555,1,Dial(SIP/Sipgate/2345678)
exten => 447052555,2,Macro(fastbusy)


Jon K



From john_reads_cnet_via_archives at covert.org  Mon Dec 11 09:00:01 2006
From: john_reads_cnet_via_archives at covert.org (John R. Covert)
Date: Mon, 11 Dec 2006 10:00:01 -0500 (EST)
Subject: [VoIP] Naive question (calling commercial VoIP subscribers)
In-Reply-To: <457CF2E2.30700@jps.net>
Message-ID: <20061211153913.8432656306@ns1.vyger.net>

Steph asked:

>Is there any practical mechanism by which people who subscribe to 
>commercial Voip services could be called on a direct Voip basis by CNET 
>users, without going out into the PSTN and NANP?

That would depend

  (a) on the provider
or
  (b) on the mechanism desired.

Forgetting about CNET for the moment, there are many commercial VoIP
providers which are more than willing to accept inbound SIP (or even
occasionally IAX) calls from non-subscribers to their subscribers.

One way to figure out whether this is likely is to see if the provider
in question has peering arrangments with Free World Dialup.  If they
do, then obviously they can be called via FWD, but it's actually also
likely that they can be called directly.  You can see who is peered
with FWD by going to http://www.freeworlddialup.com/ and clicking
on the "Peering" tab.  By the way, the second entry in the list of
peering numbers is labelled "ISN Beta".  Using that, you can make
calls to universities and companies all over the world, if they
are participating in ISN numbering.  A list of the participants
is at http://www.iana.org/assignments/trip-parameters

The real problem is knowing whether a particular E.164 number can
be reached by VoIP.  enum.arpa was supposed to solve that, but the
American providers are simply not interested in participating.  If
we all could be as wise as Austria, where when a friend told me that
he had signed up with a VoIP provider, and I looked him up in enum,
we'd be able to simply go to enum.arpa and pick out the VoIP routing
to any number that is routeable that way.

Without enum, the problem is figuring out how to route the call.
Some providers want to be given a full e.164 number (country code
plus number), but others want to see the call in their local
national format.

So in general, if they have a peering arrangement with FWD, just
call them that way.  And btw, when you call via FWD to a commercial
provider, you can send your own PSTN number (it's illegal to send
anything "confusing" and a CNET number would be "confusing").

Steph, if you have a particular provider in mind, I can probably
help you figure out how to make calls.  There are some which are
reachable directly that are not peered with FWD, such as Lingo,
but Lingo requires a special setup in both sip.conf as well as in
extensions.conf.

Now, bringing CNET (not CNET itself, but CNET's members) back into
the picture.  It's possible that a CNET member could provide the
ability to gateway to a VoIP provider to which he has a connection.
For example, I could theoretically allow calls to a Vonage customer
through my system, and if you were to call in via SIP with a
compatible (i.e. ulaw) codec, the call would even switch its audio
to go directly between you and the Vonage gateway (in the case of
NAT) or between you and the Vonage end customer (if NAT doesn't get
into the picture).  The problem with this would be that although
the kind of Vonage service which you can terminate in an Asterisk
(a Vonage Soft Phone Account) allows unlimited calls to other Vonage
subscribers, if the Vonage subscriber ports his number away from
Vonage, suddenly I start running up a phone bill, and have no way
to know that this has happened unless I examine the billing at the
end of each call.

So yes, there are a lot of interconnections out there.  It's almost
like in the good old days of telephony where you could poke around
in the routing and see if someone you had a route to had a route to
somewhere else you might want to call, and keep stacking tandems
until you reached your destination.  Except usually lossless.

/john


From jnovack at stromberg-carlson.org  Mon Dec 11 11:15:45 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Mon, 11 Dec 2006 12:15:45 -0500
Subject: [VoIP] Naive question
In-Reply-To: <457CF2E2.30700@jps.net>
References: <457CF2E2.30700@jps.net>
Message-ID: <457D9241.2090304@stromberg-carlson.org>

Both Gizmo and Stanaphone allow their subscribers to call other like 
subscribers free calls, and I am sure there are MANY others.
MANY more details regarding exactly what you are trying to accomplish 
would be needed, though
I use Stanaphone as a portal from the PSTN to all of CNET. Any other 
Stanaphone subscriber could call my Stanaphone INTERNAL number ( 
different than the PSTN number ) and access CNET. STanaphone chose to 
simplify their billing by having a PSTN number and an internal 
Stanaphone number.
Stanaphone subscribers enjoy 1.6 cents per minute US, and 2.5 cents per 
minute ( not premium ) to the UK - prepaid.

Gizmo is another possibility, though I don't have it set up as a portal, 
someone else might. Gizmo is an attempt at a Skype killer. 1 cent 
outgoing, prepaid, free Gizmo inbound. PSTN inbound  is an additional 
charge. Someone with a Gizmo account could set up an inbound portal to DISA.

All require HSIA of some sort, so why wouldn't anyone interested set up 
their own Asterisk box? Machine requirements are minimal, depending on 
what the goal is. I know of one CNET member that is running an AMD 
K6-233 with a small amount of memory and a 10-15 Gig HD
Those can be had at thrift shops for next to nothing these days.

John Novack


Steph Kerman wrote:
> Is there any practical mechanism by which people who subscribe to 
> commercial Voip services could be called on a direct Voip basis by CNET 
> users, without going out into the PSTN and NANP?
>
> Steph
>
> _______________________________________________
> 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 Dec 11 13:19:13 2006
From: stfkerman at jps.net (Steph Kerman)
Date: Mon, 11 Dec 2006 14:19:13 -0500
Subject: [VoIP] Naive question
In-Reply-To: <457D9241.2090304@stromberg-carlson.org>
References: <457CF2E2.30700@jps.net> <457D9241.2090304@stromberg-carlson.org>
Message-ID: <457DAF31.7070606@jps.net>

PC Geeks don't understand that there are people who prefer to use the PC 
as a tool rather than learning about how to fix Windows when it breaks.  
They're not mechanics and don't work on their cars.  They just drive 
them and perhaps occasionally fix a flat.  Perhaps not even that.

Likewise, there are people who see the phone as an instrument for voice 
communication.  They are not switching geeks, don't own any switching 
equipment and do not WANT to own any switching equipment, but do 
subscribe to a commercial Voip service provider for purely economic 
reasons. 

Such people would have no desire to run and maintain another PC for the 
purpose of CNET connectivity but might be open to at least having 
inbound access from CNET, perhaps mostly for the convenience of their 
CNET friends reaching them.  They might not even much care whether they 
had outbound CNET access.

I am not talking about myself here.  Though I have little appetite for 
monkeying with Windows, I do fix my own car, enjoy playing with 
switching equipment and will set up an * box of my own at the right time 
for me.  I'm exploring the idea as a means to expand the actual utility 
of CNET to something beyond what ham radio is mostly about: an end in 
itself where 90% of the conversation consists of people talking about 
their rigs.

Steph

John Novack wrote:
> All require HSIA of some sort, so why wouldn't anyone interested set 
> up their own Asterisk box? Machine requirements are minimal, depending 
> on what the goal is. I know of one CNET member that is running an AMD 
> K6-233 with a small amount of memory and a 10-15 Gig HD.  Those can be 
> had at thrift shops for next to nothing these days.
>
> John Novack
>
>
> Steph Kerman wrote:
>> Is there any practical mechanism by which people who subscribe to
>> commercial Voip services could be called on a direct Voip basis by CNET
>> users, without going out into the PSTN and NANP?
>>
>> Steph
>   


From jnovack at stromberg-carlson.org  Mon Dec 11 14:01:32 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Mon, 11 Dec 2006 15:01:32 -0500
Subject: [VoIP] Naive question
In-Reply-To: <457DAF31.7070606@jps.net>
References: <457CF2E2.30700@jps.net> <457D9241.2090304@stromberg-carlson.org>
	<457DAF31.7070606@jps.net>
Message-ID: <457DB91C.9070401@stromberg-carlson.org>



Steph Kerman wrote:
>
> I am not talking about myself here.  Though I have little appetite for 
> monkeying with Windows, I do fix my own car, enjoy playing with 
> switching equipment and will set up an * box of my own at the right 
> time for me.  I'm exploring the idea as a means to expand the actual 
> utility of CNET to something beyond what ham radio is mostly about: an 
> end in itself where 90% of the conversation consists of people talking 
> about their rigs.
>
> Steph
Gizmo may be a good choice then
Easy to set up on any Windoze machine with a sound card and mike Free 
between Gizmo users, open SIP protocol and easy for the CNET/Asterisk 
user to set up.
I can't speak for how others use CNET/Asterisk, but most of my 
conversations aren't over my gear, some involves discussions somewhat 
related, such as why this hobby attracts such strange people, but most 
of my usage with other members involves matters completely unrelated to 
telephony.
Some of us have commercial VOIP connections as well, integrated into our 
CNET dialplans
Commercial VOIP services usually are locked in some way to the provider, 
and access to them is, in one way or another through the PSTN scheme, 
even though it may be VOIP from end to end.

John Novack

> John Novack wrote:
>> All require HSIA of some sort, so why wouldn't anyone interested set 
>> up their own Asterisk box? Machine requirements are minimal, 
>> depending on what the goal is. I know of one CNET member that is 
>> running an AMD K6-233 with a small amount of memory and a 10-15 Gig 
>> HD.  Those can be had at thrift shops for next to nothing these days.
>>
>> John Novack
>>
>>
>> Steph Kerman wrote:
>>> Is there any practical mechanism by which people who subscribe to
>>> commercial Voip services could be called on a direct Voip basis by CNET
>>> users, without going out into the PSTN and NANP?
>>>
>>> Steph
>>   
>


From stfkerman at jps.net  Mon Dec 11 14:19:13 2006
From: stfkerman at jps.net (Steph Kerman)
Date: Mon, 11 Dec 2006 15:19:13 -0500
Subject: [VoIP] Naive question
In-Reply-To: <457DB91C.9070401@stromberg-carlson.org>
References: <457CF2E2.30700@jps.net>
	<457D9241.2090304@stromberg-carlson.org>	<457DAF31.7070606@jps.net>
	<457DB91C.9070401@stromberg-carlson.org>
Message-ID: <457DBD41.6030206@jps.net>

Gizmo is another techno-geek solution.  Anyone who already has an ATA 
set up and has distributed their Voip service throughout the house is 
not going to chain themselves to a PC to talk on the phone. 

If the 2nd port of a 2-port ATA could be hosted by an existing CNET *, 
that would be a step towards what I was asking about.  Ideally, there 
would be some way to register the same ATA port with 2 hosts or 
otherwise at least provide incoming service to it.  Outgoing is less 
important since the commercial Voip subscriber presumably has free/cheap 
toll service and can call the CNET user through the PSTN.  The CNET user 
OTOH might not be a commercial Voip subscriber and being able to call 
commercial Voip subscribers saves the CNET user toll charges.

Steph

John Novack wrote:
> Gizmo may be a good choice then Easy to set up on any Windoze machine 
> with a sound card and mike Free between Gizmo users, open SIP protocol 
> and easy for the CNET/Asterisk user to set up.
>
> I can't speak for how others use CNET/Asterisk, but most of my 
> conversations aren't over my gear, some involves discussions somewhat 
> related, such as why this hobby attracts such strange people, but most 
> of my usage with other members involves matters completely unrelated 
> to telephony. Some of us have commercial VOIP connections as well, 
> integrated into our CNET dialplans Commercial VOIP services usually 
> are locked in some way to the provider, and access to them is, in one 
> way or another through the PSTN scheme, even though it may be VOIP 
> from end to end.
>
> John Novack
>
> Steph Kerman wrote:
>> I am not talking about myself here. Though I have little appetite for
>> monkeying with Windows, I do fix my own car, enjoy playing with
>> switching equipment and will set up an * box of my own at the right
>> time for me. I'm exploring the idea as a means to expand the actual
>> utility of CNET to something beyond what ham radio is mostly about: an
>> end in itself where 90% of the conversation consists of people talking
>> about their rigs.
>>
>> Steph
>
>
>> John Novack wrote:
>>> All require HSIA of some sort, so why wouldn't anyone interested set
>>> up their own Asterisk box? Machine requirements are minimal,
>>> depending on what the goal is. I know of one CNET member that is
>>> running an AMD K6-233 with a small amount of memory and a 10-15 Gig
>>> HD. Those can be had at thrift shops for next to nothing these days.
>>>
>>> John Novack
>>>
>>>
>>> Steph Kerman wrote:
>>>> Is there any practical mechanism by which people who subscribe to
>>>> commercial Voip services could be called on a direct Voip basis by CNET
>>>> users, without going out into the PSTN and NANP?
>>>>
>>>> Steph
>>>
>


From jnovack at stromberg-carlson.org  Mon Dec 11 14:42:40 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Mon, 11 Dec 2006 15:42:40 -0500
Subject: [VoIP] Naive question
In-Reply-To: <457DBD41.6030206@jps.net>
References: <457CF2E2.30700@jps.net>
	<457D9241.2090304@stromberg-carlson.org>	<457DAF31.7070606@jps.net>
	<457DB91C.9070401@stromberg-carlson.org>
	<457DBD41.6030206@jps.net>
Message-ID: <457DC2C0.7050908@stromberg-carlson.org>



Steph Kerman wrote:
> Gizmo is another techno-geek solution.  Anyone who already has an ATA 
> set up and has distributed their Voip service throughout the house is 
> not going to chain themselves to a PC to talk on the phone.
Techno-geek? Doesn't that apply to this whole discussion? Obviously you 
haven't explored Gizmo either.
That isn't the only way Gizmo can be used. Once the free account is 
established the Windows application can be removed, and the information 
gleaned from the installation can be used to set up almost ANY  SIP ATA. 
I know of NO ATA that can have 2 hosts for ONE port, and many of the 
commercial VOIP services lock the box to their service for BOTH ports
Hosting by a CNET would be necessary ONLY for users that don't have a 
Gizmo account. Or is this intended to be simply an OPX?
Remote hosting has and is being done by several CNET users now. SIP can 
be a problem, depending on routers at both ends
IAX remote hosting may be a better solution, though only one IAX box 
supports pulse dial, which may or may not be an issue in this special 
case, and being at the mercy of someone elses computer is less than 
desirable. In the US cases I know of, that is a stopgap measure until a 
better solution is in place.
There is a nice IAX box being sold on eBay for $70 plus shipping, and 
for the one application I am familiar with seems to work OK, though 
higher than normal noise. Perhaps a poor level of comfort noise that 
seems not adjustable. It is operating on the users LAN

Many many ways to skin this cat.

John Novack



From chad at maine.maine.edu  Mon Dec 11 19:43:22 2006
From: chad at maine.maine.edu (Chad Perkins)
Date: Mon, 11 Dec 2006 20:43:22 -0500
Subject: [VoIP] Naive question
In-Reply-To: <457CF2E2.30700@jps.net>
Message-ID: <457DC2EA.4925.10FDC82@localhost>

> Is there any practical mechanism by which people who subscribe to
> commercial Voip services could be called on a direct Voip basis by
> CNET users, without going out into the PSTN and NANP?
> Steph

To clarify:

Q: is there a way for CNET users to call Vonage-like subscribers on a strictly IP 
basis (without going through the PSTN per se)?

A1: As Ben said, the short answer is no.

A2: I know this isn't the answer you are looking for, but as others have mentioned 
there are some "roll your own" solutions.  For example John N. could call me on my 
commercial VoIP line if he knew my VoIP number (08130174).  Creating a CNET-to-
VoIP gateway for single user would be relatively easy; making a broader, scalable 
solution is a little more complex.

A3: The broader answer is that CNET does not "peer" with other VoIP networks.  
And to do this right this is what we would need to do.  There are dozens if not a 
hundred VoIP operations that peer with one another.  FWIW I'm told Vonage no 
longer participates in any peering arrangements (but then again chances are anyone 
looking to save money isn't going to be using the most expensive provider after 
maybe VeriZon and at&t [jab, jab]).

On 11 Dec 2006 at 15:19, Steph Kerman wrote:

> If the 2nd port of a 2-port ATA could be hosted by an existing CNET *, 
> that would be a step towards what I was asking about.  

Yes, if someone has an unlocked ATA that works.  Some of us are doing that now 
(unfortunately the ATA manufacturers don't seem to know what an RJ14 is) and a 
number of VoIP providers allow you to do that; so all you need is a two-line phone (if 
that's your pleasure).

> Ideally, there 
> would be some way to register the same ATA port with 2 hosts or 
> otherwise at least provide incoming service to it.  

Each host has so many configuration parameters that may not be capatable with 
other that this would be tricky and neither the manufacturers nor the providers are 
going to want to implement some thing that could really get ugly.  On the other hand I 
think there is a slightly geeky way of accomplishing this [see* below].

> Outgoing is less 
> important since the commercial Voip subscriber presumably has free/cheap 
> toll service and can call the CNET user through the PSTN.  

If outgoing were not a requirement*, it should be possible to "drive" SIP calls to a 
properly configured SIP device without it being "registered".  This is something I have 
not tried and there a number of issues (without registration, I think the CNET tandem 
is going to need to configured for a fixed IP of the device, and manual firewall 
configuration will most likely be necessary to let the SIP & RTP in).

> The CNET user 
> OTOH might not be a commercial Voip subscriber and being able to call 
> commercial Voip subscribers saves the CNET user toll charges.

True, but signing up with a commercial provider that will give you rates under 
$0.02/minute is so easy why wouldn't one... skip that.  CNET users have (access to) 
Asterisk which comes with FWD sample configs "built in".  FWD has excellent 
peering with other SIP providers (except at&t, VeriZon $ Vonage) and is free.

c.


From stfkerman at jps.net  Mon Dec 11 20:03:38 2006
From: stfkerman at jps.net (Steph Kerman)
Date: Mon, 11 Dec 2006 21:03:38 -0500
Subject: [VoIP] Naive question
In-Reply-To: <457DC2EA.4925.10FDC82@localhost>
References: <457DC2EA.4925.10FDC82@localhost>
Message-ID: <457E0DFA.1020100@jps.net>

Chad,

Thanks for exploring some of the ramifications of my question.

ISTR the Linksys ATA having has 2 separate RJ11s.  Not having an RJ14 is 
no real drawback as far as I can see.  Anyone who sets this up with the 
intention of using it seriously will probably distribute it through 
their own inside wiring or home a residential key system or PBX on it.

Can you please clarify in what sphere the Voip number (08130174) you 
referred to below exists?  On what server or other resource would this 
be translated to an IP or other physical address associated with your ATA?

Do the alternatives you mentioned below all result in direct connections 
between the two ATAs to minimize delay?  I'm unaware of the role of the 
CNET tandem you mentioned.  Perhaps that is used when calling into one 
of Greg's inbound PSTN portals.

Steph

Chad Perkins wrote:
>> Is there any practical mechanism by which people who subscribe to
>> commercial Voip services could be called on a direct Voip basis by
>> CNET users, without going out into the PSTN and NANP?
>> Steph
>>     
>
> To clarify:
>
> Q: is there a way for CNET users to call Vonage-like subscribers on a strictly IP 
> basis (without going through the PSTN per se)?
>
> A1: As Ben said, the short answer is no.
>
> A2: I know this isn't the answer you are looking for, but as others have mentioned 
> there are some "roll your own" solutions.  For example John N. could call me on my 
> commercial VoIP line if he knew my VoIP number (08130174).  Creating a CNET-to-
> VoIP gateway for single user would be relatively easy; making a broader, scalable 
> solution is a little more complex.
>
> A3: The broader answer is that CNET does not "peer" with other VoIP networks.  
> And to do this right this is what we would need to do.  There are dozens if not a 
> hundred VoIP operations that peer with one another.  FWIW I'm told Vonage no 
> longer participates in any peering arrangements (but then again chances are anyone 
> looking to save money isn't going to be using the most expensive provider after 
> maybe VeriZon and at&t [jab, jab]).
>
> On 11 Dec 2006 at 15:19, Steph Kerman wrote:
>
>   
>> If the 2nd port of a 2-port ATA could be hosted by an existing CNET *, 
>> that would be a step towards what I was asking about.  
>>     
>
> Yes, if someone has an unlocked ATA that works.  Some of us are doing that now 
> (unfortunately the ATA manufacturers don't seem to know what an RJ14 is) and a 
> number of VoIP providers allow you to do that; so all you need is a two-line phone (if 
> that's your pleasure).
>
>   
>> Ideally, there 
>> would be some way to register the same ATA port with 2 hosts or 
>> otherwise at least provide incoming service to it.  
>>     
>
> Each host has so many configuration parameters that may not be capatable with 
> other that this would be tricky and neither the manufacturers nor the providers are 
> going to want to implement some thing that could really get ugly.  On the other hand I 
> think there is a slightly geeky way of accomplishing this [see* below].
>
>   
>> Outgoing is less 
>> important since the commercial Voip subscriber presumably has free/cheap 
>> toll service and can call the CNET user through the PSTN.  
>>     
>
> If outgoing were not a requirement*, it should be possible to "drive" SIP calls to a 
> properly configured SIP device without it being "registered".  This is something I have 
> not tried and there a number of issues (without registration, I think the CNET tandem 
> is going to need to configured for a fixed IP of the device, and manual firewall 
> configuration will most likely be necessary to let the SIP & RTP in).
>
>   
>> The CNET user 
>> OTOH might not be a commercial Voip subscriber and being able to call 
>> commercial Voip subscribers saves the CNET user toll charges.
>>     
>
> True, but signing up with a commercial provider that will give you rates under 
> $0.02/minute is so easy why wouldn't one... skip that.  CNET users have (access to) 
> Asterisk which comes with FWD sample configs "built in".  FWD has excellent 
> peering with other SIP providers (except at&t, VeriZon $ Vonage) and is free.
>
> c.
>
> _______________________________________________
> 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 Dec 11 20:17:47 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Mon, 11 Dec 2006 21:17:47 -0500
Subject: [VoIP] Naive question
In-Reply-To: <457DC2EA.4925.10FDC82@localhost>
References: <457DC2EA.4925.10FDC82@localhost>
Message-ID: <457E114B.4010409@stromberg-carlson.org>



Chad Perkins wrote:
>> Is there any practical mechanism by which people who subscribe to
>> commercial Voip services could be called on a direct Voip basis by
>> CNET users, without going out into the PSTN and NANP?
>> Steph
>>     
>
> To clarify:
>
> Q: is there a way for CNET users to call Vonage-like subscribers on a strictly IP 
> basis (without going through the PSTN per se)?
>
> A1: As Ben said, the short answer is no.
>
> A2: I know this isn't the answer you are looking for, but as others have mentioned 
> there are some "roll your own" solutions.  For example John N. could call me on my 
> commercial VoIP line if he knew my VoIP number (08130174).  
Looks like a Stanaphone number to me. Only Stanaphone subscribers have 
access
But then, since I can call you over CNET, what would be the point?
Gizmo uses non existent area code for its number, not reachable from the 
PSTN. ( 1747xxxxxxx )

JN



From chad at maine.maine.edu  Mon Dec 11 20:30:02 2006
From: chad at maine.maine.edu (Chad Perkins)
Date: Mon, 11 Dec 2006 21:30:02 -0500
Subject: [VoIP] FWD SIP question
In-Reply-To: <000601c715a6$05a36fe0$0600a8c0@bluegrasspals.com>
Message-ID: <457DCDDA.27124.13A99A4@localhost>

On 1 Dec 2006 at 19:08, Jayson Smith wrote:
> Hi,
> For a while now I have been having trouble with Ipkall.  Calls would
> never reach me.  Now I just signed up with Sipnumber, thinking Ipkall
> was at fault, and set my forwarding to XXXXXX at fwd.pulver.com, where
> XXXXXX is my FWD number.  Sipnumber can't reach me either, or am I
> doing something wrong? Thanks for any help. Jayson.
 
More firewall problems?
c.


From stfkerman at jps.net  Mon Dec 11 21:12:43 2006
From: stfkerman at jps.net (Steph Kerman)
Date: Mon, 11 Dec 2006 22:12:43 -0500
Subject: [VoIP] Nortel/Ambit ATA
In-Reply-To: <193A6F11-1A61-4017-A4DE-22F453F90856@mail.grenander.com>
References: <454E20A4.26428.F2C83AC@localhost>
	<193A6F11-1A61-4017-A4DE-22F453F90856@mail.grenander.com>
Message-ID: <457E1E2B.6030900@jps.net>

Hi Keelan,

I just inspected inside a Linksys RTP300.  It has an unmarked row of 5 
pins on 0.1" centers and an umnarked and unpopulated 2x7" header 
location.  Both have a few pins with +3.3V on them and ample number of 
obvious grounds.  I'm going to inspect the pins with a scope during 
power up to see whether any of them wiggle in any rhythmic way.  If they 
do I'll find an RS232 driver chip.

Was there just the basic RXD, TXD and ground on the Ambit/Nortel or were 
there flow control lines too?

Steph

Keelan Lightfoot wrote:
> I thought I'd take a minute to share some information on what seems  
> to be a unknown brand of ATA.
>
> I happened across a couple Nortel single port ATAs (re-badged Ambit  
> ATAs) a while back. Like Vonage ATAs, they were locked up. The only  
> access I had was to a useless client-side interface where I could  
> change their network port settings. I popped the case open, and there  
> was a well-labeled serial port header on the board. With a RS-232- 
>  >3.3v level convertor I was able to access the serial console  
> interface, but it unfortunately was also limited without the correct  
> password.
>
> The good news is that if I rebooted it and hit ctrl-c during the boot  
> sequence, it would dump to a very simple boot monitor (commands:  
> help, exit, boot, memdump). I searched the resulting dump for the  
> password for the area that I had access to. About 2 lines of garbage  
> above my password was the supervisor password that gave me access to  
> the entire system.
>
> The interesting thing is that my two ATAs both had the same 6  
> character password -- It makes me think twice about the security of  
> some VOIP providers.
>
> Anyway, if you have access to one of these things, finding the  
> password is not too difficult given time and the motivation to do it.  
> They support pulse dialing, and have a fairly competent web based  
> management interface, and they play well with Asterisk.
>
> - Keelan
>
>
>   


From uax13 at nildram.co.uk  Tue Dec 12 01:25:36 2006
From: uax13 at nildram.co.uk (Nick Wellington)
Date: Tue, 12 Dec 2006 07:25:36 -0000
Subject: [VoIP] Naive question
References: <457CF2E2.30700@jps.net> <457D9241.2090304@stromberg-carlson.org>
Message-ID: <005a01c71dbe$c1e74760$1200000a@Dell>

John Novack posted:
So why wouldn't anyone interested set up their own Asterisk box?

TIME!!!

I work between 32 and 72 hours for the mainline railway.

I work about 35 hours for a private / preserved steam railway.

Plus have a wife, computer network and home to look after occasionally.....

I'm interested in VOIP and CNET and enjoy learning a little from these 
discussions.  But I am not convinced that I have the time to look after an 
asterix as well, learn about it, and Linux etc.

But I am interested in being parented off someone else's, as I already have 
a VOIP phone, which MAY have two ports / lines, I'm not sure.... 
(Grandstream BT100)


NSW 




From chad at maine.maine.edu  Tue Dec 12 15:00:47 2006
From: chad at maine.maine.edu (Chad Perkins)
Date: Tue, 12 Dec 2006 16:00:47 -0500
Subject: [VoIP] Naive question
In-Reply-To: <457E0DFA.1020100@jps.net>
References: <457DC2EA.4925.10FDC82@localhost>
Message-ID: <457ED22F.12380.6451CE@localhost>

On 11 Dec 2006 at 21:03, Steph Kerman wrote:
> Chad,
> Thanks for exploring some of the ramifications of my question.
> 
> ISTR the Linksys ATA having has 2 separate RJ11s.  Not having an RJ14
> is no real drawback as far as I can see.  Anyone who sets this up with
> the intention of using it seriously will probably distribute it
> through their own inside wiring or home a residential key system or
> PBX on it.

There are many models of Linksys (and the pre-Linksys branded Sipura).  The unit 
many people think of is the PAP2 (-XX) that Vonage and a number of the other 
commercial providers sell/supply with their service; it does have two RJ11s for the 
two lines/services it can support.  I have three (v3.1.3) and it is basically the same as 
the Sipura SPA-200X of which I had one (v3.1.2), which Rick Miller is trying to get 
online with, if anyone cares.

To be fair most people aren't phone geeks that aren't going to want to have to set up 
a PBX/key system in their house; no to mention know how or want to spend the 
money.  My irritation with the manufacturer is that they make this consumer product 
(the two line ATA(s)) but didn't take the time (the cost of place two extra traces on 
the circuit broad is negligible  in terms of the cost of the product) to make it (RJ14) so 
a normal person could plug a two-line phone into it.
 
> Can you please clarify in what sphere the Voip number (08130174) you
> referred to below exists?  On what server or other resource would this
> be translated to an IP or other physical address associated with your
> ATA?

Sure.  08130174 is my Stanaphone "number".  Stanaphone, for those who aren't 
familiar with them, is a VoIP company akin to FWD that offers a free service 
(between subscribers).  Like FWD they also offer an optional "opt in" PSTN service 
for a fee ($0.016/min outbound), and they Peer with many of the other providers like 
FWD.  That means that I can not only call everyone on Stanaphone free, but 
everyone on every other network that Stanaphone peers with for free (and vice 
versa).  When I bring up my system and register with my SIP provider they capture 
my IP address so they will know where to send incoming calls, etc.  However, just 
because the number-to-IP association is automatically made on their server doesn't 
mean that the address couldn't be leveraged by a savvy customer to do other things 
(just not automatically).
 
> Do the alternatives you mentioned below all result in direct
> connections between the two ATAs to minimize delay?  I'm unaware of
> the role of the CNET tandem you mentioned.  Perhaps that is used when
> calling into one of Greg's inbound PSTN portals.

No, not really.  Although that is possible and I have heard of people doing it, it is 
almost be like getting a dry copper pair between two locations.  It wouldn't be useful 
for connecting to anyone else.  So, in reality you need a lookup function like ENUM 
or a switching function in the middle.

In our CNET we use Asterisk switches as tandems to front end the EM machines 
and equipment.  In addition to Cl.4-ish tandem functions most of us also have Cl.5 
services in our Asterisk tandems too; for some that includes SIP extensions.  As I 
eluded to in my previous post, if the SIP device is registered to a commercial 
providers switch as the opposed to a CNET switch (Asterisk tandem) it still may be 
possible to have both on one (port of a) SIP device, but the configuration would 
become totally manual.

Chad
1-955-9924 
(6-19pm or VM)
 
> Steph
> 
> Chad Perkins wrote:
> >> Is there any practical mechanism by which people who subscribe to
> >> commercial Voip services could be called on a direct Voip basis by
> >> CNET users, without going out into the PSTN and NANP? Steph
> >>     
> >
> > To clarify:
> >
> > Q: is there a way for CNET users to call Vonage-like subscribers on
> > a strictly IP basis (without going through the PSTN per se)?
> >
> > A1: As Ben said, the short answer is no.
> >
> > A2: I know this isn't the answer you are looking for, but as others
> > have mentioned there are some "roll your own" solutions.  For
> > example John N. could call me on my commercial VoIP line if he knew
> > my VoIP number (08130174).  Creating a CNET-to- VoIP gateway for
> > single user would be relatively easy; making a broader, scalable
> > solution is a little more complex.
> >
> > A3: The broader answer is that CNET does not "peer" with other VoIP
> > networks.  And to do this right this is what we would need to do. 
> > There are dozens if not a hundred VoIP operations that peer with one
> > another.  FWIW I'm told Vonage no longer participates in any peering
> > arrangements (but then again chances are anyone looking to save
> > money isn't going to be using the most expensive provider after
> > maybe VeriZon and at&t [jab, jab]).
> >
> > On 11 Dec 2006 at 15:19, Steph Kerman wrote:
> >
> >   
> >> If the 2nd port of a 2-port ATA could be hosted by an existing CNET
> >> *, that would be a step towards what I was asking about.  
> >>     
> >
> > Yes, if someone has an unlocked ATA that works.  Some of us are
> > doing that now (unfortunately the ATA manufacturers don't seem to
> > know what an RJ14 is) and a number of VoIP providers allow you to do
> > that; so all you need is a two-line phone (if that's your pleasure).
> >
> >   
> >> Ideally, there 
> >> would be some way to register the same ATA port with 2 hosts or
> >> otherwise at least provide incoming service to it.  
> >>     
> >
> > Each host has so many configuration parameters that may not be
> > capatable with other that this would be tricky and neither the
> > manufacturers nor the providers are going to want to implement some
> > thing that could really get ugly.  On the other hand I think there
> > is a slightly geeky way of accomplishing this [see* below].
> >
> >   
> >> Outgoing is less 
> >> important since the commercial Voip subscriber presumably has
> >> free/cheap toll service and can call the CNET user through the
> >> PSTN.  
> >>     
> >
> > If outgoing were not a requirement*, it should be possible to
> > "drive" SIP calls to a properly configured SIP device without it
> > being "registered".  This is something I have not tried and there a
> > number of issues (without registration, I think the CNET tandem is
> > going to need to configured for a fixed IP of the device, and manual
> > firewall configuration will most likely be necessary to let the SIP
> > & RTP in).
> >
> >   
> >> The CNET user 
> >> OTOH might not be a commercial Voip subscriber and being able to
> >> call commercial Voip subscribers saves the CNET user toll charges.
> >>     
> >
> > True, but signing up with a commercial provider that will give you
> > rates under $0.02/minute is so easy why wouldn't one... skip that. 
> > CNET users have (access to) Asterisk which comes with FWD sample
> > configs "built in".  FWD has excellent peering with other SIP
> > providers (except at&t, VeriZon $ Vonage) and is free.
> >
> > c.
> >
> > _______________________________________________
> > 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/
> 
> 
> 
> -- 
> No virus found in this incoming message.
> Checked by AVG Free Edition.
> Version: 7.1.409 / Virus Database: 268.15.16/582 - Release Date:
> 12/11/06
> 




From stfkerman at jps.net  Tue Dec 12 15:29:01 2006
From: stfkerman at jps.net (Steph Kerman)
Date: Tue, 12 Dec 2006 16:29:01 -0500
Subject: [VoIP] Naive question
In-Reply-To: <457ED22F.12380.6451CE@localhost>
References: <457DC2EA.4925.10FDC82@localhost>
	<457ED22F.12380.6451CE@localhost>
Message-ID: <457F1F1D.4020101@jps.net>

Chad Perkins wrote:
> On 11 Dec 2006 at 21:03, Steph Kerman wrote:
>> Chad,
>> Thanks for exploring some of the ramifications of my question.
>>
>> ISTR the Linksys ATA having has 2 separate RJ11s. Not having an RJ14 
>> is no real drawback as far as I can see. Anyone who sets this up with 
>> the intention of using it seriously will probably distribute it 
>> through their own inside wiring or home a residential key system or 
>> PBX on it.
> There are many models of Linksys (and the pre-Linksys branded Sipura). 
> The unit many people think of is the PAP2 (-XX) that Vonage and a 
> number of the other commercial providers sell/supply with their 
> service; it does have two RJ11s for the two lines/services it can 
> support. I have three (v3.1.3) and it is basically the same as the 
> Sipura SPA-200X of which I had one (v3.1.2), which Rick Miller is 
> trying to get online with, if anyone cares.
>
> To be fair most people aren't phone geeks that aren't going to want to 
> have to set up a PBX/key system in their house; not to mention know 
> how or want to spend the money. 
Most people in the population at large, for sure.  I was thinking in 
terms of the people who have commercial Voip and might be inclined to 
provide a second access to CNET. 

The instructions many of the Voip providers give for connecting up the 
ATA can result in back-feeding the analog side of the ATA towards the CO 
on the cable pair that originally delivered the PSTN dial tone.  Or 
limit you to using one phone plugged directly into the ATA.  Or 
stringing modular cords around.  Many ATAs, when installed as 
instructed, deliver reversed loop voltage and will not work with older 
2500 and Trimline sets.
> My irritation with the manufacturer is that they make this consumer 
> product (the two line ATA(s)) but didn't take the time (the cost of 
> place two extra traces on the circuit broad is negligible in terms of 
> the cost of the product) to make it (RJ14) so a normal person could 
> plug a two-line phone into it.
Probably not done more due to lack of competence on the part of the 
designer.  A large part of the non-phone population has no idea what an 
RJ14 is.

That said, there are ways to combine the two separate lines into an RJ14 
config. using OTS stuff from Radio Shack.
>> Can you please clarify in what sphere the Voip number (08130174) you 
>> referred to below exists? On what server or other resource would this 
>> be translated to an IP or other physical address associated with your 
>> ATA?
> Sure. 08130174 is my Stanaphone "number". Stanaphone, for those who 
> aren't familiar with them, is a VoIP company akin to FWD that offers a 
> free service (between subscribers). Like FWD they also offer an 
> optional "opt in" PSTN service for a fee ($0.016/min outbound), and 
> they Peer with many of the other providers like FWD. That means that I 
> can not only call everyone on Stanaphone free, but everyone on every 
> other network that Stanaphone peers with for free (and vice versa). 
> When I bring up my system and register with my SIP provider they 
> capture my IP address so they will know where to send incoming calls, 
> etc. However, just
> because the number-to-IP association is automatically made on their 
> server doesn't mean that the address couldn't be leveraged by a savvy 
> customer to do other things (just not automatically).
So who is your "SIP provider" from whom they capture your IP address?  
Is that another party other than Stanaphone or FWD?
>> Do the alternatives you mentioned below all result in direct 
>> connections between the two ATAs to minimize delay? I'm unaware of
>> the role of the CNET tandem you mentioned. Perhaps that is used when 
>> calling into one of Greg's inbound PSTN portals.
> No, not really. Although that is possible and I have heard of people 
> doing it, it is almost be like getting a dry copper pair between two 
> locations. It wouldn't be useful for connecting to anyone else. So, in 
> reality you need a lookup function like ENUM or a switching function 
> in the middle.
I see.  But a switching function, if I understand correctly, would 
double the delay so an ENUM would seem to be the best way.
> In our CNET we use Asterisk switches as tandems to front end the EM 
> machines and equipment. In addition to Cl.4-ish tandem functions most 
> of us also have Cl.5 services in our Asterisk tandems too; for some 
> that includes SIP extensions.
Please ID what "CI.4" and "CI.5" refer to .  Are these abbreviations for 
something?
> As I eluded to in my previous post, if the SIP device is registered to 
> a commercial providers switch as the opposed to a CNET switch 
> (Asterisk tandem) it still may be possible to have both on one (port 
> of a) SIP device, but the configuration would become totally manual.
I assumed that if it were feasible at all, it would have to be manually 
installed.  Is any of this configuration at the ATA itself or is it all 
done at the Asterisk that is sharing it?

Thanks
Steph
>> Chad Perkins wrote:
>>>> Is there any practical mechanism by which people who subscribe to
>>>> commercial Voip services could be called on a direct Voip basis by
>>>> CNET users, without going out into the PSTN and NANP? Steph
>>>>
>>> To clarify:
>>>
>>> Q: is there a way for CNET users to call Vonage-like subscribers on
>>> a strictly IP basis (without going through the PSTN per se)?
>>>
>>> A1: As Ben said, the short answer is no.
>>>
>>> A2: I know this isn't the answer you are looking for, but as others
>>> have mentioned there are some "roll your own" solutions. For
>>> example John N. could call me on my commercial VoIP line if he knew
>>> my VoIP number (08130174). Creating a CNET-to- VoIP gateway for
>>> single user would be relatively easy; making a broader, scalable
>>> solution is a little more complex.
>>>
>>> A3: The broader answer is that CNET does not "peer" with other VoIP
>>> networks. And to do this right this is what we would need to do.
>>> There are dozens if not a hundred VoIP operations that peer with one
>>> another. FWIW I'm told Vonage no longer participates in any peering
>>> arrangements (but then again chances are anyone looking to save
>>> money isn't going to be using the most expensive provider after
>>> maybe VeriZon and at&t [jab, jab]).
>>>
>>> On 11 Dec 2006 at 15:19, Steph Kerman wrote:
>>>
>>>
>>>> If the 2nd port of a 2-port ATA could be hosted by an existing CNET
>>>> *, that would be a step towards what I was asking about.
>>>>
>>> Yes, if someone has an unlocked ATA that works. Some of us are
>>> doing that now (unfortunately the ATA manufacturers don't seem to
>>> know what an RJ14 is) and a number of VoIP providers allow you to do
>>> that; so all you need is a two-line phone (if that's your pleasure).
>>>
>>>
>>>> Ideally, there
>>>> would be some way to register the same ATA port with 2 hosts or
>>>> otherwise at least provide incoming service to it.
>>>>
>>> Each host has so many configuration parameters that may not be
>>> capatable with other that this would be tricky and neither the
>>> manufacturers nor the providers are going to want to implement some
>>> thing that could really get ugly. On the other hand I think there
>>> is a slightly geeky way of accomplishing this [see* below].
>>>
>>>
>>>> Outgoing is less
>>>> important since the commercial Voip subscriber presumably has
>>>> free/cheap toll service and can call the CNET user through the
>>>> PSTN.
>>>>
>>> If outgoing were not a requirement*, it should be possible to
>>> "drive" SIP calls to a properly configured SIP device without it
>>> being "registered". This is something I have not tried and there a
>>> number of issues (without registration, I think the CNET tandem is
>>> going to need to configured for a fixed IP of the device, and manual
>>> firewall configuration will most likely be necessary to let the SIP
>>> & RTP in).
>>>
>>>
>>>> The CNET user
>>>> OTOH might not be a commercial Voip subscriber and being able to
>>>> call commercial Voip subscribers saves the CNET user toll charges.
>>>>
>>> True, but signing up with a commercial provider that will give you
>>> rates under $0.02/minute is so easy why wouldn't one... skip that.
>>> CNET users have (access to) Asterisk which comes with FWD sample
>>> configs "built in". FWD has excellent peering with other SIP
>>> providers (except at&t, VeriZon $ Vonage) and is free.
>>>
>


From keelan at mail.grenander.com  Tue Dec 12 20:50:10 2006
From: keelan at mail.grenander.com (Keelan Lightfoot)
Date: Tue, 12 Dec 2006 18:50:10 -0800
Subject: [VoIP] Nortel/Ambit ATA
In-Reply-To: <457E1E2B.6030900@jps.net>
References: <454E20A4.26428.F2C83AC@localhost>
	<193A6F11-1A61-4017-A4DE-22F453F90856@mail.grenander.com>
	<457E1E2B.6030900@jps.net>
Message-ID: <7536774D-6362-49A7-818A-BD9AAF1580BF@mail.grenander.com>

Steph,

The Ambit ATA has a 5 pin header with ground, TXD, 3.3V, RXD and  
ground again. You can see it here:

http://pixalis.com/keelan/ambitata.html

What controller does the RTP300 use? The Ambit utilizes a Broadcom  
BCM1112; I'm curious if the RTP300 uses something similar.

- Keelan

On Dec 11, 2006, at 7:12 PM, Steph Kerman wrote:

> Hi Keelan,
>
> I just inspected inside a Linksys RTP300.  It has an unmarked row of 5
> pins on 0.1" centers and an umnarked and unpopulated 2x7" header
> location.  Both have a few pins with +3.3V on them and ample number of
> obvious grounds.  I'm going to inspect the pins with a scope during
> power up to see whether any of them wiggle in any rhythmic way.  If  
> they
> do I'll find an RS232 driver chip.
>
> Was there just the basic RXD, TXD and ground on the Ambit/Nortel or  
> were
> there flow control lines too?
>
> Steph
>
> Keelan Lightfoot wrote:
>> I thought I'd take a minute to share some information on what seems
>> to be a unknown brand of ATA.
>>
>> I happened across a couple Nortel single port ATAs (re-badged Ambit
>> ATAs) a while back. Like Vonage ATAs, they were locked up. The only
>> access I had was to a useless client-side interface where I could
>> change their network port settings. I popped the case open, and there
>> was a well-labeled serial port header on the board. With a RS-232-
>>> 3.3v level convertor I was able to access the serial console
>> interface, but it unfortunately was also limited without the correct
>> password.
>>
>> The good news is that if I rebooted it and hit ctrl-c during the boot
>> sequence, it would dump to a very simple boot monitor (commands:
>> help, exit, boot, memdump). I searched the resulting dump for the
>> password for the area that I had access to. About 2 lines of garbage
>> above my password was the supervisor password that gave me access to
>> the entire system.
>>
>> The interesting thing is that my two ATAs both had the same 6
>> character password -- It makes me think twice about the security of
>> some VOIP providers.
>>
>> Anyway, if you have access to one of these things, finding the
>> password is not too difficult given time and the motivation to do it.
>> They support pulse dialing, and have a fairly competent web based
>> management interface, and they play well with Asterisk.
>>
>> - Keelan
>>
>>
>>
>
> _______________________________________________
> 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  Tue Dec 12 20:54:35 2006
From: stfkerman at jps.net (Steph Kerman)
Date: Tue, 12 Dec 2006 21:54:35 -0500
Subject: [VoIP] Nortel/Ambit ATA
In-Reply-To: <7536774D-6362-49A7-818A-BD9AAF1580BF@mail.grenander.com>
References: <454E20A4.26428.F2C83AC@localhost>	<193A6F11-1A61-4017-A4DE-22F453F90856@mail.grenander.com>	<457E1E2B.6030900@jps.net>
	<7536774D-6362-49A7-818A-BD9AAF1580BF@mail.grenander.com>
Message-ID: <457F6B6B.7010502@jps.net>

Keelan,

Thanks.

I took photos of the board.  I'll inspect them and if they are nice and 
clear I will put them where you can fetch them.  Otherwise I will dig it 
back out and answer the specific question.

Steph

Keelan Lightfoot wrote:
> Steph,
>
> The Ambit ATA has a 5 pin header with ground, TXD, 3.3V, RXD and  
> ground again. You can see it here:
>
> http://pixalis.com/keelan/ambitata.html
>
> What controller does the RTP300 use? The Ambit utilizes a Broadcom  
> BCM1112; I'm curious if the RTP300 uses something similar.
>
> - Keelan
>
> On Dec 11, 2006, at 7:12 PM, Steph Kerman wrote:
>
>   
>> Hi Keelan,
>>
>> I just inspected inside a Linksys RTP300.  It has an unmarked row of 5
>> pins on 0.1" centers and an umnarked and unpopulated 2x7" header
>> location.  Both have a few pins with +3.3V on them and ample number of
>> obvious grounds.  I'm going to inspect the pins with a scope during
>> power up to see whether any of them wiggle in any rhythmic way.  If  
>> they
>> do I'll find an RS232 driver chip.
>>
>> Was there just the basic RXD, TXD and ground on the Ambit/Nortel or  
>> were
>> there flow control lines too?
>>
>> Steph
>>
>> Keelan Lightfoot wrote:
>>     
>>> I thought I'd take a minute to share some information on what seems
>>> to be a unknown brand of ATA.
>>>
>>> I happened across a couple Nortel single port ATAs (re-badged Ambit
>>> ATAs) a while back. Like Vonage ATAs, they were locked up. The only
>>> access I had was to a useless client-side interface where I could
>>> change their network port settings. I popped the case open, and there
>>> was a well-labeled serial port header on the board. With a RS-232-
>>>       
>>>> 3.3v level convertor I was able to access the serial console
>>>>         
>>> interface, but it unfortunately was also limited without the correct
>>> password.
>>>
>>> The good news is that if I rebooted it and hit ctrl-c during the boot
>>> sequence, it would dump to a very simple boot monitor (commands:
>>> help, exit, boot, memdump). I searched the resulting dump for the
>>> password for the area that I had access to. About 2 lines of garbage
>>> above my password was the supervisor password that gave me access to
>>> the entire system.
>>>
>>> The interesting thing is that my two ATAs both had the same 6
>>> character password -- It makes me think twice about the security of
>>> some VOIP providers.
>>>
>>> Anyway, if you have access to one of these things, finding the
>>> password is not too difficult given time and the motivation to do it.
>>> They support pulse dialing, and have a fairly competent web based
>>> management interface, and they play well with Asterisk.
>>>
>>> - Keelan
>>>
>>>
>>>
>>>       
>> _______________________________________________
>> 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  Tue Dec 12 21:30:22 2006
From: greg at vyger.net (Greg Blakely)
Date: Tue, 12 Dec 2006 21:30:22 -0600
Subject: [VoIP] Naive question
Message-ID: <A691C3685A152E4AAC9608B8D8E11316685F07@godzilla.hickory.vyger.net>

Let me add to the clutter.  John Covert mentioned E.164, which is what I
use.  I query both e164.org and e164.arpa for my PSTN calls.  If a
number is listed in either location, a direct ip-to-ip connection is
made.

>From an ATA, direct VOIP connections to off-net locations is clunky at
best.  

Another reason to get an asterisk box going...

> -----Original Message-----
> From: voip-bounces at ckts.info [mailto:voip-bounces at ckts.info] On Behalf
Of
> John Novack
> Sent: Monday, December 11, 2006 8:18 PM
> To: Voice Over IP Tandem for Analog Switches
> Subject: Re: [VoIP] Naive question
> 
> 
> 
> Chad Perkins wrote:
> >> Is there any practical mechanism by which people who subscribe to
> >> commercial Voip services could be called on a direct Voip basis by
> >> CNET users, without going out into the PSTN and NANP?
> >> Steph
> >>
> >
> > To clarify:
> >
> > Q: is there a way for CNET users to call Vonage-like subscribers on
a
> strictly IP
> > basis (without going through the PSTN per se)?
> >
> > A1: As Ben said, the short answer is no.
> >
> > A2: I know this isn't the answer you are looking for, but as others
have
> mentioned
> > there are some "roll your own" solutions.  For example John N. could
> call me on my
> > commercial VoIP line if he knew my VoIP number (08130174).
> Looks like a Stanaphone number to me. Only Stanaphone subscribers have
> access
> But then, since I can call you over CNET, what would be the point?
> Gizmo uses non existent area code for its number, not reachable from
the
> PSTN. ( 1747xxxxxxx )
> 
> JN
> 
> 
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
> 




From chad at maine.maine.edu  Wed Dec 13 21:32:42 2006
From: chad at maine.maine.edu (Chad Perkins)
Date: Wed, 13 Dec 2006 22:32:42 -0500
Subject: [VoIP] Naive question
In-Reply-To: <457F1F1D.4020101@jps.net>
References: <457ED22F.12380.6451CE@localhost>
Message-ID: <45807F8A.8227.15DA3DB@localhost>

On 12 Dec 2006 at 16:29, Steph Kerman wrote:

> Chad Perkins wrote:
> > On 11 Dec 2006 at 21:03, Steph Kerman wrote:
> >> Chad,
> >> Thanks for exploring some of the ramifications of my question.
[big snip]
> Most people in the population at large, for sure.  I was thinking in
> terms of the people who have commercial Voip and might be inclined to
> provide a second access to CNET. 

CNET is a "techie" bunch for the most part, some more than others, some in different 
areas.  In an effort to get more of our Telephone Museum members involved I have 
been trying to perfect a configuration using two-line ATAs (similar to what Greg has 
done) as many of these folks are "telephone people" and/or "collectors" but not 
"computer" or "Internet" people.

> That said, there are ways to combine the two separate lines into an
> RJ14 config. using OTS stuff from Radio Shack. 

I know, probably $10.  Two extra line cords, a splitter and a coupler hitched up back 
to front.  I'm still tempted to warm up the solder gun... [snip]

> So who is your "SIP provider" from
> whom they capture your IP address?

Normally the commercial (SIP) provider is going to capture the IP address when the 
SIP application (device) starts so that they know where to send calls, status 
monitoring and "messages" (message waiting indication for example), etc.  

Here's a question if anyone knows, if a Vonage customer calls another Vonage 
customer, does it ever hit the PSTN?  I think I can safely say the other providers I 
use/am acquainted with do not.

> Is that another party other than Stanaphone or FWD?

No, one in the same.

> I see.  But a switching
> function, if I understand correctly, would double the delay so an ENUM
> would seem to be the best way. 

Yes and no.  Even with a switching function in the middle many VoIP 
implementations allow for the voice (and video) channels to be redirected to the end 
points as soon as the connection comes to avoid this very issue.  The end points do 
however maintain a control connection with the switch for things like the ability to 
transfer, conference, etc., update Call Detail Records, and more.

>> In our CNET we use Asterisk switches
>> as tandems to front end the EM machines and equipment. In addition
>> to Cl.4-ish tandem functions most of us also have Cl.5 services in
>> our Asterisk tandems too; for some that includes SIP extensions.
> Please ID what "CI.4" and "CI.5" refer to .  Are these abbreviations
> for something?

Sorry, Class 4 and Class 5.

>> As I eluded to in my previous post, if the SIP device
>> is registered to a commercial providers switch as the opposed to a
>> CNET switch (Asterisk tandem) it still may be possible to have both
>> on one (port of a) SIP device, but the configuration would become
>> totally manual. 
> I assumed that if it were feasible at all, it would
> have to be manually installed.  Is any of this configuration at the
> ATA itself or is it all done at the Asterisk that is sharing it?

This is a theoretical answer and based on some assumptions not having done this 
quite this way personally ...

1. SIP device: The ATA (or IP phone) would have to be (re)configured to accept  
unsolicited connections (if you think that sounds kind of risky you are right, read on).

2. Firewall: Based on #1 the SIP device had better be "behind" are firewall (I have no 
experience with the combo units).  Because the device is only registered with the 
commercial provider, the firewall isn't going to be expecting, or know what to do with 
incoming IP packets from the Asterisk "tandem"; therefore the firewall will need to 
have port forwarding enabled for the proper (SIP) ports ... and because the device is 
now open to the Internet on those ports you would want to add an access list 
restricting connections to just the Asterisk tandem (and hope you don't kill the 
commercial connection in the process).  This is an area that varies widely from 
vendor to vendor.

3. Asterisk tandem: Because the device isn't registering the him he isn't going to 
automatically know where on the Internet the device is in terms of IP address, so the 
'sip.conf' file in the /etc/asterisk directory will created/modified to specify the address 
to use in connection with that device.  This would probably be done one of two ways, 
either by hardcoded IP address, or by a dynamic DNS (DDNS) name.

4. Internet Address:  Because of #3 above, the end device is going to have to either 
have/share a fixed ISP IP address; or you would need to setup a DDNS entry.  
Unfortunately I don't think I've seen a device with firmware that supports DDNS.  One 
should be able to install a DDNS client on a regular PC on the same network as the 
device to get around this though.

FWIW these four basic items are roughly the same steps one needs to address in an 
ENUM setup, in fact I see no reason why under the right circumstances such a 
device couldn't participate in an ENUM based network (mind you while CNET is an 
ENUM network, it is also based primarily on the IAX2 protocol as opposed to SIP 
which can be a pain in the neck.

Moral of the story, go with an unlocked two line unit, use the standard registration 
process; it will be a lot easier with a lot less problems.

c.


From jnovack at stromberg-carlson.org  Wed Dec 13 21:37:53 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Wed, 13 Dec 2006 22:37:53 -0500
Subject: [VoIP] Naive question
In-Reply-To: <45807F8A.8227.15DA3DB@localhost>
References: <457ED22F.12380.6451CE@localhost>
	<45807F8A.8227.15DA3DB@localhost>
Message-ID: <4580C711.3060307@stromberg-carlson.org>



Chad Perkins wrote:
>
> Here's a question if anyone knows, if a Vonage customer calls another Vonage customer, does it ever hit the PSTN? 
I feel sure, from the speed of the call setup time that it does NOT

John Novack



From jnovack at stromberg-carlson.org  Wed Dec 13 21:46:15 2006
From: jnovack at stromberg-carlson.org (John Novack)
Date: Wed, 13 Dec 2006 22:46:15 -0500
Subject: [VoIP] Naive question
In-Reply-To: <4580C711.3060307@stromberg-carlson.org>
References: <457ED22F.12380.6451CE@localhost>	<45807F8A.8227.15DA3DB@localhost>
	<4580C711.3060307@stromberg-carlson.org>
Message-ID: <4580C907.5070401@stromberg-carlson.org>



John Novack wrote:
> Chad P