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SipStation - with connectivity provided by Bandwidth.com they’ve proven to be very reliable
and low cost.
8x8 - Based in Silicon Valley, they deal mostly in hosted PBX service. Reliability is very good.
Flowroute - Based in las Vegas offers very attractively priced DID numbers and special deals on
blocks of 20, 50 and even hundreds of numbers.
Local Number Portability in the USA and Canada
Telephone number porting is fairly common these days, mainly due to the large number of
wireless customers changing carriers and plans frequently and wanting to keep their phone
numbers.
Each carrier has its own rules and policies for number porting with the average port taking
from 10 days to 3 weeks. It is important to note that high volume or “choke” numbers cannot
be ported. You can try, but we haven’t seen it work yet. Usually, the order will be taken, a due
date given, and then on the day of the port, you’ll get a call from the utility or provider inform-
ing you of the bad news.
Porting is usually done in the morning, but special arrangements can be made with the carriers
to port at a specific time. Once ported, your old POTS lines are automatically canceled by the
old company. We have seen situations where CLECs and Wireless companies have some prob-
lems routing calls for the first few hours after a port, though this seems to be getting better.
Don’t run crazy “win a house or car” contests on these providers, though they’re fine for
request lines and even talk show call in numbers. If you plan to do heavy contesting with big
prizes, it’s not nice and potentially dangerous to bring your carriers switching office down.
Introduction to SIP’s insides for the Curious
The following is a peek inside of SIP. There is no reason you need to know any of this to use
the VX. It’s an excerpt from Steve and Skip’s book Audio over IP.
SIP is fast rising to be the big-daddy buzz-acronym among telecom technology acolytes. SIP is
how calls are set up over IP connections, so it is actually pretty important. Together with help-
ers like Proxy Servers and User Agents, SIP permits all the familiar telephone-like operations:
dialing a number, causing a phone to ring, hearing ringback tones or a busy signal. It also
enables next-generation capabilities such as finding people and directing calls to them wher-
ever their location, Instant Messaging, and relaying so-called ‘presence’ (near the phone or not,
do-not-disturb, etc.) information. SIP began, rather humbly, as a simple message protocol for
setting up connections. But the term has grown to be an umbrella for the family of protocols
and tools that have been developed by the IETF to enable VoIP telephony and related services.
By the mid-1990s, audio and video were becoming routine on the Internet. Going beyond
email, academic researchers were imagining on-line audio/video/whiteboard conferences
where ideas could be shared live. It became clear that the Mr. Watson come here, I want to
see you! function had to be done more efficiently than by shouting across the college quad or
sending invitation mails. Thus was the IETF’s working group MMUSIC (Multiparty Multimedia
Session Control) born. There had already been work within the telephone world that had
resulted in an ITU standard, but Internet types didn’t much like it. “Too complicated,” they
averred. “Too limited,” they sniffed. “Too Phone Company,” they huffed. So off they went to
do it the Internet way. The document describing SIP was eventually published as proposed
standard RFC 2543 in 1999. Work has been ongoing, with the latest version of the specification,
as this is written, being RFC 3261.
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