Other than an additional line - what benefits does VOIP bring?

I picked up a cheap 2nd user Adsl2+ modem router with VOIP at my local
bootsale - all of 5 quid! - and configured it up to use a free account at
sipgate.co.uk

It works and I now have in effect a second telephone line which is free to
receive incoming calls and to ring other sipgate users. As I know no other
sipgate users this line is in effect incoming calls only.

Now the prices of outgoing calls on this "line" to UK numbers are nowhere as
cheap as 18185, and the cost of international calls cannot beat services
such as dialaround and dialwise. The same applies to other UK voip services.

I like the idea of having a telephone number in the US of A, but this seems
a costly option now that sipgate.com is out of numbers. Unfortunately IPKall
(suggested in another thread) does not work with sipgate.co.uk

Now I'd like to be able to do fancy stuff like block incoming international
calls (the bane of my life), annonymous call reject and routing certain
calls to voicemail based on CLI. This does not seem to be possible through
the router or the sipgate portal.

I was under the impression that the stuff above was possible with VOIP. Can
anyone point me in the right direction? Do I need to run some type of VOIP
software on a computer on my home network to achieve this?

Brgds,

 

The above is all possible...

Sipgate is a service you connect a phone to.

Ipkall is a service you connect a phone to.

You don't make one work with with the other - it's not designed to work
like that - you can have 2 phones, or 2 accounts on one phone, or some
sort of device that pretends to be N phones and allows you to plumb them
together and do clever stuff...

That device is a VoIP capable PBX, or software running on a PC such as
Asterisk in the Linux world, or 3CX in the MS world.

Then you can program it to accept calls on either service, place outgoung
calls via either service (or via another service) and so on. Limited
only by your imagination and progrmaming ability. (Or, you pay someone
to do it all for you - which most businesses are happy to do)

As for the point of VoIP... Different people interpret it in different
ways and I've never sold it based on cost-savings (however, generally
in the business world it is cheaper than standard BT, etc. lines)
- flexibility is what I sell it on. The ability to connect offices
together, work from home, work away from home and so on, and in some
offices, it's sometimes easier to use an existing cat5 network rather
than run separate voice cabling.

Gordon

 

But you can dial out on it, which is useful. OK, more expensive (for some
things anyway) but it means you have a fallback if someone else is on the
main line.

Agreed, most of the time. Of course, BT is cheaper (i.e. free) for many
calls at the weekends, even on the most basic normal tariff.

It's VoIp. It works if you have a way to integrate it.

In a word...Asterisk. But although it's documented, it's a moving target.
If you get it on a suitable box (most use Linux but I use FreeBSD, not
wishing to use a jumped up wannabe UNIX!) then it can integrate those,
plus (optionally) your normal line, plus other stuff. Not trivial, but
easier if you get the O'Reilly book (but download the errata, of which I
have contributed a lot).

You can route calls the cheapest way (depending on day/time) as well as
having a fallback if a line is busy. All that screening stuff, etc. is
also easy.

Whether that's all worth it is up to you.

 

That's not completely correct and OP Peter does indeed have a valid point.

Unfortunately these various "services" don't seem to have standardised names
but I'll try to explain my understanding. If anyone can contribute standard
names, please do so.

IPKall is a purely a DID company. It issues phone numbers and can receive
incoming calls from the PSTN and redirect them over IP to a proxy server in
the form of [email protected] The call recipient's own connection, using an ATA
or softphone or whatever, can connect to that proxy server and then terminate
the incoming calls in a wired handset or PC or mobile phone etc.

A company such as callcentric.com and callwithus.com offers such proxy
services and I have uses them as server for calls redirected from IPKall.

Sipgate is a company that offers both part of that sequence, they issue DID
phone numbers and redirect them to [email protected] and they also
operate the domain server for the proxy service. However, as Peter points
out, Sipgate configures their server to reject calls from services they have
not registered for peering, and for now calls are not accepted from IPKall.

 

Sorry bad edit should be [email protected]@sipgate.co.uk.

 

This probably doesn't address your question, but one thing you
can to with VOIP is use an iPod, iPhone, or Android device as a
remote extension for the purpose of placing calls.

I've got an account with CallCentric that I use for my home phone
system via a LinkSys SPA3102 gateway.

But I also have an app called "Bria" on my iPod that connects to
that same CallCentric account no matter where I am (as long as
I've got a WiFi connection... I'm too cheap to have a data plan)
and lets me use the iPod as an outgoing phone.

Yeah, one could do the same thing with Skype and "Skype
Minutes"... but this sounds cooler.... -)

 

<snipped>

Thanks for all the extremely helpful replies so far. I've visited the
Asterisk website and read some of the beginners guide this morning, and it
seems quite interesting indeed.

It looks like Asterisk Now may be the best option for me given that I am a
newbie to the software and to VOIP//telecomms in general. I did dabble
fairly heavily with Linux several years ago - to kernel compilation level,
but it has been a while.

I'll download the software and see whether it will work on a tiny Emachine
"Mini-e" ER1401 that I have. It is currently running Windows 7 Ultimate
64-bit quite happily - I will be interested to see how it takes to Linux

Following on from Bob's suggestion I'll have a look at the O'Reilly Asterisk
book online as I have access to Safari IT books.

Let's see how I get on

 

I would not recommend any Asterisk GUI to a newbie. You should start
with bare Asterisk and treat the GUI as a productivity aid, rather than
a substitute for learning. The GUIs are naturally limited and also
their users tend not to have deep understandings, so when you try and
get support, you will find you have to start talking in terms of what is
really happening, as the GUI users will be unable to help.

 

That's mindbogglingly fast for Asterisk. I'm running it on a 200MHz
wireless router with 32MB RAM, where it works OK as long as you don't try to
transcode audio. Since that machine is also Ubuntu certified [1], whatever
flavour of Linux used in the Asterisk GUI should be fine too.

[1] http://www.ubuntu.com/certification/hardware/201006-5930

Theo

 

I'm running on a 400MHz Geode machine - it pulled 10 watts measured at the
mains, and that was before I replaced its hard disc with an SDHC card.

This extra bit of speed is good, it allows for audio transcoding, as I use
PCMA for my devices and trunks (one only allows PCMU...) but I have Sipdroid
configured to use GSM. I've found that works a bit better when bandwidth
can be limited.

 

Difficult to measure accurately at that level - but IME most hard disks
draw anything up to 10W, so it might be a bit more.

My Asterisk box has mirrored hard disks, and it's a 1GHz VIA board -
still only about 35W. At that speed, transcoding isn't aproblem - not
unusual to have two or more calls going at once!

 

I build systems out of 500MHz Geode boards - for Asterisk and routers
and other (semi) embedded projects... I usually use a 256MB industrial
CF card:

http://unicorn.drogon.net/cutie.jpg
http://unicorn.drogon.net/cutie2+power.jpg

Because I sell such systems, I stress test them to work out their exact
limits, so I have some good ideas... However I also have a highly tuned
Linux kernel and userland geared up to the exact hardware platform,
which does help.

On the 1GHz VIA platforms, I've had well over 100 concurrent calls
handling media with no transcoding. I limit these boxes to 120 extensions
(so a max. of 60 calls) the Geodes topped out at 80 concurrent calls,
handling media (no transcoding) (These I limit to 30 extensions)

There is a non-linear relationship to clock speed and concurrent
calls managable. Probably due to kernel switching & hardware handling
times. (and cache, etc. too) On some (very!) old 533MHz VIA systems
which I use for development, they struggle to match the 500MHz Geodes.

I wish I could turn off the graphics on the VIA boards - I'm sure that
accounts for 10 of the 15 watts my VIA systems consume )-:

http://unicorn.drogon.net/stuff/power.jpg

There's a 1GHz VIA inside that box.

The 1GHz VIA boards will transcode 8-9 instances of G711 to G729 and
about 15 to GSM.

Even very fast systems eventually tail off - the limitation then is
the kernel & hardwares ability to handle the packet switching load -
VoIP is small (160 byte) packets at 50 packets per second - each way,
so the kernel is handling 100 packets per second per call - that, more
than the physical line speed is more a limiting factor than anything else
(except transcoding) on slower boards, and even on faster boards it's
the packet rate that will cripple the system before you get anywhere
near physical line rate.

e.g. The Geodes fail non-linearly once past 80 calls handling media -
at 82 calls the CPU load was moderate - under 10% - add one more call
and the load doubles, another call and it doubles again and at 85 calls
packets are being lost and the CPU is pegged at 100%

VIA boards are becoming harder to get these days, so I'm moving to Atom
boards. Cheaper, faster and with the right chipset, no more power either.

Gordon

 

I understand that....and you mentioned the CF card before. I was just
wondering about the power level before you took out the real HD, that's
all....it implied that the power consumption after removing it was only
about 5W. I can't remember what Geode boards use...I looked at them a
while ago but had trouble sourcing them so went with VIA.

I use a CF card in one of my VIA machines, too...I have built quite a few
of them.

I've done the same, but with FreeBSD. As a BSD user since 1977, it was a
more logical choice.

I'm sure it does. I was being pessimistic and allowing for transcoding;
also, I built 'standard' VIA systems; they are used for many other
purposes, one of those being Asterisk.

That's useful to know; a bit more than I expected.

Probably what I'll do. At present I just have lots of reasonably low
power systems, all in 1U boxes.

 

I've no idea - as I started from scaratch with CF cards in the Geodes
(and Disk on Flash IDE modules for the VIAs and Atoms)

77. You're boasting now Sure it was BSD though? Thought that started
in '78... My first experience with Unix was v6 in a PDP11/40 - and that
was in 1979/80, closely followed by v7... Some say it all went downhill
after that...

Anyway, BSD's not relevant today ;-)

http://bsd.slashdot.org/story/11/07/16/0020243/Lennart-Poettering-BSD-Isnt-Relevant-Anymore

The C7's are supposed to be more efficient than the C3's, but I've not
really noticed any difference. Maybe for some of the graphics stuff
they are though.

For my home/office, I've moved to virtualisation just to cut down the
box count, however as it's relatively carefully controlled, it works well
for me. (4 or 5 PBXs - one my office, one home, one my wifes office and
one or 2 under test) Mostly all hosted on an Atom server.

Biggest hassle I've had with 1U boxes was getting them with the right IO
backplanes )-:

Cheers,

Gordon