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lists at minotaur.cc
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Posted: Fri Apr 04, 2008 2:19 pm Post subject: [asterisk-biz] "on-net" dialplans

Greetings list, As many of us on the list operate networks with large numbers of otherwise unconnected customers, I thought I'd canvass opinion on how you number your customers for "internal" calls between them. There have been some attempts in recent years to simplify this with things like ITAD/Freenum (http://freenum.org/) and probably plenty of others. What sort of market penetration have those schemes achieved? What do you do in your network in terms of allowing calls between clients without dialling the full e164 number? Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz

acunningham at integri...
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Posted: Fri Apr 04, 2008 2:29 pm Post subject: [asterisk-biz] "on-net" dialplans

Chris Bagnall wrote:

Quote:

Greetings list,

As many of us on the list operate networks with large numbers of otherwise unconnected customers, I thought I'd canvass opinion on how you number your customers for "internal" calls between them.

There have been some attempts in recent years to simplify this with things like ITAD/Freenum (http://freenum.org/) and probably plenty of others. What sort of market penetration have those schemes achieved?

What do you do in your network in terms of allowing calls between clients without dialling the full e164 number?

Regards,

Chris

Chris,

In Enswitch (at least in the default configuration), users can dial
other users' SIP accounts directly, but in our experience only a very
tiny fraction choose to do so, and almost all users choose to dial the
full PSTN number, even though it's simply looped back within the system.
We haven't done any research to find out why, but I suspect it's simply
down to customers not being aware of which other customers are on the
same system and/or the effort of remembering both a DID and an internal
number. I'd therefore say at this point that schemes to publish internal
numbers are not worth the effort.

Alistair Cunningham
+1 888 468 3111
+44 20 799 39 799
sip:acunningham@integrics.com
http://integrics.com/

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nk3569 at yahoo.com
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Posted: Fri Apr 04, 2008 5:02 pm Post subject: [asterisk-biz] "on-net" dialplans

--- Alistair Cunningham <acunningham@integrics.com> wrote:

Quote:

I'd therefore say at this point that schemes to publish
internal numbers are not worth the effort.

I tend to agree.. maybe in a few years.

Right now, 99.9% of our internal users calling each other do so by the
PSTN number. Our outgoing dialplan just defaults to connect it directly
if available, and if not sends it out to a trunk.

We do have the option to dial someone by their internal number instead,
but I don't think anybody uses it. Just included it because it's
cheap/free to provide and it allows people to forward SIP calls to
their already-connected ATA.

Would be nice if most VSP's joined and created a database to route
calls to each other via SIP instead of PSTN.. but that would take a lot
more than wishful thinking to achieve... Smile

--
Nitzan Kon, CEO
Future Nine Corporation

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m345 at netvision.net.il
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Posted: Sat Apr 05, 2008 2:32 pm Post subject: [asterisk-biz] "on-net" dialplans

Nitzan
such databases and services already exist
such as those from Xconnect and Telx

From what I understand, some of the biggest carriers use them, thus bypassing the PSTN.

FWD I think also has peering arrangements with voip providers.

I would look in those directions first, before reinventing the wheel....

Moshe

Nitzan Kon wrote:

Quote:

--- Alistair Cunningham <acunningham@integrics.com> (acunningham@integrics.com) wrote:

Quote:

I'd therefore say at this point that schemes to publish
internal numbers are not worth the effort.

trixter at 0xdecafbad.com
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Posted: Sun Apr 06, 2008 10:02 am Post subject: [asterisk-biz] "on-net" dialplans

On Sat, 2008-04-05 at 22:25 +0300, Moshe Maeir wrote:

Quote:

Nitzan
such databases and services already exist
such as those from Xconnect and Telx

For the US at least, currently just over 10% of phones (landline and
mobile) are voip connected, carriers such as verizon, cingular, global
crossing, qwest and others have connected their PSTN customers to a voip
gateway, and rates are generally much better than if you call via the
pstn for those destinations.

Of course peering with those companies is a whole different deal, they
arent required to peer, and generally they dont want every small
wanna-be provider peering for super cheap next to free calling, but if
you are reasonably sized it shouldnt be that difficult and you can then
have a much lower termination cost.

In many situations there are enum servers that are used to allow for
lookups given that there are porting issues and such that need to be
taken care of.
--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
The GPL is a software license not a religion

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andy at nosignal.org
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Posted: Sun Apr 06, 2008 4:39 pm Post subject: [asterisk-biz] "on-net" dialplans

On 6 Apr 2008, at 15:55, Trixter aka Bret McDanel wrote:

Quote:

Of course peering with those companies is a whole different deal,
they arent required to peer, and generally they dont want every
small wanna-be provider peering for super cheap next to free
calling, but if you are reasonably sized it shouldnt be that
difficult and you can then have a much lower termination cost.

Peering doesn't have to mean settlement free, and in fact it rarely
does in the telecoms world.

Increasing the "meshability" of your telecoms edge might have little
to no effect on your average termination cost, because you might pay a
peer what you pay your regional or a-z provider, but it stands a good
chance of improving your sounds quality or call setup time
performance, so any financial incentive to peer comes from increased
revenues from satisfied customers and a higher ACD.

Lastly, replacing A-Z style termination that relies on incumbent/
legacy legs, with lots of short, IP-only paths between end-sites is
the only way we're going to get widespread adoption of new
technologies like wideband audio codecs.

I have various roles in organisations that mean I spent a lot of time
thinking about both IP and VoIP interconnection, and would really like
to talk with other people who do as well. Perhaps this is the scope
for another discussion list !

Best wishes
Andy Davidson
www.devonshire.it

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trixter at 0xdecafbad.com
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Posted: Sun Apr 06, 2008 5:32 pm Post subject: [asterisk-biz] "on-net" dialplans

On Sun, 2008-04-06 at 22:35 +0100, Andy Davidson wrote:

Quote:

On 6 Apr 2008, at 15:55, Trixter aka Bret McDanel wrote:

Quote:

Peering doesn't have to mean settlement free, and in fact it rarely
does in the telecoms world.

I never used the word 'free' by itself, I instead intentionally used
terms like "super cheap next to free". I never intended for anyone to
think that actually meant free, specifically since I did not say that.

--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
The GPL is a software license not a religion

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andy at nosignal.org
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Posted: Mon Apr 07, 2008 1:25 am Post subject: [asterisk-biz] "on-net" dialplans

On 6 Apr 2008, at 23:23, Trixter aka Bret McDanel wrote:

Quote:

That's fine. I was presenting the advantages of mass-interconnection
as I see them. You'll note that pricing didn't come into any of my
plus-points. I think this means we effectively we are in agreement,
so don't be upset.

Thanks,
Andy

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trixter at 0xdecafbad.com
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Posted: Mon Apr 07, 2008 1:53 am Post subject: [asterisk-biz] "on-net" dialplans

On Mon, 2008-04-07 at 07:19 +0100, Andy Davidson wrote:

Quote:

On 6 Apr 2008, at 23:23, Trixter aka Bret McDanel wrote:

Quote:

didnt realize I was, I will be more careful next time to not get upset
when giving an explanation. Again, I am sorry for getting upset, as you
can see from what I said I really let you have it, and that just isnt
fair to you.

However, we arent in complete agreement since many phone companies are
going to a "bill and keep" method, which generally does completely
counter what you said when you were "correcting" me saying that I said
that it was free.

For about a decade SBC has been trying to push that as the default,
unless of course you are pushing more traffic to them, their
interconnection agreement (as in phone company to phone company
agreement) gives them 30 day outs to bill you, but you do not have the
same privilege. Quite biased if you ask me, although you dont have to
take that agreement it just becomes more costly and takes considerably
longer to get some other deal.

This is also not just a US thing, BT in the UK does a voip interconnect
where they pay for minutes they terminate onto your network, you pay for
ones that you terminate, and generally you are required to have more or
less symmetric traffic so it ends up being free. The symmetric traffic
rule is why they dont do this as a wholesale product, oftel/ofcom
wouldnt like that too much there, but they offer it as a retail one so
they can do that.

But hey if you do have symmetric traffic at least your UK calling is
free right?

--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
The GPL is a software license not a religion

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jweisman at ibell.net
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Posted: Mon Apr 07, 2008 2:11 pm Post subject: [asterisk-biz] "on-net" dialplans

Quote:

Would be nice if most VSP's joined and created a database to route
calls to each other via SIP instead of PSTN.. but that would take a lot
more than wishful thinking to achieve... Smile

--
Nitzan Kon, CEO
Future Nine Corporation

Isnt that what voip peering is all about?

-Jon

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