asterisk users - Aug 2005 - Small office setup/using analog lines w/ Asterisk

Hi,

We recently tried installing Asterisk for a small office. We figured the 
safest way to go would be to buy from someone who sold equipment 
specifically for Asterisk and to use a consultant that they 
recommended.  However ... it didn't turn out so great.  Sound quality is 
terrible -- the echo is pretty bad, and there are popping noises, too. 
Callers say that people on the Asterisk end sound very faint, while 
people on the Asterisk end hear people maybe too loundly (might be 
related to the popping noises -- sounds like when you have stereo turned 
up too high).  The reseller and the consultant both say that the most 
likely cause for this is using Digium cards w/ analog phone lines. 
Apparently, they say, sound quality can be pretty bad.

I called Digium and they gave me some suggestions for settings, but 
nothing has worked well. So I wanted to ask others ... has anyone had 
good luck with using analog phone lines and Asterisk? Especially with 
Digium cards (we use the TDM400P)? Although from reading articles on the 
net it sounds like people do have a lot of echo problems, it also sounds 
like some people are using analog phone lines with some success.

FYI, what I've mainly done is try changing echotraining, echocancel, 
echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard 
from the reseller that what might work better is to trade the Digium 
cards in for VegaStream gateway. It's more expensive, but apparently has 
a DSP built in that should increase voice quality. Of course, they say 
there are no guarantees with this.  They also mentioned (after the fact) 
that Asterisk systems don't necessarily save money. So far, the 
experience has been very frustrating and I'd love to hear some success 
stories from others (or more info on what I can realistically expect 
from an Asterisk system)! And, of course, some ideas on how I can get 
things to work better.

One of the next tests will be using Asterisk with a VoIP provider to see 
what the sound quality is like with digital on both ends. PRI sounds 
like it'd be even better, but for an office w/ 5 people, it sounds 
pretty expensive. How do other people do this?

Thanks in advance for any pointers!

Jen

Jenny -

I'm glad I'm not the only one! I just installed Asterisk on Friday and I
spent all day trying to de-gremlin my system. I'm glad I'm doing it for
myself and we haven't switched from our legacy system yet, but I have a
potential client that wants to see how well I can implement this system
for myself...

(Speaking of which, don't forget that your supplier and consultant
probably make more off of other phone systems and a lot of old-line
suppliers are kind of afraid of Asterisk. There were some big companies
that lost out when the State of Oregon's court system went to Asterisk for
their teleconferencing needs... 600 MeetMe sessions a day does say
something, I think.)

First, it would help -me- to know precisely what hardware you're using
(including network... switches and cards) and what phones you're using.
You can email me off-list if you want.

I'm personally using a AMD Sempron-based system with a TDM-04B (4x FXO) in
it, routing out through an IntelPro 10/100 card. The data switch is
currently a crappy D-Link, but more on that and the network card in a
second. The other end is one soft phone on Linux and two SPA-841 hardware
phones.

When I first installed, the echo was terrible. Tweaking around with the
echo settings in zapata.conf endlessly didn't do a whole lot. So, using my
go-go-gadget troubleshooting instincts (and my handy-dandy go-go-gadget
credit card) I figured the problem might be network latency ... so I
replaced the crappy realtek network card with a high-quality Intel one,
and borrwed a Dell managed switch from a client for the day. Voila, a few
more tweaks to the echo learning rate and my echo problems went away, and
I spent 45 minutes learning about my sister's love life in the wilds of
downtown San Francisco. (Note to self: Don't use family to test telecom
systems.) The clicking persisted, though. I thought it was the phone at
first, but if you're experiencing it too... well, apparently it's not.

I searched the archives and couldn't find any other reference to clicking
noises, so I'm at a loss but I'm hoping it's wiring-based and
heading back
to the office tomorrow to try and solve it. I think you're right that
Digium is focusing on their T-1/E-1 market. I've noticed it when I called
with a question, and that really ticks me off. It might be a decent, sound
business decision based on profits for the different cards, but it's
leaving a lot of us smaller business operators -- those who might take
great experiences with Digium and Asterisk products into potential large
customers -- swinging in the wind.

-Karl Katzke
Streetlamp Software Solutions
http://www.streetlampsoftware.com

> Hi,
>
> We recently tried installing Asterisk for a small office. We figured the
> safest way to go would be to buy from someone who sold equipment
> specifically for Asterisk and to use a consultant that they
> recommended.  However ... it didn't turn out so great.  Sound quality
is
> terrible -- the echo is pretty bad, and there are popping noises, too.
> Callers say that people on the Asterisk end sound very faint, while
> people on the Asterisk end hear people maybe too loundly (might be
> related to the popping noises -- sounds like when you have stereo turned
> up too high).  The reseller and the consultant both say that the most
> likely cause for this is using Digium cards w/ analog phone lines.
> Apparently, they say, sound quality can be pretty bad.
>
> I called Digium and they gave me some suggestions for settings, but
> nothing has worked well. So I wanted to ask others ... has anyone had
> good luck with using analog phone lines and Asterisk? Especially with
> Digium cards (we use the TDM400P)? Although from reading articles on the
> net it sounds like people do have a lot of echo problems, it also sounds
> like some people are using analog phone lines with some success.
>
> FYI, what I've mainly done is try changing echotraining, echocancel,
> echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard
> from the reseller that what might work better is to trade the Digium
> cards in for VegaStream gateway. It's more expensive, but apparently
has
> a DSP built in that should increase voice quality. Of course, they say
> there are no guarantees with this.  They also mentioned (after the fact)
> that Asterisk systems don't necessarily save money. So far, the
> experience has been very frustrating and I'd love to hear some success
> stories from others (or more info on what I can realistically expect
> from an Asterisk system)! And, of course, some ideas on how I can get
> things to work better.
>
> One of the next tests will be using Asterisk with a VoIP provider to see
> what the sound quality is like with digital on both ends. PRI sounds
> like it'd be even better, but for an office w/ 5 people, it sounds
> pretty expensive. How do other people do this?
>
> Thanks in advance for any pointers!
>
> Jen
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

jennyw wrote:
> Hi,
> 
> We recently tried installing Asterisk for a small office. We figured the 
> safest way to go would be to buy from someone who sold equipment 
> specifically for Asterisk and to use a consultant that they 
> recommended.  However ... it didn't turn out so great.  Sound quality
is
> terrible -- the echo is pretty bad, and there are popping noises, too. 
> Callers say that people on the Asterisk end sound very faint, while 
> people on the Asterisk end hear people maybe too loundly (might be 
> related to the popping noises -- sounds like when you have stereo turned 
> up too high).  The reseller and the consultant both say that the most 
> likely cause for this is using Digium cards w/ analog phone lines. 
> Apparently, they say, sound quality can be pretty bad.
> 
> I called Digium and they gave me some suggestions for settings, but 
> nothing has worked well. So I wanted to ask others ... has anyone had 
> good luck with using analog phone lines and Asterisk? Especially with 
> Digium cards (we use the TDM400P)? Although from reading articles on the 
> net it sounds like people do have a lot of echo problems, it also sounds 
> like some people are using analog phone lines with some success.
> 
> FYI, what I've mainly done is try changing echotraining, echocancel, 
> echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard 
> from the reseller that what might work better is to trade the Digium 
> cards in for VegaStream gateway. It's more expensive, but apparently
has
> a DSP built in that should increase voice quality. Of course, they say 
> there are no guarantees with this.  They also mentioned (after the fact) 
> that Asterisk systems don't necessarily save money. So far, the 
> experience has been very frustrating and I'd love to hear some success 
> stories from others (or more info on what I can realistically expect 
> from an Asterisk system)! And, of course, some ideas on how I can get 
> things to work better.
> 
> One of the next tests will be using Asterisk with a VoIP provider to see 
> what the sound quality is like with digital on both ends. PRI sounds 
> like it'd be even better, but for an office w/ 5 people, it sounds 
> pretty expensive. How do other people do this?
I started using Asterisk for my own small business about a year ago.

Externally we have a single analogue PSTN line (it's the house one...), 
an ISDN2e connection and an IAX2 connection (over 20:1 256/512kbps ADSL) 
with a DID in central London. The analogue line comes in to an old 
X100P, and the ISDN into an AVM Fritz! passive card.

Internally, we have a TDM400 which talks to analogue phones in the 
house. In my office (which is in a different building) we have a mixture 
of Snom and ipDialog phones and a Grandstream ATA attached to a fax machine.

We get a little echo on the ipDialog phone (but not enough to be a 
problem) when we talk to people on analogue phones. One of the handsets 
  attached to the TDM400 is a DECT phone, and there's a little flurry of 
training noise at the beginning of an incoming call, but after that the 
quality is good to perfect.

I'm just beginning to sell Asterisk systems. I agree that for some 
installations, it doesn't really make economic sense. In the UK, at 
least, you have to fall into a specific band of numbers-of-users and 
minutes-per-month for IP telephony to show a saving. Some of the small 
3-line-8-extension systems from (say) Panasonic will be cheaper than 
Asterisk once the hardware is bought and the time (or consultancy) 
applied. Of course, these systems don't have much in the way of 
flexibility or features, and I'm talking at the moment to a company that 
has three sites, is using Cisco's Call Manager, and has an Asterisk 
system merely to convert the H.323 from the Cisco to IAX2.  In this 
case, * could replace the CCM system in its entirety.

By the time you have 100 users, * is a no-brainer in economic terms. 
Small users only really save (IMHO) if they a) use an awful lot of 
minutes (or call abroad a lot), b) need flexibility of features, or c) 
need internal control.

Of course there may be local or exceptional circumstances which make 
this all a load of rubbish ! YMMV.

Oh, and on echo; read :

http://lists.digium.com/pipermail/asterisk-users/2005-March/096754.html

jd

-- 

John Daragon                                          john@argv.co.uk
argv[0] limited
Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127

Jennyw,

I have setup about 8 Asterisk systems with The TDM400p boards in them. Yes 
allot of them had at the beginning some echo and other things. But I have 
been able to work and get them fixed.

1) Make sure your motherboard is able to assign it's own IRQ for the board. 
This is one of the most important things.
2) There is a tool on the system that lets you set the txgain and rxgain. 
It's called ztmonitor which you can use to see how the volume is setup then 
you can make some adjustments.
3) It's important to make sure your system has good network card. I have had
some problems with Realtech with echo.
4) You also need to make sure you have the phones on connect to a switch.
5) Asterisk system does save allot of money in the long run.  I am sorry 
that the person you got was not able to help. But I know that the TDM400p 
boards have there well critic's but they do work.
6) For us to give you more help we are going to need to know more about you 
system. What is the server your using? What phones? How is your network 
setup?  If you want you can email me directly. I will try to help you out 
with your setup.

Ariel Batista

jennyw wrote:
> Hi,
>
> We recently tried installing Asterisk for a small office. We figured
> the safest way to go would be to buy from someone who sold equipment
> specifically for Asterisk and to use a consultant that they
> recommended.  However ... it didn't turn out so great.  Sound quality
> is terrible -- the echo is pretty bad, and there are popping noises,
> too. Callers say that people on the Asterisk end sound very faint,
> while people on the Asterisk end hear people maybe too loundly (might
> be related to the popping noises -- sounds like when you have stereo
> turned up too high).  The reseller and the consultant both say that
> the most likely cause for this is using Digium cards w/ analog phone
> lines. Apparently, they say, sound quality can be pretty bad.
>
> I called Digium and they gave me some suggestions for settings, but
> nothing has worked well. So I wanted to ask others ... has anyone had
> good luck with using analog phone lines and Asterisk? Especially with
> Digium cards (we use the TDM400P)? Although from reading articles on
> the net it sounds like people do have a lot of echo problems, it also
> sounds like some people are using analog phone lines with some
> success.
> FYI, what I've mainly done is try changing echotraining, echocancel,
> echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard
> from the reseller that what might work better is to trade the Digium
> cards in for VegaStream gateway. It's more expensive, but apparently
> has a DSP built in that should increase voice quality. Of course,
> they say there are no guarantees with this.  They also mentioned
> (after the fact) that Asterisk systems don't necessarily save money.
> So far, the experience has been very frustrating and I'd love to hear
> some success stories from others (or more info on what I can
> realistically expect from an Asterisk system)! And, of course, some
> ideas on how I can get things to work better.
>
> One of the next tests will be using Asterisk with a VoIP provider to
> see what the sound quality is like with digital on both ends. PRI
> sounds like it'd be even better, but for an office w/ 5 people, it
> sounds pretty expensive. How do other people do this?
>
> Thanks in advance for any pointers!
>
> Jen
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

On Sunday 21 August 2005 01:05, jennyw wrote:
> up too high).  The reseller and the consultant both say that the most
> likely cause for this is using Digium cards w/ analog phone lines.
> Apparently, they say, sound quality can be pretty bad.
The reseller/consultant aren't worth the money you paid them, then.

Interfacing to plain-old analog lines can be problematic.  The TDM400P FXO 
modules are tunable but it takes time and testing to get it right, not 
playing about willy-nilly with settings in an attempt to solve the problem 
through entropic little adjustments without a clear idea of what they do and 
how they work.

Asterisk is a very difficult application on a system.  Interfacing to anything 
outside the computer, whether it be an analog telephone line, a local SIP 
phone or a remote VOIP provider requires that the system's ability to access
its resources reliably and with repeatable access times.  As simple as this 
sounds it is a very difficult problem and the #1 reason why VOIP is so 
difficult to roll out on commodity hardware.  You simply can't use any old 
system and any old network card and any old network gear (router/firewall, 
switches, etc.) and get good results.
> nothing has worked well. So I wanted to ask others ... has anyone had
> good luck with using analog phone lines and Asterisk? Especially with
> Digium cards (we use the TDM400P)? Although from reading articles on the
> net it sounds like people do have a lot of echo problems, it also sounds
> like some people are using analog phone lines with some success.
The echo problems are almost always due to one of two things: poor line tuning 
or crappy base hardware (computer).

Now the older version of the TDM cards and FXO modules specifically had 
issues, but they have, to my knowledge, all been resolved.  I used to 
recommend a T1 card + channel bank (Adit600) even for a couple channels, but 
nowadays I have no compunctions in recommending the TDM400 and FXO modules.
> echocancelwhenbridged, txgain, and rxgain in zapata.conf. I've heard
> from the reseller that what might work better is to trade the Digium
> cards in for VegaStream gateway. It's more expensive, but apparently
has
> a DSP built in that should increase voice quality. Of course, they say
> there are no guarantees with this.  They also mentioned (after the fact)
> that Asterisk systems don't necessarily save money. So far, the
Find a new reseller, and post their name here so we can all avoid them.

I've rolled out numerous asterisk installations with good success.  As I 
mentioned earlier, the trick is measured, controlled tests and methodical  
experimentation.  

http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html

is the method I use to tune the line, but there is more to it now since there 
is the fxotune application which is used to set up the FIR filter on the FXO 
card's DAA.

If you're using a SIP phone to test with, put a second network card in the 
Asterisk box and plug the phone directly into it (with a crossover cable) and 
adjust the Asterisk settings to see and use the second card.  This will 
determine if your network is causing issues by eliminating it from the 
equation.  Make sure you use a decent network card (I love the Intel 
eepro/100 cards myself).

Also, if you're going out through a VOIP provider, make damn sure you either
use a dedicated DSL link (my personal recommendation) or make really sure 
that your router can properly tag and prioritize outgoing traffic, and that 
it is also doing its best to prevent the "other side" of the link from
flooding your incoming pipe.  http://www.mixdown.ca/~andrew/dump/rc.tc is the 
script I use with good success.

As far as Asterisk being more expensive than other systems...  doubtful.  You 
can get a cheap Nortel 3x8 for cheap, sure, but then its limitations will 
have you buying a small MICS... Now add their $4000 voicemail system, $500 
trunk cards for four FXO channels... oh wait, you want caller-id on those?  
$600 then...  oh wait, you want VOIP on it?  $2500 here, $500 there, $1000 
the other place...  

Make sure you're comparing apples to apples.  I feel that Asterisk runs
*very*
well on most hardware I've thrown it at, and it is far far far more 
configurable than any proprietary KSU or PBX, and a damn sight cheaper than 
*ANY* PBX out there.
> One of the next tests will be using Asterisk with a VoIP provider to see
> what the sound quality is like with digital on both ends. PRI sounds
> like it'd be even better, but for an office w/ 5 people, it sounds
> pretty expensive. How do other people do this?
Yes, an office with 5 people (probably only two POTS lines I am guessing) is 
not really a good choice for PRI.  ISDN BRI if you can get it would be 
better, or simply finding a trusted VOIP provider and getting a DID from them 
would be easiest, but I would then recommend two DSL providers on the same 
POTS line (it's all PPPoE anyway) for failover.

-A.

I too have had trouble with FXO interfaces. I tried the Sipura SPA-3000
FXS/FXO device , X101p cards and a TDM11B card. None were satisfactory
for my small office with 6 extensions and 3 lines.

My longer term workaround was something that I setup just to bridge a
perioud when I was taking down a test server and putting a poduction
server online. I put provider based call forwarding on my POTS lines
(only 2) and forwarded them to my IP based 800 number. This turned out
so reliable that I have remained operating this way ever since, about 4
months now. People have remarked that the call quality is superb.

I ever there is a problem with my DSL line or * server I simply turn
off the call forwarding using a pair of analogue phones that I leave on
the POTS lines for just such emergencies.

This only addreses incomming calls. All outgoing calls are handled via
IP through another termination provider.

Michael

On Sat, 20 Aug 2005 22:05:31 -0700, jennyw wrote:
>Hi,
>
>We recently tried installing Asterisk for a small office. We figured the 
>safest way to go would be to buy from someone who sold equipment 
>specifically for Asterisk and to use a consultant that they 
>recommended.  However ... it didn't turn out so great.  Sound quality is
>terrible -- the echo is pretty bad, and there are popping noises, too. 
>Callers say that people on the Asterisk end sound very faint, while 
<,snip>
>One of the next tests will be using Asterisk with a VoIP provider to see 
>what the sound quality is like with digital on both ends. PRI sounds 
>like it'd be even better, but for an office w/ 5 people, it sounds 
>pretty expensive. How do other people do this?
>
>Thanks in advance for any pointers!
>
>Jen
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
--
Michael Graves                           mgraves@pixelpower.com
Sr. Product Specialist                          www.pixelpower.com
Pixel Power Inc.                                 mgraves@mstvp.com

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245

I say start small and then go big... Oh I don't know a Proliant 1500 or
3000 should work nicely -- if you can handle the noise =)~
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com 
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of 
> Andrew Latham
> Sent: Monday, August 22, 2005 1:40 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Small office setup/using analog 
> lines w/ Asterisk
> 
> Or IRQs can be set via ACPI 
> 
> On 8/22/05, Kevin P. Fleming <kpfleming@digium.com> wrote:
> > jennyw wrote:
> > 
> > > AMD Sempron 2400+
> > > 3Ware Escalade 8006-2LP (2 channel SATA RAID) Asus A7N8X-E Deluxe
> > > 1 GB RAM
> > > Jetway Radeon 9000 64MB 128Bit 4X AGP DVI/TV Out Dual Head
> > 
> > This is a bad sign... are you running a graphical 
> environment on this 
> > machine?
> > 
> > > I'm not sure if I can change the IRQ setting on the 
> Radeon, but will 
> > > try. Can the Digium cards change IRQs?
> > 
> > No PCI card can 'change IRQs'. IRQ assignment is under the 
> control of 
> > the motherboard wiring and the BIOS settings.
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> 
> --
> ---
> Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) 
> lathama@lathama.com - lathama@yahoo.com - lathama@gmail.com 
> If any of the above are down we have bigger problems than my email!
> ---
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

I'm not having any problems with the SPA-841's at the moment. I have 15
of them in use right now. The other phones we use are the Polycom
IP30X's and they are really nice phones for that price range. I haven't
tried a "really expensive" phone so I may not know what I'm
missing.

-Jonathan
>I'm having some volume fluctuation problems with the SPA-841 phones, 
>which I have as well. The next phone I'm ordering will be something 
>that's a lot more expensive, so I'll let you know if that fixes
it...
>but I'd put a nice network card in your *server before I'd replace
all
>of your phones.
>-K
risk-users

"What good does RAID give you on writes?  None whatsoever.  RAID only
helps performance on reading."

Come again?  Writing to multiple hard drives in parallel is way faster
than writing the same file to one HDD.
You should Google the words RAID and Write Performance.

I assume you must have meant certain RAID levels are better than others.
If that was your meaning then you would be correct.

Cheers,
Wiley