[kwlug-disc] VoIP - Even without a server

[unsolicited at swiz.ca]

On Sun, 13 Feb 2011 09:10:37 -0800 (PST), Raul Suarez <rarsa at yahoo.com>
wrote:
> This one falls into the "I should have know this before!!!" category.
> 
> You don't need an Asterisk server to use a DID !! (d'oh)
> 
> I made my deposit and Ordered a DID with VoIP.ms. It all took 5 minutes
> online. 
> (I paid through PayPal)
> 
> I then explored the account management site and realized that I could
> connect my 
> ATA directly to the DID, create extensions, voicemail, call queues, Call

> forwarding rules and even create my own IVR.
> 
> 
> This is, I am able to do what I was trying to do by self hosting an
> Asterisk 
> server.
>  
> Actually if you have an smart phone with a SoftPhone app, you don't even
> need an 
> ATA. Just create the extension on the VoIP.ms server and configure your
> soft 
> phone application.
> 
> Of course this is the Linux User's Group so the bottom line is: You can
> get and 
> use the VoIP line and then start playing with your Asterisk server
without 
> compromising the stability of your phone system.
> 
> All for $0.99 cents per month + 1 cent per minute inbound, 0.5 cent per
> minute 
> out. (less than a single cup of coffee). And by the way, calling between

> extensions doesn't cost.
> 
> Something that I wasn't expecting, was total honesty. I read the Terms
of 
> service and they are very clear: "VoIP.ms does not pretend to offer 100%

> reliable service ... The customer shall not use this service as their
sole
> call 
> termination service".
> 
> So, if you are planning to make this your ONLY phone line, you must be
> aware of 
> the risk. If you have another line or a cellphone, then you should be
OK.

Interesting.

The sole phone line issue isn't one as most people have cells. In an
emergency, e.g. fire, it's likely you'll grab the cell and be calling 911
from there, not the home phone. (-:

Thank you for the post, didn't know voip.ms offered those services.
(Didn't look, either.)

Do please keep posting as to your experiences. e.g. ...

When I was thinking about such things a couple years back, one of the
things I encountered in my investigation (specifically with checking out
unlimitel / 'how things work'), was that only one 'phone' can register with
the provider at a time. There were a few scenario's I envisioned at the
time:
- each computer could have a softphone, and would or could therefore be an
extension. By functional extension, one might then want them as a (hunt?)
group. e.g. All computers ring at the same time, or if the computers don't
answer, fall over to the internal hard line, then to a cell, and so on and
so forth. Granted, this is all dialplan stuff, and all depends upon what
one wants to do, but I came out of that period with a solid sense that only
a home asterisk server was going to get me the flexibility I was looking
for. It sounds like from your investigation that that might not be true any
more. Do keep us apprised if you run into such things. Like, "I thought I
needed a home asterisk server for <that>, but don't", or, "I do still need
a home asterisk server for <this>."
- suppose I went out of town and have the laptop with a softphone. I would
want to be able to point it at my unlimitel account, and take home calls
while away. My understanding at the time was that I wouldn't be able to
register a second 'phone'. In essence I would have to redirect the
registering phone to the laptop softphone instead of the home asterisk box
- which was problematic (yanking control away / loss of functionality,
etc.)
- there are a number of free (non-DID) VoIP services out there, such as
Ekiga.net as you have found. From what you've seen with voip.ms, are you
able to pull them all together into a coherent whole / call path? e.g. I
imagine you might now forward your Ekiga.net account to your voip.ms
account. (Or vice versa?) [Additional charges, maybe, but at the price,
perhaps not a significant enough issue to worry about.]

Thanks for your post. Do please keep doing so.

- Glenn: With such voip.ms functionality as described, are we pretty close
to your seminar vision? This almost sounds like a WWITPRO / Working Centre
event, demonstrating voip.ms - perhaps with an ata or two around. Not to
say it's not appropriate for kwlug, but this scenario seems to move away
somewhat from a FOSS-only solution.