VOIP / SIP gateway in New Zealand?

VOIP / SIP gateway in New Zealand?

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Subject Author Date
VOIP / SIP gateway in New Zealand? smooter.guy 02-24-2006
Posted by on February 24, 2006, 3:45 am
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.. pls help me get my head around this.

Business has a 12 month old NEC IPS 2000 PABX with voip capability (use
IP cards for trunk to trunk calling over frame relay to remote office
with same NEC PABX model, also use NEC softphones successfully).
GM now asking for us to set up the existing NEC IPS to utilise Voip
calls in addition to PSTN calls to overseas customers and suppliers as
a cheaper alternative.

So I figure we need these 3 things:

1. A decent internet feed - Tick, we have 5mbit Ethernet feed.
2. Some sort of hardware gateway to talk to a VOIP provider (where do I
start? Cisco?)
3. A VOIP provider that bills us for calls at less than the local
Telco.

Am I on the right track here?
Does such a VOIP provider exist down here in NZ? Could I use overseas
prov instead?
How do callers make VOIP calls back to you (e.g a staff member being
contactable at their desk phone via a Voip address just like using a
DDI?)
Phone system wise how would the NEC phone system determine routing of
calls via VOIP vs PSTN.

Sorry for all the questions, newbie to this. Want to get my head around
principles before going any further. Technology assumptions tend to
cost money :-)


Pure Networks
Posted by Wolfgang S. Rupprecht on February 24, 2006, 10:29 am
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smooter.guy@gmail.com writes:
> Does such a VOIP provider exist down here in NZ? Could I use overseas
> prov instead?

Poke around on this page. There are two business-to-business voip
companies listed under NZ. Probably there are a few multi-regional
ones too.

http://www.voip-info.org/tiki-index.php?page=VOIP+Service+Providers+Business

> How do callers make VOIP calls back to you (e.g a staff member being
> contactable at their desk phone via a Voip address just like using a
> DDI?)

If you want calls from the PSTN sent to you over VOIP, the easiest way
is to have a voip provider get you a set of DIDs. They will then
route calls made to those DIDS to your sip equipment.

Conventional wisdom is that you don't really want to do this in a
business setting. Voip is less reliable than direct PSTN connections
and you don't want to lose any incoming customer calls. The
conservative answer is use PRI trunking for incoming calls and use
voip for outgoing. If the outgoing voip is down, roll the outgoing
calls over to the PRI.

There are a few proposals that attempt to make it easier for companies
with sip-capable pbx's to call each other without going through the
PSTN, but that is still in its infancy. The one that, to me at least,
seems to have the feature set is called ISN. One dials a two-part
phone number with an embedded "*" as separator. One number identifies
the company, the other part is the internal phone number. The beauty
of this scheme is that there is no need to have a DID per phone for
this type of dialing. The intro page is at http://www.freenum.org/ .
It seems to work, I've used it, the problem is, very few organizations
have registered their sip servers so there just isn't much to call
yet.

> Phone system wise how would the NEC phone system determine routing of
> calls via VOIP vs PSTN.

Sorry, I'm only familiar with "homebrew" systems like asterisk. In
asterisk case one writes a ordered small list of what the system is
supposed to do when it gets a call. For outgoing calls one normally
has it try to route to the voip provider first, then a fallback voip
provider if one has one, then a direct PSTN line.

> Sorry for all the questions, newbie to this. Want to get my head around
> principles before going any further. Technology assumptions tend to
> cost money :-)

You might also find it interesting to poke around the top-level pages
of the following URL. It has some good background info, but the
indexing / searching could be better.

http://www.voip-info.org/wiki/

-wolfgang
--
Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html

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