[Asterisk-Users] Re: Direct SIP connection to Vonage service

[Stewart Nelson]

Hi,

Thanks for the replies.

Brian wrote:

> FYI these so called "unlimited" monthly plans are RARELY,
> if _EVER_ truly unlimited. They CAN (read the TOS), and
> WILL terminate you if you use too many minutes more then
> whatever average they calculated for when pricing the
> plan.

> I personally know several people who were using the Vonage
> "unlimited" calling plan and were terminated for
> _"EXCESSIVE USAGE"_

Ouch.  I am aware that service is not really unlimited.  My
present POTS service has "unlimited" long distance; the TOS
makes it clear that you are billed four cents per minute for
usage beyond 5000 min. per month.  That's pretty steep, but
going a little over won't break you, and it sure beats
having your service disconnected.  I usually run 2000-3000
min., and have never gone over 3500, so I'm not in any
danger.

Vonage, OTOH, is quite vague; their TOS speaks of
"inconsistent with normal residential usage patterns".  Do
you know what they consider "excessive", or if my usage
would be acceptable?

Benjk wrote:

> I personally wouldn't bother and I wouldn't want to take
> my money to a company that uses a business model that I
> despise. So, vote with your wallet. Don't use Vonage. Use
> a true VoIP service. And while we are at it, support IAX:
> Use a provider that offers IAX.

I looked at NuFone.net and some others, but it appears that
IAX is not right for my system.  I live near Reno, NV, and
have a second home in Paris.  Most of my calling is to the
US, via an H.323 gateway to the Reno POTS line; overflow
traffic is sent to an H.323 ITSP.  I run GnuGk on a shared
server at a hosting provider in New York.  Paris has a Cisco
827-4V (ADSL modem / NAT / 4 FXS) that speaks H.323 and SIP.
There are also some associates on the system using ATA-186.

When calling from an H.323 or SIP client to an IAX service
(or vice-versa), I believe that Asterisk must proxy the
media stream.  If * is run at the hosting service, I'm
worried that delays caused by other users will result in
choppy voice.  I'd rather run * in Reno, where it could also
replace an ancient DOS-based voice mail, and possibly my
Partner key system.  However, that configuration would have
lots of extra delay.  For example, if the IAX provider is in
Michigan, a call from Paris to San Francisco would go
Paris->Reno->Michigan->California.  With SIP, a REINVITE
would cause it to go Paris->Michigan->California, saving two
trips across the country.

Have I missed something?  Or did you mean that I should use
a provider that *offers* IAX, but connect via SIP :)

Thanks,

Stewart