[Asterisk-Users] Small office setup/using analog lines w/ Asterisk

[Andrew Kohlsmith]

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.