freenum.org - a suggestion to examine a new communications pointer method

[jtodd]

Hello -
I was referenced to this group by Steve Song. I'd like to bring to
your attention some of the benefits of taking a look at the numbering
methods provided by freenum.org for various components of the Village
Telco project. Freenum provides a globally-unique, telephone-friendly
numbering plan that scales extremely well and is more compatible with
Internet-based communications concepts while still allowing for
integration into legacy PSTN switching networks in circumstances where
carriers are appropriately forward-thinking.

To give you a quick summary of what Freenum provides, it's easier to
describe how one would use it. My Freenum number is 2203*256. The
"2203" part happens to be the extension on my PBX; it could be any
number of digits, but I have 4 digit dialing in my system. The "256"
portion is a globally unique number that was allocated to my
"organization" by IANA, much like a domain name except that I had no
choice in the number that was given to me; they are allocated
sequentially. Anyone who is using the ISN (aka: Freenum) dialing
system can dial that number. Typically they would have their system
connected to the Internet somehow if they wished to reach me. A DNS
lookup would then be done by their switching system on
3.0.2.2.256.freenum.org and an NAPTR would be returned. In this
example, a SIP NAPTR would be what the caller's switch is looking
for. The SIP NAPTR that would be returned would map to "sip:
22...@loligo.com" which in turn would create a call via IP to my PBX
platform, ringing extension 2203.

I won't go into the complex and extraordinarily flexible nature of
the system, since that would take a lot more characters than perhaps
you're interested in. But I'd be happy to start a discussion on this
in more detail if anyone thinks this might be interesting for your
user community. I think that the question of pointers is a
significant one, especially when attempting to reconcile the
shortcomings of E.164 while still retaining handset compatibility for
the vast majority of the world's telephony devices.

Here are a few reasons you might be interested in Freenum for
Village Telco puroposes:

1) Freenum is standards-based. Any client that can do ENUM resolution
lookups will work. There are even extensions to many ENUM resolver
libraries that support freenum.org style lookups without doing any
regular expression re-writing (Asterisk has it built in, SER has a
module, the other SER-like projects are easily adapted or have it
already.)

2) Freenum supports many modalities. Usually it runs with SIP
endpoints. But it can just as easily map a numeric identity to chat,
mail, SMS, or even web pages. Anything that is supported in the NAPTR
style lookup scheme can be associated with a "number".

3) Even in non-connected isolated entities, it may make sense to use
Freenum-style dialing strings for eventual integration into larger
networks. Getting an ITAD (aka: "numeric domain name") may make sense
even before connection.

4) Freenum is unencumbered by local or international artificial
scarcity problems. There are no fees for use, or registration, or
lookups, and it is the intention to never have any such fees.


Again, I'm available for any discussions of the concept if anyone is
interested further. We have about 1000 organizations now using the
system at the moment, from 60 nations.

JT

---
John Todd - Advocate - jt...@freenum.org
ISN: 2203*256 tel: +1-301-561-1020

[Antoine van Gelder]

On 08 Dec 2009, at 07:21 , jtodd wrote:
> I was referenced to this group by Steve Song. I'd like to bring to
> your attention some of the benefits of taking a look at the numbering
> methods provided by freenum.org for various components of the Village
> Telco project. Freenum provides a globally-unique, telephone-friendly
> numbering plan that scales extremely well and is more compatible with
> Internet-based communications concepts while still allowing for
> integration into legacy PSTN switching networks in circumstances where
> carriers are appropriately forward-thinking.

...

> Again, I'm available for any discussions of the concept if anyone is
> interested further. We have about 1000 organizations now using the
> system at the moment, from 60 nations.

An ITAD can be requested by following these instructions:

http://www.freenum.org/cookbook/#requesting-an-itad-for-your-domain

IANA website says it will take around a week to process applications.

I've submitted mine.

Anyone else want to send in an application so I'll have someone to phone?

:-D

- a

[John Todd]

If you want some numbers to test with, here's a post for "dead-end" numbers that go to echo, or to the date/time, or to screaming monkeys:

https://mail.internet2.edu/wws/arc/isn-discuss/2009-10/msg00003.html

Typically, I let people try calling 1234*256, which is my army of monkeys followed by an echo, but there are many test numbers out there.  Note that you don't need to have an ITAD to use Freenum for outbound calling, and you don't even need to have Freenum/ISN style resolution in your SIP server.  Just do "1234*2...@freenum.org" for example and our SIP platform will kick back a 302 Redirect for the top SIP URI in the NAPTR list (which is nearly 100% of the time correct since I don't know of anyone who uses more than one NAPTR reply currently.)

JT

 

[Steve Song]

On Tue, Dec 8, 2009 at 9:51 AM, Antoine van Gelder
<ant...@7degrees.co.za> wrote:
[snip]

> I've submitted mine.
>
> Anyone else want to send in an application so I'll have someone to phone?
>
> :-D

Application now pending for villagetelco.org :-)

-Steve

[John Todd]

Great news!  

So there are two options as to how you can enable this service on your "client" (customer? users? not sure what your terminology is) endpoints.   You could (though it is ill-advised) allow them to all share the villagetelco ITAD (organization number) and then create some sort of automated assignment system centrally to allocate extension numbers to each node, and then install the DNS in one of your servers to point the queries to each organization.  This would be fairly complex to build the configuration tools, and would not be flexible for the end users, but would ease the installation process and eliminate administrative confusion.  The other method would be to give written instructions on how to obtain an ITAD and register it with the freenum.org system, and then allow the administrator to put in their ITAD somewhere in the system in a fill-in box easily.

I have a question on your platform in general: how are user "numbers" allocated, typically?  Four, five, ... digit extensions?  E.164 addresses?  Usernames?

I know it's early to start talking about how you'd roll this out (if you choose to at all) but it's worth at least pondering the options for a while.

JT

[Alberto Escudero]

Hi John,

Nice to meet the "screaming monkeys" :) that I have dialed several times.
From the billing point of view, we will like the local extensions to have
a form like

3+4 digits. Being the first 3 digits the Village Telco "installation code"
(our wizard ask for a 3 digit prefix for sequence of numbers) . We are now
in the right time to start discussing this anyway. Maybe we should just
try to emulate as close as possible freenum.org.

I have no bias. But local calls do not need to rely on outside Internet
at all. That this means that we need to include a local DNS sever?

/aep

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[David Rowe]

Hello List,

A simple test of the MP01 got me thinking about the challenges of making
the Village Telco easy to use.

I've set up a MP01 with a small mesh network as an every day phone
for my house. A rather obvious test - can the thing be used for phone
calls every day over an extended period? I'm using an IP04 as a gateway
to the PSTN, with this configuration:

PSTN - IP04 - LAN - Nanostation - mesh Wifi - MP01 - Analog phone

The IP04 also has a few analog phones hanging off it so we can use the
system to make calls between rooms in the house.

Here were the steps:

1/ The mesh network has the IP range of 10.130.1.x, the LAN 10.30.1.x.
So I needed to set up an IP04 Ethernet interface on the 10.30.1.x
network and a route to the mesh network via the Nanostation:

[ip04]$ ifconfig eth0 10.30.1.4 netmask 255.255.255.0;
[ip04]$ route add -net 10.130.1.0/24 gw 10.30.1.1;

Batman on the Nanostation 2 was configured to export the 10.30.1.x
network:

[nanostation2]$ batmand.3.2 -a 10.30.1.0/24 -g 5000 ath0

This automatically set up routing on the MP01. I checked routes in both
direction using ping.

2/ Set up a SIP trunk between the IP04 and MP

Here is the /etc/asterisk/sip.conf entry on the IP04:

[potato]
type=peer
username=potato
fromuser=potato
secret=password
context=default
disallow=all
qualify=500
dtmfmode=rfc2833
callerid=server
canreinvite=no
host=10.130.1.38
allow=ulaw

It's the same on the MP01 except for:

host=10.3.1.4

Then on both Asterisks check the SIP trunk:

ip04*CLI> sip reload
ip04*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
potato/potato 10.130.1.38 5060 OK (6
ms)

3/ Some dialplan on the Mesh Potato, /etc/asterisk/extensions.conf:

[default]
exten => 4000,1,Dial(MP/1)
eexten => _3,1,Dial(SIP/potato/6003)
exten => _4,1,Dial(SIP/potato/6004)
exten => _0,1,DIal(SIP/potato/6001)

Dialling 0 on the MP will take IP04 extension 6001 (connected to the
PSTN line) off hook. The MP can then dial a PSTN number, or answer an
incoming PSTN call. Dialling 3 & 4 make IP04 FXS extensions connected
to analog phones ring.

Extension 4000 calls the MP01's FXS phone, used below by the IP04.

4/ Some dial plan on the IP04:

[default]
exten => 4000,1,Dial(SIP/potato/4000)

So dialling 4000 on an IP04 phone will place a call to the MP01.

To check the dialplan entries:

CLI> dialplan reload
CLI> show dialplan default

So here is the point of this post:

The simple example above took a lot of thought and involved an
understanding of IP routing, the Linux command line, vi, Asterisk
dialplan and SIP configuration. The challenge for us is to make this
really easy to do by people with the minimum possible skill level.

Diagnostics are important, we need easy ways to check that the IP links
are up, that SIP connectivity exists, and alarms if anything goes down.
Preferably without using the Linux command line, or vi, or the Asterisk
CLI.

The "guy in a village" may not have access to the Internet. How will he
support himself and diagnose a problem if something goes wrong?

Ease of use is a significant challenge, not to be underestimated. One
example that comes to mind is the enormous amount of work that has gone
into GUIs to make Asterisk easier to use. Another example: in the
developed world many small offices still pay "the phone guy" to maintain
their small phone systems, even though they might administer their IT
themselves.

We have made some steps in the right direction - Alberto and Louise are
doing some fine work on an easy to set up Asterisk/Billing server,
Afrimesh is looking good. Elektra's latest firmware images allow phone
calls between MPs with zero configuration - just apply power and dial
the IP of another MP.

BTW I have discovered an novel use for the Mesh Potato. When my wife
woke up today she used it to call me in my home-office. She wanted me
to make her some coffee!

Cheers,

David

[victor tejada]

DIOS te Bendiga y Guarde --- On Sat, 12/12/09, David Rowe <da...@rowetel.com> wrote:

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[Akshay Mishra]

On 13 December 2009 04:31, David Rowe <da...@rowetel.com> wrote:
> Hello List,
>
> A simple test of the MP01 got me thinking about the challenges of making
> the Village Telco easy to use.

Hello,

My 2 MPs talk to each other across walls. I can communicate and talk
and dial. As yet, the quality is acceptable. I will try with longer
distances over weekend. This is the first time I am hacking asterisk
and it is fun (slightly non-intuitive until one gets the hang). Mails
from Elektra/David gave a good start too.

However, when I through call my cell phone (registered on the asterisk
server, it has wi-fi), i can dial and get a call, but there is no
audio in either direction.

Any reason why this should happen ?

regards,
Akshay

[Alexander Chemeris]

Akshay,

On Wed, Feb 10, 2010 at 05:56, Akshay Mishra <akshay...@gmail.com> wrote:
> However, when I through call my cell phone (registered on the asterisk
> server, it has wi-fi), i can dial and get a call, but there is no
> audio in either direction.
>
> Any reason why this should happen ?

It may be a codec compatibility problem, so check:
What codecs are configured on your phone?
What codecs are enabled in your Asterisk setup?
(MPs use GSM-FR, IIRC)

--
Regards,
Alexander Chemeris.

[Akshay Mishra]

Thanks Alexandar,
Also, now since Asterisk supports google talk, can
we also not configure MPs to ring when somebody on google calls me ? I
shall try this out if not already done.

regards,
Akshay

On 10 February 2010 10:35, Alexander Chemeris