asterisk users - Aug 2004 - Help - is voip good for in-house calls?

Hi there everyone!

I work at an office where we plant to have about 12-15 phone
extensions. Costs of PBX are cheaper, but they are not expandable and,
as the office is brand new, I want to use all modern stuff.

My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and
install and asterisk server, as well as a Digium TDM400 for POTS
access, will I have the same voice quality and standards as a
PBX-only, with "traditional" phones? Or should I go all the way to
Digium's TDM? Or should I forget the whole thing and get a traditional
PBX? ;)

My concerns are most latencies. Our network will be a switch with lots
of ports, all 100mb/s, with VERY low traffic.

I've read lots about voip, and I'm quite impressed with it, but most
case studies show voip being used to interconnect offices. My case is
different - I want to replace a traditional PBX to handle in-house
phone calls, either from room-to-room in the same building and
room-to-POTS.

Any comment, help, tip or link would be greatly appreciated!

Yours truly,

Francis

Asterisk should work fine for this application - but you and/or your
users may be expecting the Grandstreams to look/act like traditional
key system phones, where you've got a bunch of buttons labeled
"Computer Room" or "Joe" and "Bob", or whatever,
where you can press
that button to call that extension.

The Budgetones don't do that - users will need to remember (or have an
extension list to tell them) that the Computer Room is at extension
110, and Joe is at 111, and Bob is 112, and so forth.

My suggestion is that you buy 2 Budgetones and set up Asterisk on an
old PC - so your total investment in the experiment will be < $200.
Get that up & running, and let users play with the phones and the
functions you can provide. If they like it, great. If they don't like
it, you're not out much money, and you ought to be able to resell the
Budgetones for something like 80% of the new price on Ebay or
whatever. You can get set up with incoming and outgoing IAX
connections via someone like Voicepulse or Nufone or IPKall (or some
combination thereof) so you can even let people experiment with
incoming and outgoing call quality and behavior without spending a lot
on interface cards.

You might also look at some of the other VoIP phones, which aren't a
whole lot more money and might look more like the PBX/key phones that
people are used to. The Budgetones are more similar to consumer/home
telephones from the early 1990's.

-- 
Greg Broiles, JD, EA
gbroiles@gmail.com (Lists only. Not for confidential communications.)
Law Office of Gregory A. Broiles
San Jose, CA

Hello Francis,

My office build is the same as yours.  15 or so extensions, low traffic
100MB network, and a desire to have a phone system that uses VoIP.  I
have my system working as a PBX just like you would.  I use two TDM400s
for my 8 POTS lines and Polycom IP 500 phones at the desktop.  I also
tested with the Grandstream phones you suggested.  SO, we have the same
system requirements so here are the answers as I have found them for my
implementation....

Voice quality on the SIP based phones has a lot to do with the codec you
use.  The lowest compression codec is uLaw and that is what I use since
we have tons of bandwidth to spare.  Also, my HP switch has COS (class
of service which is like QOS) so I can prioritize the packets coming
from my phones over the standard network traffic.  Even without this
switching feature turned on, performance was great.  The phones
themselves play another role in the quality.  Grandstreams are pretty
good and I have only used mine for testing so I will not disparage them.
However, the old saying stands.  You get what you pay for.  Raising your
phone budget from $85 to more like $150-250 will get you a phone with
more features and greater expandability in my IHO.  However, you can
still do great things with the cheaper Grandstream phones and still have
a system that works very well. IT is all up to what you can spend and
what you need.  Google the archive by putting "site:lists.digium.com"
in
front of your search string (no quotes though).  You should see some
good info on phones.  

Latency is gonna be there on any network.  However, on my network (which
is just like yours) the latency is very very low.  We are talking
20-40ms tops and it is completely unnoticeable when using the phone.
The only problem I have had at all has been with occasional echo.  It
does not happen often and it usually takes about 5 seconds for the * box
to train up and remove it.  Most of this seems to originate in the fact
that I am using POTS lines.  The solution that uses a T1 PRI has better
features and I think it has less echo potential.  However, that would
not work for me since my T1 provider wanted to make me pay 6 grand to
switch to a PRI from my standard data T1 with POTS.  Just some food for
thought...

I have been a VoIP user for about 1 month after spending another
researching what when where how...  So, we know I am not an expert...
but as a fellow user and new VoIP initiate, I can tell you that Asterisk
is a phenomenal product for SMB level offices like yours and mine.  When
I compared it to a PBX system of comparable power, expandability, and
feature set, Asterisk won easily since the only real cost I have had was
for my phones.  I have my system in place for around 3000 dollars and it
is competitive with all the 10K dollar solutions the vendors threw at me
plus it has an undeniable advantage in upgrade path.  All upgrades to
the system are free and the sky is the limit to what you can build using
the framework that all the * gurus have built into this system.  Not to
mention the fact that if anything ever goes wrong with the server, I can
have a new one in place in under and hour.  Try that with a PBX when
some proprietary part goes belly up.  You could wait days potentially.
My $.02.  Hope this helps.

Cheers,
Wiley




-----Original Message-----
From: Francis Augusto Medeiros [mailto:francismedeiros@gmail.com] 
Sent: Saturday, August 14, 2004 1:08 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Help - is voip good for in-house calls?

Hi there everyone!

I work at an office where we plant to have about 12-15 phone
extensions. Costs of PBX are cheaper, but they are not expandable and,
as the office is brand new, I want to use all modern stuff.

My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and
install and asterisk server, as well as a Digium TDM400 for POTS
access, will I have the same voice quality and standards as a
PBX-only, with "traditional" phones? Or should I go all the way to
Digium's TDM? Or should I forget the whole thing and get a traditional
PBX? ;)

My concerns are most latencies. Our network will be a switch with lots
of ports, all 100mb/s, with VERY low traffic.

I've read lots about voip, and I'm quite impressed with it, but most
case studies show voip being used to interconnect offices. My case is
different - I want to replace a traditional PBX to handle in-house
phone calls, either from room-to-room in the same building and
room-to-POTS.

Any comment, help, tip or link would be greatly appreciated!

Yours truly,

Francis
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Hello Francis,
> I'll most likely use a BRI. Do you think this will help to avoid echo?
I could not say as I have never used a BRI and I am pretty new to this
too.  I do know that BRI is supported from watching conversations in
this email list and reading online.  People seem to use it a bit so it
must work well.  Googling the list with BRI should get you tons of good
leads.

Greg had a great idea in having you set it up and try it.  In fact, that
is exactly how I did mine.  I purchase a cheap clone card for $15 and
used it to test on one POTS line while I tweaked my configuration files
and got the system validated.  I tested the system with soft phones, one
Polycom IP 500, and one Grandstream Budgetone 101.  The Budgetone worked
well and was leagues easier to setup than my Polycom actually.  

For expandability, I believe that the cap I have seen is about 60
concurrent calls for one Asterisk box and that is with a pretty serious
server by most users standards.  I cannot imagine having that many calls
at this point so I am fine but I jus though t you would want to know.
The nice thing about * is that you can just build another server and
link them together over IAX. Again, the low cost of implementation pays
off and you get to continue growth.  I will never go back to proprietary
PBX now that I finally have a solution that I can control.

Cheers,
Wiley

On Sat, 14 Aug 2004, Wiley E. Siler wrote:
> Greg had a great idea in having you set it up and try it.  In fact, that
> is exactly how I did mine.  I purchase a cheap clone card for $15 and
> used it to test on one POTS line while I tweaked my configuration files
> and got the system validated.  I tested the system with soft phones, one
> Polycom IP 500, and one Grandstream Budgetone 101.  The Budgetone worked
> well and was leagues easier to setup than my Polycom actually.  
Using a pots may not give an accurate picture. It is a source of echos 
which can, when combined with a slight latency introduced in the voip 
links, change an acceptable reverb to a nasty echo.

The cheap clone cards are all of the x100 card I believe. It has a fixed 
impedance of 600 ohms pure resistive. A lot of countries outside USA seem 
to use other line impedances. The mismatch leads to echos.
> For expandability, I believe that the cap I have seen is about 60
> concurrent calls for one Asterisk box and that is with a pretty serious
> server by most users standards.  I cannot imagine having that many calls
> at this point so I am fine but I jus though t you would want to know.
> The nice thing about * is that you can just build another server and
> link them together over IAX. Again, the low cost of implementation pays
> off and you get to continue growth.  I will never go back to proprietary
> PBX now that I finally have a solution that I can control.
With no transcoding you should be able to go higher than that I expect. 
For local ip phones g.711 is probably usable. Most low numbers I have 
seen seemed to do a lot of transcoding to liwer bitrate formats.

Peter

Hi Francis,

Francis Augusto Medeiros wrote:
> Hi there everyone!
> 
> I work at an office where we plant to have about 12-15 phone
> extensions. Costs of PBX are cheaper, but they are not expandable and,
> as the office is brand new, I want to use all modern stuff.
> 
> My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and
> install and asterisk server, as well as a Digium TDM400 for POTS
> access, will I have the same voice quality and standards as a
> PBX-only, with "traditional" phones? Or should I go all the way
to
> Digium's TDM? Or should I forget the whole thing and get a traditional
> PBX? ;)
If you already have the analog telephone wiring in place, and you are on 
a budget, I recomend you to use sipura spa-2000 adapters. They are a 
whole lot better than GS phones. You can have 3way conferences and 
attendant transfers. With GS you cannot do that. The price is as good 
for the sipuras as the GS phones, about $50 per FXS port, plus a cheap 
analog phone and you will be all set.
> 
> My concerns are most latencies. Our network will be a switch with lots
> of ports, all 100mb/s, with VERY low traffic.
> 
Internal calls (SIP to SIP) will sound great. You will probably 
experience some echo when going to POTS. I did not try the Sipura 
SPA-3000 yet, but it seems to be a cheap alternative to a gateway, 
providing you with one FXO and one FXS for $130 or so. the echo 
cancellation in the sipura works well for fxs, it might work well to for 
fxo.


-- 
Nicolas Gudino
House Internet S.R.L.
Buenos Aires - Argentina

On Sunday 15 August 2004 12:03, Nicolas Gudino wrote:
> If you already have the analog telephone wiring in place, and you are on
> a budget, I recomend you to use sipura spa-2000 adapters. They are a
> whole lot better than GS phones. You can have 3way conferences and
> attendant transfers. With GS you cannot do that. The price is as good
> for the sipuras as the GS phones, about $50 per FXS port, plus a cheap
> analog phone and you will be all set.
Why on earth would you install SPA-2000s and endure that wiring mess?  An FXS 
channel bank and a BIX strip will save you YEARS in lost time due to wiring 
and general messiness!

-A.

Thanks a lot Daryl!!

Yours,

Francis

On Wed, 18 Aug 2004 10:25:21 +0930, Darryl Ross <darryl@oeg.com.au>
wrote:
> Hi Francis,
> 
> >>>so no need to make a special dialplan to
> >>>acomodate the weird numbering system we have in Brazil
(sometimes we
> >>>dial 7 numbers, sometimes 8, sometimes 12, sometimes 13, etc.)
> >>
> >>Actually, we also have non-fixed phone numbers in Germany. I think
this is
> >>not weird, I think this is very good. And again, Asterisk supports
this.
> >
> > Oh, so I how does Asterisk knows when to start dialing out the
> > numbers, if there are no rules?
> 
> Have a look at http://www.voip-info.org/wiki-Asterisk+Extension+Matching
> 
> Regards
> Darryl
> 
> --
> Darryl Ross
> Senior Network Engineer
> OEG Australia
> Email: darryl@oeg.com.au
> Phone: 08 81228363
> Office: 08 81226361
> 
> If you want to live up to the whole "There is more than one way to
>      do it" slogan, you have to give someone a swiss army chainsaw ...
> 
> 
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