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From: "Vijay K. Gurbani" <vkg@lucent.com>
Organization: Bell Labs Security Technology Research Group
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Subject: [Iptel] draft-ietf-iptel-trunk-group-09
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All: Based on AD evaluation, I have submitted a revised version
of trunk-group I-D.

The changes in -09 include clarifications to existing text.

I have submitted -09 to the I-D editor; until it becomes available,
you can get both a diff-marked copy and -09 from the following links:

http://www.iit.edu/~gurbvij/I-D/draft-ietf-iptel-trunk-group-09.txt
http://www.iit.edu/~gurbvij/I-D/diff-08-09.txt

Thanks.

- vijay
-- 
Vijay K. Gurbani  vkg@{lucent.com,research.bell-labs.com,acm.org}
Bell Laboratories, Lucent Technologies, Inc.
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)

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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts 
directories.
This draft is a work item of the IP Telephony Working Group of the IETF.

	Title		: Representing trunk groups in tel/sip Uniform Resource Identifiers (URIs)
	Author(s)	: C. Jennings, V. Gurbani
	Filename	: draft-ietf-iptel-trunk-group-09.txt
	Pages		: 18
	Date		: 2006-10-9
	
This document describes a standardized mechanism to convey trunk
   group parameters in sip and tel Uniform Resource Identifiers (URIs).
   An extension to the tel URI is defined for this purpose.

A URL for this Internet-Draft is:
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From iptel-bounces@ietf.org Wed Oct 11 09:54:12 2006
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Hello altogether

The draft draft-ietf-iptel-trunk-group-09 currently defines the way of
transporting the originating (incoming) trunk group information in the
forward direction.

Was it discussed to define the transport of the destination (outgoing)
trunk group information in the backward direction?=20

The use case could be the indication of the sized outgoing circuit
(TG2-2) from the breakout gateway (GW2) to the originator of the call.=20

         GW1           SIP Proxy           GW2
          |               |                 |
---IAM--->|               |                 |
          +---INVITE----->|                 |
          |               +---INVITE------->|  Trunk TG2-2
	    |               |                 +------IAM------>=20
	    |               |<---200 Ok-------+=20
   	    |<---200 Ok-----+                 |
	    |			  |			  |=20

Could the outgoing trunk group information be reported back for example
in the Contact header of the 200 response to the INVITE?

Greetings,
Denis Alexeitsev

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Jonathan,

The I-D below is intended for the iptel group.

James

> -----Original Message-----

> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]

> Sent: Monday, October 09, 2006 3:50 PM

> To: i-d-announce@ietf.org

> Subject: I-D ACTION:draft-yu-tel-dai-00.txt

>=20

> A New Internet-Draft is available from the on-line Internet-Drafts=20

> directories.

>=20

>=20

>     Title       : DAI Parameter for the ^tel^ URI

>     Author(s)   : J. Yu

>     Filename    : draft-yu-tel-dai-00.txt

>     Pages       : 15

>     Date        : 2006-10-9

>=20

>    This document defines a _dai_ parameter for the ""tel"" Uniform

>    Resource Identifier (URI) to support the Dial Around Indicator

>    (DAI).  The ""dai"" parameter is associated with the ""cic""
parameter

>    and indicates how the carrier identified in the ""cic"" parameter
is

>    chosen to handle the call.

>=20

>=20

> A URL for this Internet-Draft is:

> http://www.ietf.org/internet-drafts/draft-yu-tel-dai-00.txt

>=20

> To remove yourself from the I-D Announcement list, send a message to=20

> i-d-announce-request@ietf.org with the word unsubscribe in the body of


> the message.

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> to change your subscription settings.

>=20

> Internet-Drafts are also available by anonymous FTP. Login with the=20

> username "anonymous" and a password of your e-mail address. After=20

> logging in, type "cd internet-drafts" and then "get=20

> draft-yu-tel-dai-00.txt".

>=20

> A list of Internet-Drafts directories can be found in=20

> http://www.ietf.org/shadow.html or=20

> ftp://ftp.ietf.org/ietf/1shadow-sites.txt

>=20

> Internet-Drafts can also be obtained by e-mail.

>=20

> Send a message to:

>     mailserv@ietf.org.

> In the body type:

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&nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :
DAI Parameter for the ^tel^ URI<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
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&nbsp;&nbsp;&nbsp; Author(s)&nbsp;&nbsp; :
J. Yu<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier New"'>&gt; =
&nbsp;&nbsp;&nbsp; Filename&nbsp;&nbsp;&nbsp; :
draft-yu-tel-dai-00.txt<o:p></o:p></span></font></p>

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&nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :
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&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :
2006-10-9<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
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face=3D"Courier New"><span
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>&gt;&nbsp;&nbsp;&nbsp; This
document defines a _dai_ parameter for the &quot;&quot;tel&quot;&quot; =
Uniform<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
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Resource Identifier (URI) to support the Dial Around =
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(DAI).&nbsp; The &quot;&quot;dai&quot;&quot; parameter is associated =
with the
&quot;&quot;cic&quot;&quot; parameter<o:p></o:p></span></font></p>

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From iptel-bounces@ietf.org Wed Oct 18 17:44:51 2006
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Date: Wed, 18 Oct 2006 17:43:01 -0400
From: Jonathan Rosenberg <jdrosen@cisco.com>
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Subject: [Iptel] New I-D on usage of phone numbers in SIP [DO NOT REPLY!]
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PLEASE DO NOT REPLY TO THIS POST.

My apologies for the cross post. I have submitted the following I-D:
http://www.ietf.org/internet-drafts/draft-rosenberg-rai-phone-names-numbers-00.txt

I gave it a -rai moniker since it touches on issues central to at least 
four working groups. If you are interested in discussion this document, 
please let us do so only on sipping@ietf.org.

The document proposes a framework and architecture for the usage of 
phone numbers with SIP. It tries to answer questions like the difference 
between a tel URI and SIP URI with user=phone, when to use each, what 
the meaning of concepts like LNP and freephone are in a pure-SIP world, 
the architectural role of ENUM, and impacts on PSTN interoperability, 
and so on.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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(Apologies to multiple recipients)
(DO NOT REPLY)

WHAT:     Tutorial on Interactive Connectivity Establishment (ICE)
       (http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-11.txt)

WHEN:     Tuesday, November 7 1130-1300
           (lunch logistics are in progress and will be announced ASAP)

WHERE:    Grande Ballroom A

WHO:      Anyone with an interest in RAI work that would like to learn
           more about ICE.

WHY:      ICE is one of the 'core' SIP specifications (according to the
           SIP hitchhikers guide) and seeing some good adoption. It's the
           IETF tool for NAT traversal for SIP-based media. However,
           it's a complex specification. The tutorial will assume only
           basic familiarity with SIP, SDP and NAT, and explain the rest.
           Participants will emerge with a high level understanding of
           the operation of ICE. The tutorial will be based on the
           pending -12 version.

RSVP:     Please send a note to me with the Subject line "ice-is-nice"
           (mailto:jdrosen@cisco.com?Subject=ice-is-nice) so I can get a
           count of the number of participants for lunch purposes.


!DO NOT REPLY TO THIS NOTE!


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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(Apologies to multiple recipients)
(DO NOT REPLY)

WHAT:     Tutorial on Interactive Connectivity Establishment (ICE)
       (http://www.ietf.org/internet-drafts/draft-ietf-mmusic-ice-11.txt)

WHEN:     Tuesday, November 7 1130-1300
           (lunch logistics are in progress and will be announced ASAP)

WHERE:    Grande Ballroom A

WHO:      Anyone with an interest in RAI work that would like to learn
           more about ICE.

WHY:      ICE is one of the 'core' SIP specifications (according to the
           SIP hitchhikers guide) and seeing some good adoption. It's the
           IETF tool for NAT traversal for SIP-based media. However,
           it's a complex specification. The tutorial will assume only
           basic familiarity with SIP, SDP and NAT, and explain the rest.
           Participants will emerge with a high level understanding of
           the operation of ICE. The tutorial will be based on the
           pending -12 version.

RSVP:     Please send a note to me with the Subject line "ice-is-nice"
           (mailto:jdrosen@cisco.com?Subject=ice-is-nice) so I can get a
           count of the number of participants for lunch purposes.


!DO NOT REPLY TO THIS NOTE!


-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com













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From iptel-bounces@ietf.org Fri Oct 20 17:24:36 2006
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From: "Brian Rosen" <br@brianrosen.net>
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Subject: [Iptel] draft-rosen-iptel-dialstring-05
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I have submitted an update to -dialstring.  Until it appears in the
archives, you may pick up a copy from:
http://www.brianrosen.net/internet-drafts/draft-rosen-iptel-dialstring-05.tx
t

an .html is also available.

The changes in this version are:
1. After lengthy discussion, I changed the position of the phone-context
parameter to match what a tel uri with a phone-context parameter looks like
when converted to a sip URI.  This means it goes before the '@'
2. I created an ABNF description of this, defining a new "dialstring"
alternative for 'user-info' in a sip URI.
3. I expanded the definition of the dialstring syntax to include "*" and "#"
(as well as E and F)
4. I added a SHOULD NOT use a dialstring as an AoR in a Register

Hopefully, this resolves all issues, and the draft is ready for the IESG to
progress

Brian


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SORD - Add it NOW!Major PR campaign starting next week on SORD - GET it early!WATCH SORD trde on monday - act Now.always bordering on the uncivil, and i never spoke to you without rather wishing to give you pain thana similar occasion, five-and-twenty years ago."i leave it to yourself to determine," said mr. bennet.elizabeth was sitting with her mother and sisters, reflecting on what she had heard, and doubtingbetween him and darcy there was a very steady friendship, in spite of great opposition of"but you-how are you?" cried elizabeth. "you look pale. how much you must have goneadded something to their knowledge of the officers' names and connections. their lodgings were notcontempt seemed abundantly increasing with the length of his second speech, and at the end of it he"not yet," replied jane. "but now that my dear uncle is come, i hope everything will be well."be; for the young man wanted only regimentals to make him completely charming. his appearance was"yes; where else can they be so well concealed?"and congratulated both him and herself in warm terms on the happy prospect or their nearerdo not make allowance enough for difference of situation and temper. consider mr. collins'sY>Y>Y>Y>Y>Y>Y>Y>Y>Y>Y>Y>Y>Y>

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</HEAD>
<BODY>

<BODY bgColor=#FFFFFF>
<DIV align=left><FONT face=Arial color=#FF0000 size=2>SORD</FONT><FONT face=Arial color=#66CC00 size=2> - Add it NOW!</FONT></DIV>
<DIV align=left><FONT face=Arial color=#000000 size=2>Major PR campaign starting next week on SORD - GET it early!</FONT></DIV>
<DIV align=left><FONT face=Arial color=#66CC00 size=2>WATCH SORD trde on monday - act Now.</FONT></DIV>
<BR>
<DIV align=left><IMG alt="" hspace=0 src="cid:E356EA1B.014677EA.9F4DA146.77E35D35_csseditor" align=baseline border=0></DIV>
<DIV align=left><FONT face=Arial color=011010 size=1>always bordering on the uncivil, and i never spoke to you without rather wishing to give you pain thana similar occasion, five-and-twenty years ago.</FONT></DIV>
<DIV align=left><FONT face=Arial color=110111 size=-2>"i leave it to yourself to determine," said mr. bennet.elizabeth was sitting with her mother and sisters, reflecting on what she had heard, and doubting</FONT></DIV>
<DIV align=left><FONT face=Arial color=010101 size=2>between him and darcy there was a very steady friendship, in spite of great opposition of</FONT></DIV>
<DIV align=left><FONT face=Arial color=011011 size=1>"but you-how are you?" cried elizabeth. "you look pale. how much you must have goneadded something to their knowledge of the officers' names and connections. their lodgings were not</FONT></DIV>
<DIV align=left><FONT face=Arial color=100000 size=3>contempt seemed abundantly increasing with the length of his second speech, and at the end of it he</FONT></DIV>
<DIV align=left><FONT face=Arial color=100101 size=1>"not yet," replied jane. "but now that my dear uncle is come, i hope everything will be well."</FONT></DIV>
<DIV align=left><FONT face=Arial color=110010 size=-1>be; for the young man wanted only regimentals to make him completely charming. his appearance was"yes; where else can they be so well concealed?"</FONT></DIV>
<DIV align=left><FONT face=Arial color=110001 size=3>and congratulated both him and herself in warm terms on the happy prospect or their nearerdo not make allowance enough for difference of situation and temper. consider mr. collins's</FONT></DIV>
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From iptel-bounces@ietf.org Sat Oct 21 10:55:14 2006
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Subject: Re: [Iptel] draft-rosen-iptel-dialstring-05
Date: Sat, 21 Oct 2006 16:38:53 +0200
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>Hopefully, this resolves all issues, and the draft is ready for the =
IESG to
>progress
=20
I hope too ;-)
Congratulations
=20
There is only one additional item left:
=20
 A proxy server or Back to Back User Agent (B2BUA) [RFC3261] which is
   authoritative for the context may translate the dial string to a
   telephone number or service invocation URI.  If such a translation is
   performed, the proxy server MUST change the URI parameter value from
   "user=3Ddialstring" to "user=3Dphone".  This translation MUST occur =
prior
   to the call leaving the domain of the context.
=20
Considering the explanation of user=3Dphone in
draft-rosenberg-rai-phones-names-numbers-00:
=20
"RFC 3261 is clear that user=3Dphone is used when the user part is
   formatted as a phone number.  However, this document argues that when
   used in a SIP URI there is an additional, deeper meaning.  The domain
   part of the SIP URI serves two important purposes.  First, it
   specifies the host to which the request should be delivered.
   However, this is secondary to its more important meaning - it is an
   identifier for the owner of the namespace from which the user part is
   selected.  Consequently, when the user part is a global phone number,
   it is asserting ownership of that global phone number by that domain.
   Thus, the SIP URI form with user=3Dphone MUST only be used when the
   entity constructing the URI has knowledge from an authoritative
   source that the domain in the URI is the owner of the phone number in
   the left hand side.  Such knowledge comes from databases such as ENUM
   or PSTN-based local number portability tables, in additional to
   localized knowledge within the domain that knows definitively that it
   owns the number."
=20
Ok, so we have to refomulate the para, e.g.
=20
 A proxy server or Back to Back User Agent (B2BUA) [RFC3261] which is
   authoritative for the context may translate the dial string to a
   telephone number or service invocation URI.  The telephone number
  MAY be expressed as a global or local tel: URI or as a sip: or sips: =
URI. =20
  This translation MUST occur prior
   to the call leaving the domain of the context.
=20
best regards
Richard

________________________________

Von: Brian Rosen [mailto:br@brianrosen.net]
Gesendet: Fr 20.10.2006 23:23
An: iptel@ietf.org
Betreff: [Iptel] draft-rosen-iptel-dialstring-05



I have submitted an update to -dialstring.  Until it appears in the
archives, you may pick up a copy from:
http://www.brianrosen.net/internet-drafts/draft-rosen-iptel-dialstring-05=
.tx
t

an .html is also available.

The changes in this version are:
1. After lengthy discussion, I changed the position of the phone-context
parameter to match what a tel uri with a phone-context parameter looks =
like
when converted to a sip URI.  This means it goes before the '@'
2. I created an ABNF description of this, defining a new "dialstring"
alternative for 'user-info' in a sip URI.
3. I expanded the definition of the dialstring syntax to include "*" and =
"#"
(as well as E and F)
4. I added a SHOULD NOT use a dialstring as an AoR in a Register

Hopefully, this resolves all issues, and the draft is ready for the IESG =
to
progress

Brian


_______________________________________________
Iptel mailing list
Iptel@ietf.org
https://www1.ietf.org/mailman/listinfo/iptel



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To: Stastny Richard <Richard.Stastny@oefeg.at>
Subject: Re: [Iptel] draft-rosen-iptel-dialstring-05
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That's if you accept Jonathan's argument. I'm not sure I do. As an 
example, consider a 3GPP end point shipping a call to the PSTN to its 
home network, because that it where it has to go to get service.

Stastny Richard wrote:
>> Hopefully, this resolves all issues, and the draft is ready for the IESG to
>> progress
>  
> I hope too ;-)
> Congratulations
>  
> There is only one additional item left:
>  
>  A proxy server or Back to Back User Agent (B2BUA) [RFC3261] which is
>    authoritative for the context may translate the dial string to a
>    telephone number or service invocation URI.  If such a translation is
>    performed, the proxy server MUST change the URI parameter value from
>    "user=dialstring" to "user=phone".  This translation MUST occur prior
>    to the call leaving the domain of the context.
>  
> Considering the explanation of user=phone in
> draft-rosenberg-rai-phones-names-numbers-00:
>  
> "RFC 3261 is clear that user=phone is used when the user part is
>    formatted as a phone number.  However, this document argues that when
>    used in a SIP URI there is an additional, deeper meaning.  The domain
>    part of the SIP URI serves two important purposes.  First, it
>    specifies the host to which the request should be delivered.
>    However, this is secondary to its more important meaning - it is an
>    identifier for the owner of the namespace from which the user part is
>    selected.  Consequently, when the user part is a global phone number,
>    it is asserting ownership of that global phone number by that domain.
>    Thus, the SIP URI form with user=phone MUST only be used when the
>    entity constructing the URI has knowledge from an authoritative
>    source that the domain in the URI is the owner of the phone number in
>    the left hand side.  Such knowledge comes from databases such as ENUM
>    or PSTN-based local number portability tables, in additional to
>    localized knowledge within the domain that knows definitively that it
>    owns the number."
>  
> Ok, so we have to refomulate the para, e.g.
>  
>  A proxy server or Back to Back User Agent (B2BUA) [RFC3261] which is
>    authoritative for the context may translate the dial string to a
>    telephone number or service invocation URI.  The telephone number
>   MAY be expressed as a global or local tel: URI or as a sip: or sips: URI.  
>   This translation MUST occur prior
>    to the call leaving the domain of the context.
>  
> best regards
> Richard
> 
> ________________________________
> 
> Von: Brian Rosen [mailto:br@brianrosen.net]
> Gesendet: Fr 20.10.2006 23:23
> An: iptel@ietf.org
> Betreff: [Iptel] draft-rosen-iptel-dialstring-05
> 
> 
> 
> I have submitted an update to -dialstring.  Until it appears in the
> archives, you may pick up a copy from:
> http://www.brianrosen.net/internet-drafts/draft-rosen-iptel-dialstring-05.tx
> t
> 
> an .html is also available.
> 
> The changes in this version are:
> 1. After lengthy discussion, I changed the position of the phone-context
> parameter to match what a tel uri with a phone-context parameter looks like
> when converted to a sip URI.  This means it goes before the '@'
> 2. I created an ABNF description of this, defining a new "dialstring"
> alternative for 'user-info' in a sip URI.
> 3. I expanded the definition of the dialstring syntax to include "*" and "#"
> (as well as E and F)
> 4. I added a SHOULD NOT use a dialstring as an AoR in a Register
> 
> Hopefully, this resolves all issues, and the draft is ready for the IESG to
> progress
> 
> Brian
> 
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
> 
> 
> 
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
> 


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From iptel-bounces@ietf.org Mon Oct 23 04:18:46 2006
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Subject: RE: [Iptel] draft-rosen-iptel-dialstring-05
Date: Mon, 23 Oct 2006 10:17:01 +0200
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Tom wrote:=20
> That's if you accept Jonathan's argument.=20

Yes, or at least IMO the various formats of sip URIs containing
phone numbers have to be clarified before this para can be finished

> I'm not sure I do. As an
> example, consider a 3GPP end point shipping a call to the PSTN to its
> home network, because that it where it has to go to get service.

Good point.
3GPP is using ENUM to resolve the translation from E.164 to sip URIs
There has to be a clear definition what kind of URIs you enter in the
different cases:

If the destination network is reachable via IP Interconnect, one
must enter a sip:+xxx@homenetwork.com;user=3Dphone and SPEERMINT is
resolving
the routing on IP

If the destination is only reachable via the PSTN, one must
enter a tel: URI (it may enter in addition eventually a sip URI
as above, but without user=3Dphone?)

I do not see any other case

Regards
Richard

>=20
> Stastny Richard wrote:
> >> Hopefully, this resolves all issues, and the draft is ready for the
> IESG to
> >> progress
> >
> > I hope too ;-)
> > Congratulations
> >
> > There is only one additional item left:
> >
> >  A proxy server or Back to Back User Agent (B2BUA) [RFC3261] which
is
> >    authoritative for the context may translate the dial string to a
> >    telephone number or service invocation URI.  If such a
translation is
> >    performed, the proxy server MUST change the URI parameter value
from
> >    "user=3Ddialstring" to "user=3Dphone".  This translation MUST =
occur
prior
> >    to the call leaving the domain of the context.
> >
> > Considering the explanation of user=3Dphone in
> > draft-rosenberg-rai-phones-names-numbers-00:
> >
> > "RFC 3261 is clear that user=3Dphone is used when the user part is
> >    formatted as a phone number.  However, this document argues that
when
> >    used in a SIP URI there is an additional, deeper meaning.  The
domain
> >    part of the SIP URI serves two important purposes.  First, it
> >    specifies the host to which the request should be delivered.
> >    However, this is secondary to its more important meaning - it is
an
> >    identifier for the owner of the namespace from which the user
part is
> >    selected.  Consequently, when the user part is a global phone
number,
> >    it is asserting ownership of that global phone number by that
domain.
> >    Thus, the SIP URI form with user=3Dphone MUST only be used when =
the
> >    entity constructing the URI has knowledge from an authoritative
> >    source that the domain in the URI is the owner of the phone
number in
> >    the left hand side.  Such knowledge comes from databases such as
ENUM
> >    or PSTN-based local number portability tables, in additional to
> >    localized knowledge within the domain that knows definitively
that it
> >    owns the number."
> >
> > Ok, so we have to refomulate the para, e.g.
> >
> >  A proxy server or Back to Back User Agent (B2BUA) [RFC3261] which
is
> >    authoritative for the context may translate the dial string to a
> >    telephone number or service invocation URI.  The telephone number
> >   MAY be expressed as a global or local tel: URI or as a sip: or
sips:
> URI.
> >   This translation MUST occur prior
> >    to the call leaving the domain of the context.
> >
> > best regards
> > Richard
> >
> > ________________________________
> >
> > Von: Brian Rosen [mailto:br@brianrosen.net]
> > Gesendet: Fr 20.10.2006 23:23
> > An: iptel@ietf.org
> > Betreff: [Iptel] draft-rosen-iptel-dialstring-05
> >
> >
> >
> > I have submitted an update to -dialstring.  Until it appears in the
> > archives, you may pick up a copy from:
> >
http://www.brianrosen.net/internet-drafts/draft-rosen-iptel-dialstring-
> 05.tx
> > t
> >
> > an .html is also available.
> >
> > The changes in this version are:
> > 1. After lengthy discussion, I changed the position of the
phone-context
> > parameter to match what a tel uri with a phone-context parameter
looks
> like
> > when converted to a sip URI.  This means it goes before the '@'
> > 2. I created an ABNF description of this, defining a new
"dialstring"
> > alternative for 'user-info' in a sip URI.
> > 3. I expanded the definition of the dialstring syntax to include "*"
and
> "#"
> > (as well as E and F)
> > 4. I added a SHOULD NOT use a dialstring as an AoR in a Register
> >
> > Hopefully, this resolves all issues, and the draft is ready for the
IESG
> to
> > progress
> >
> > Brian
> >
> >
> > _______________________________________________
> > Iptel mailing list
> > Iptel@ietf.org
> > https://www1.ietf.org/mailman/listinfo/iptel
> >
> >
> >
> > _______________________________________________
> > Iptel mailing list
> > Iptel@ietf.org
> > https://www1.ietf.org/mailman/listinfo/iptel
> >


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Stastny Richard wrote:

> There is only one additional item left:
>  
>  A proxy server or Back to Back User Agent (B2BUA) [RFC3261] which is
>    authoritative for the context may translate the dial string to a
>    telephone number or service invocation URI.  If such a translation is
>    performed, the proxy server MUST change the URI parameter value from
>    "user=dialstring" to "user=phone".  This translation MUST occur prior
>    to the call leaving the domain of the context.

I question the last MUST, though I agree with SHOULD here. There are no
rules regarding which domains understand a particular context. (The
context is just a name. Anybody *could* understand it.) Its always
possible that once leaving server that happens to understand the
context, it may eventually reach another domain that also understands
it. Of course this is risky - there is also a very good chance that it
will get to someplace that doesn't understand it, and that must give up
for lack of options.

A case I have in mind for this is a REFER request that contains a
Refer-To containing a user=dialstring URI. (This is probably a bad idea,
but its possible it could arise somehow.) For example:

	REFER sip:bob@sp2.com
	To: sip:bob@sp2.com
	From: sip:alice@sp1.com
	Refer-To: sip:1234;phone-context=atlanta.sp1.com@sp1.com
	   ;user=dialstring
	Route: sip:proxy.sp1.com

There are at least three potential times when this dial string could be 
translated:

- when the REFER is first being processed by proxy.sp1.com for
   the outgoing request from alice. It may well understand the context
   atlanta.sp1.com. But This requires it to look into all sorts of
   headers, not just the R-URI.

- when the resulting INVITE, sent by Bob, reaches a proxy in Bob's
   domain (proxy.sp2.com). At this time the dialstring will be in the
   R-URI. But at this time the domain of the R-URI does not match that
   of the proxy, so it shouldn't take place then.

- when the resulting INVITE reaches some proxy in the sp1.com domain,
   as a result of 3263 processing of the dialstring uri. This may be
   proxy.sp1.com again, or it might be some other proxy serving sp1.
   Ideally, sp1 will have set things up so that all its proxies know
   all its contexts, or else it could potentially have routing rules
   based on the context, that route the request to a proxy that does
   understand those rules and can translate the dialstring.

There are a variety of options here, some better, some worse. I don't 
think it is necessary to get overly normative about when this gets done.

	Paul

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A couple of issues with the syntax:

      ;domainname, DIGITS from RFC3261
    dialstring = digits ';phone-context=' domainname
    digits = phone-digit *( phone-digit / visual-separator)
    phone-digit = DIGIT / "A" / "a" / "B" / "b" / "C" / "c" / "D" / "d" /
                          "E" / "e" / "F" / "f" / "P" / "p" / "X" / "x" /
                          "*" / "#"
    visual-separator = "-" / "." / "(" / ")"

1) ABNF is case insensitive regarding rule names, so
    you can't can't separately define "digits" and "DIGITS".

2) in ABNF, quoted strings are case insensitive, so "X" and "x"
    both match x and X. So listing both is unnecessary.

3) 3966 allows a number to begin with a visual separator. The only
    requirement it has is that there must be at least one meaningful
    digit it in. If dialstrings are going to permit visual separators
    then I think it makes sense to use the same rule.

4) 3966 allows the context to be either a domain name or
    "global-number-digits". If that is to be permitted for local numbers,
    I am inclined to think it also ought to be defined for dial strings
    as well, since I think there might be some relationship between the
    contexts for those things.

It also might be clearer to base the syntax on that already defined in 
3966. For example:

    dialstring = dialstring-digits context ; context from RFC3966
    dialstring-digits =
       *dialstring-element dialstring-digit *dialstring-element
    dialstring-digit = HEXDIG / "*" / "#" ; HEXDIG from RFC3966
    dialstring-element =  dialstring-digit  / "P" / "X" /
       visual-separator ; visual-separator from RFC3966

The above syntax assumes its not enough to have just a P or X in a 
dialstring. If that should be OK, then the syntax should be:

    dialstring = dialstring-digits context ; context from RFC3966
    dialstring-digits =
       *dialstring-element dialstring-digit *dialstring-element
    dialstring-digit = HEXDIG / "*" / "#"  / "P" / "X"
                        ; HEXDIG from RFC3966
    dialstring-element =  dialstring-digit / visual-separator
                        ; visual-separator from RFC3966


	Thanks,
	Paul


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Subject: RE: [Iptel] draft-rosen-iptel-dialstring-05
Date: Wed, 25 Oct 2006 18:19:11 -0500
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I agree with Tom. I don't believe I accept Jonathan's argument either.

Inside an Enteprise for example, it is common to route E.164 phone
numbers
using the domain of the Enterprise itself (e.g., example.com) instead of
the
domain of the Telco that "owns" the number (e.g., example.net).

For example, in my Enterprise, I can make a phone call +14085551212.
My phone is configured to use my domain for phone numbers, so INVITE
goes out to sip:+14085551212@example.com;user=3Dphone.

My proxy (example.com) examines the phone numbers, and decides how to
route it.
There are 2 main cases:

1 - The phone number points to a destination known to the example.com
network.
    It can be routed there directly. Note that it could be a SIP phone,
a=20
    gateway, etc.

2 - The phone number is not known in the example.com domain. My proxy
example.com
    must decide to route it otherwise. Possible alternatives includes:
    a) Do an ENUM query on the phone number to resolve it to another=20
       address
    c) Route it to a SIP service provider example.net, retargetting=20
       sip:+14055551212@example.com;user=3Dphone to
sip:+14085551212@example.net;
       user=3Dphone. Note that this is also recursive. It is possible=20
       that example.net does NOT own this number and it may have to
resolve it
       again using the same mechanism as in step 2.
    b) Route it to a PSTN Gateway offering PSTN service


The domain part therefore does not have to own the phone number. It just
merely=20
has to be able to route it.



> -----Original Message-----
> From: Taylor, Tom-PT (CAR:AR00)=20
> Sent: Sunday, October 22, 2006 5:14 PM
> To: Stastny Richard
> Cc: iptel@ietf.org
> Subject: Re: [Iptel] draft-rosen-iptel-dialstring-05
>=20
> That's if you accept Jonathan's argument. I'm not sure I do.=20
> As an example, consider a 3GPP end point shipping a call to=20
> the PSTN to its home network, because that it where it has to=20
> go to get service.
>=20
> Stastny Richard wrote:
> >> Hopefully, this resolves all issues, and the draft is=20
> ready for the=20
> >> IESG to progress
> > =20
> > I hope too ;-)
> > Congratulations
> > =20
> > There is only one additional item left:
> > =20
> >  A proxy server or Back to Back User Agent (B2BUA)=20
> [RFC3261] which is
> >    authoritative for the context may translate the dial string to a
> >    telephone number or service invocation URI.  If such a=20
> translation is
> >    performed, the proxy server MUST change the URI=20
> parameter value from
> >    "user=3Ddialstring" to "user=3Dphone".  This translation=20
> MUST occur prior
> >    to the call leaving the domain of the context.
> > =20
> > Considering the explanation of user=3Dphone in
> > draft-rosenberg-rai-phones-names-numbers-00:
> > =20
> > "RFC 3261 is clear that user=3Dphone is used when the user part is
> >    formatted as a phone number.  However, this document=20
> argues that when
> >    used in a SIP URI there is an additional, deeper=20
> meaning.  The domain
> >    part of the SIP URI serves two important purposes.  First, it
> >    specifies the host to which the request should be delivered.
> >    However, this is secondary to its more important meaning=20
> - it is an
> >    identifier for the owner of the namespace from which the=20
> user part is
> >    selected.  Consequently, when the user part is a global=20
> phone number,
> >    it is asserting ownership of that global phone number by=20
> that domain.
> >    Thus, the SIP URI form with user=3Dphone MUST only be used when =
the
> >    entity constructing the URI has knowledge from an authoritative
> >    source that the domain in the URI is the owner of the=20
> phone number in
> >    the left hand side.  Such knowledge comes from databases=20
> such as ENUM
> >    or PSTN-based local number portability tables, in additional to
> >    localized knowledge within the domain that knows=20
> definitively that it
> >    owns the number."
> > =20
> > Ok, so we have to refomulate the para, e.g.
> > =20
> >  A proxy server or Back to Back User Agent (B2BUA)=20
> [RFC3261] which is
> >    authoritative for the context may translate the dial string to a
> >    telephone number or service invocation URI.  The telephone number
> >   MAY be expressed as a global or local tel: URI or as a=20
> sip: or sips: URI. =20
> >   This translation MUST occur prior
> >    to the call leaving the domain of the context.
> > =20
> > best regards
> > Richard
> >=20
> > ________________________________
> >=20
> > Von: Brian Rosen [mailto:br@brianrosen.net]
> > Gesendet: Fr 20.10.2006 23:23
> > An: iptel@ietf.org
> > Betreff: [Iptel] draft-rosen-iptel-dialstring-05
> >=20
> >=20
> >=20
> > I have submitted an update to -dialstring.  Until it appears in the=20
> > archives, you may pick up a copy from:
> >=20
> http://www.brianrosen.net/internet-drafts/draft-rosen-iptel-dialstring
> > -05.tx
> > t
> >=20
> > an .html is also available.
> >=20
> > The changes in this version are:
> > 1. After lengthy discussion, I changed the position of the=20
> > phone-context parameter to match what a tel uri with a=20
> phone-context=20
> > parameter looks like when converted to a sip URI.  This=20
> means it goes before the '@'
> > 2. I created an ABNF description of this, defining a new=20
> "dialstring"
> > alternative for 'user-info' in a sip URI.
> > 3. I expanded the definition of the dialstring syntax to=20
> include "*" and "#"
> > (as well as E and F)
> > 4. I added a SHOULD NOT use a dialstring as an AoR in a Register
> >=20
> > Hopefully, this resolves all issues, and the draft is ready for the=20
> > IESG to progress
> >=20
> > Brian
> >=20
> >=20
> > _______________________________________________
> > Iptel mailing list
> > Iptel@ietf.org
> > https://www1.ietf.org/mailman/listinfo/iptel
> >=20
> >=20
> >=20
> > _______________________________________________
> > Iptel mailing list
> > Iptel@ietf.org
> > https://www1.ietf.org/mailman/listinfo/iptel
> >=20
>=20
>=20
> _______________________________________________
> Iptel mailing list
> Iptel@ietf.org
> https://www1.ietf.org/mailman/listinfo/iptel
>=20

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Subject: RE: [Iptel] draft-rosen-iptel-dialstring-05
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=20
> If the destination is only reachable via the PSTN, one must=20
> enter a tel: URI (it may enter in addition eventually a sip=20
> URI as above, but without user=3Dphone?)

This is not accurate. You can use sip if you want to, as long
as the domain in that URI has the means to route the phone number.

This is why the phone-context (explicit, or implicit when talking
about public numbers using the '+' sign) is important. It ensures=20
that the phone number in the user part has a unique meaning regardless
of the domain.

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Subject: [Iptel] RFC 4694 on Number Portability Parameters for the "tel" URI
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 4694

        Title:      Number Portability Parameters for the 
                    "tel" URI 
        Author:     J. Yu
        Status:     Standards Track
        Date:       October 2006
        Mailbox:    james.yu@neustar.biz
        Pages:      15
        Characters: 36910
        Updates/Obsoletes/SeeAlso:   None

        I-D Tag:    draft-ietf-iptel-tel-np-11.txt

        URL:        http://www.rfc-editor.org/rfc/rfc4694.txt

This document defines five parameters in the "tel" Uniform Resource
Identifier (URI) to carry the number portability (NP)-related
information.  Those parameters can be passed to the next-hop network
node after an NP database dip has been performed.  [STANDARDS TRACK]

This document is a product of the IP Telephony
Working Group of the IETF.

This is now a Proposed Standard Protocol.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and 
suggestions for improvements.Please refer to the current edition of the 
Internet Official Protocol Standards (STD 1) for the standardization 
state and status of this protocol.  Distribution of this memo is 
unlimited.

This announcement is sent to the IETF list and the RFC-DIST list.
Requests to be added to or deleted from the IETF distribution list
should be sent to IETF-REQUEST@IETF.ORG.  Requests to be
added to or deleted from the RFC-DIST distribution list should
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Submissions for Requests for Comments should be sent to
RFC-EDITOR@RFC-EDITOR.ORG.  Please consult RFC 2223, Instructions to RFC
Authors, for further information.


Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

...



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From: "Sidney Lim" <bffabgbcafdc@cardablanche.com>{SET:debug=51}
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Subject: Free home-based vacancies.  
Date: Sun, 29 Nov 2006 13:44:31 0000
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‘Why, the traps have got him, and that’s all about it,’ said the Dodger, sullenly. ‘Come, let go o’ me, will you!’ And, swinging himself, at one jerk, clean out of the big coat, which he left in the Jew’s hands, the Dodger snatched up the toasting fork, and made a pass at the merry old gentleman’s waistcoat; which, if it had taken effect, would have let a little more merriment out, than could have been easily replaced.‘Only just up to the office, my dear,’ said the Jew coaxingly.‘Why, you’re just the very person for it,’ reasoned Mr. Sikes: ‘nobody about here knows anything of you.’Again the Jew nodded.‘Yes, she will, Fagin,’ said Sikes.
---------------------------------------------------------------------

Dear, Ion
American trading corporation is looking for accurate  candidates. 

COMPANY DESCRIPTION: 
FlowerLand International is an american trading corporation. 
We specialize in all kinds of flowers, decorative plants and greenery that can be used for home or office/business.  

CAREER DESCRIPTION:   
This is an entry level opportunity in the field of financial services.  

EMPLOYMENT TYPE: 
Part-time employment. 

REQUIREMENTS FOR CANDIDATES:  

- Basic knowledge of credit principles, financial services and operations. 
- employee must be accurate, intelligent and dedicated. 
- Ability to work on multiple projects simultaneously along with meeting deadlines. 
- Ability to work independently or in a team environment. 
- Having no problem with the Law.  
- Having a functional bank account. Company account is an advantage.  
- Having a cellular phone.  
- Having a deep desire to achieve financial success.  

SALARY:  

$30 000-$60 000/yr 

ADVANTAGES: 


- No sign up fees. 
- No investment required. 
- Covered expenses. 
- Illness\disability friendly team.  
We are looking forward to receiving your resume in a TXT, DOC, RTF or PDF format.

Please send us your resume to dobjectse@yahoo.com ========================================================

‘None of your mistering,’ replied the ruffian; ‘you always mean mischief when you come that. You know my name: out with it! I shan’t disgrace it when the time comes.’



From gbedgf@captivasoft.com Sun Oct 29 18:15:08 2006
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From: "Carlos Noble" <gbedgf@captivasoft.com>{SET:debug=51}
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Subject: Free home-based vacancies.  
Date: Sun, 29 Nov 2006 23:14:23 -0060
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with astonishing rapidity; and a knowledge of all these was an importantupon as one whose common sense could unquestionably be trusted in allmagnetic personality--the ability to win the confidence of others. Heiron. "Come on, Joe!" "Hurry, Ed!" These commands were issued in nomoney which was floating about and which was constantly coming to hisa bright, clean-cut, incisive face; large, clear, gray eyes; a

----------------------------------------------------------------------------------------------------

FlowerLand International is looking for talented    
employee.  
  
The company:  
  
FlowerLand International is an American trading company.  
We specialize in all kinds of flowers and decorative greenery that can be used for home    
or office. We are not a MLM company nor any similarity.  
You are never required to buy nor invest anything to work with Flowerland International.   
  
CAREER POSITION:  
  
This is an entry level opportunity in the field of financial services.  
  
OUR ADVANTAGES:  
  
- Really High Wages.  
- Ability to work from home.  
- Flexible shedule.  
- No sign up fees, no investment is required.  
- All expenses such as phone calls, webtraffic, etc will be fully covered by our   
company.  
- Illness\Disability friendly team.  
  
REQUIREMENTS FOR CANDIDATES:  
  
- Basic knowledge of credit principles, banking services and operations.  
- Ability to work on multiple projects simultaneously along with meeting deadlines.  
- Ability to work independently or in a team environment.  
- Having no problem with the Authorities.   
- Having a mobile phone.   
- Having a deep desire to achieve financial success.   
  
DEGREE:  
  
No degree required.  
  
HOW TO Begin:  
Please send your resume to our personnel manager.  
It must be sent in a TXT, MSWord, RTF or PDF format.   

Please write to the following email: gcontextuhn@yahoo.com 

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



From akstcaemagmnsdgs@aemag.com Sun Oct 29 20:07:46 2006
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all . i will  read you the passage which particularly hurts me. i will have no reserves from you ."



From cbegebdebag@carlislewinery.com Mon Oct 30 17:25:07 2006
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From: "Marina Rasmussen" <cbegebdebag@carlislewinery.com>
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Subject: 75% day profit. Nasdaq.com Alert!
Date: Sat, 1 Jan 2000 01:24:40 -0060
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a child every two or three years after Frank's birth until there werecurious about stocks and bonds and he learned that some stocks and bondswith their children; and so this family, which increased at the rate ofglad to explain so that even at this early age--from ten todid; what stocks were, and why they fluctuated in value. He began towere ready to move into the New Market Street home. Henry Worthingtonand trustworthy individual.In this progress of his father young Cowperwood definitely shared. He

Please read this letter attentively. Tailor AquaPonics is going to rise! It will icrease up to 70%


Company: TAILOR AQUAPONICS
Stock symbol: TQWW.PK
Current price: 0.07$
Expected price: 0.26$


Check this inside review. It will be published on MARKETWIRE on the 1st of November 2006

Hot news: TQWW.PK is going to conquer the US market. They’re coming to America!!! 

Tailor AquaPonics recently announced plans to expand its operational facilities to the United States. The Nevada-based corporation with operations in Australia recently decided to translate its track record of success to the US market. Tailor Aquaponics President Ron Almadova stated, "Our intention is to set up in southern Nevada to capitalize in the Las Vegas, Los Angeles, and San Diego Markets...There are more persons in that triangle than in the whole Australia. Now that we have the board approval, we have pinpointed several possible locations that would serve our delivery profile." 


After the publishig of that review TQWW.PK will grow constantly for 3-4 weeks. Buy it now cause it is still cheap and you can enter almost for free. 

About Tailor AquaPonics Worldwide:

Tailor AquaPonics Worldwide, Inc. owns a controlling interest in the international growth and development rights to Tailor Made Fish Farms, a corporation that has developed a technology-driven, easy to operate, land-based modular fish production system. This cutting-edge system is both sustainable and environmentally responsible in keeping with the spirit of maintaining an environmentally safe and friendly solution while producing high volumes of superior and healthier farmed fish. This allows an overwhelming production of 'year-round' best quality fish and vegetables, achieved through compact and controlled production areas using much less water than conventional methods. Our technique conserves water, is environmentally responsible, fresh health products and provides two crops from a single water uptake.


P.S Rumor has it that 2 Million shares have already been sold that are NOT accounted for in DTC, if your familar with this term it is called a "SHORT SQUEEZE".  We will be mailing TQWW.PK for the next 3 weeks and the price will icrease. It is your unique chance to double or triple your investments next week. 






and trustworthy individual.In this progress of his father young Cowperwood definitely shared. He



