[Asterisk-Users] Analog FXO Woes Continue

[Damon Estep]

We have also decided that FXO interfacing is not reliable enough, even when using the Digium 4 four port FXO card, the lines hang frequently and there have been various quality issues. All of our production deployments are PRI interface, and they are rock solid. While I have not done it myself, it seems that the solution to provide digital interface without the expense of a PRI would be ISDN using a card like the Eicon Diva BRI. No experience with this from our end, but in our region a BRI is less money than two busienss DS0s anyways.

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From: asterisk-users-bounces at lists.digium.com on behalf of Rich Adamson
Sent: Tue 12/7/2004 10:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Analog FXO Woes Continue



> I've been struggling with a test * install for a couple months now in a
> small office and am just about ready to give up on it.  It's not that the
> system itself is a problem.  I've got everything (attendant, voicemail,
> FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers)
> working except for the frigging analog FXO interfaces.  These things are
> driving me completely mad.  Since this is obvioiusly a deal breaker, I'm
> looking for any more suggestions on how I might fet these things working.
>
> The hitch is pretty clearly the quality of the lines I have from BellSouth
> but I can't get thim to identify anything wrong.  I have tried a Digium
> 1-port FXO card (can't remember part number and it's no longer on  their
> site, hmmm...) as well as a Sipura SPA3000.  With both of these
> interfaces, I'm getting consistent mis-dials on outbound calls, broken
> inbound fax-detection, broken DTMF detection in the attendant menus.
> Hours of adjustments to the gains on the Digium card only added echo and
> failed to reduce the offurenc of the other issues.  These same two
> interfaces worked fine on a line at my office so I'm pretty sure the issue
> is with the lines at the test site.
>
> So, what are my options here for interfacing with these lines?  Would the
> channel-bank route affect this?
>
> Thanks in advance for any suggestions,

Don't have any real answers, but might check the following... at least
to rule them out.

Telco folks _always_ check lines from their demarc (which in some cases
is the protector box on the outside of the building). Most will not come
inside to measure anything from the customer equipment jack. If that's
true in your case, then you have to question the cabling inside the
building (to asterisk). That cabling is most often simple inside wire
that can easily pick up noise (eg, induction from florescent lights,
motors, wall-wart transformers, some desk lamps). If you don't know
where the inside wire is run, might try to find out or bypass it with
cabling laying on the floor for at least an elementary test.

If you did not _see_ a telco person on site doing the transmission
checks, you have to assume that someone did them from the central
office (most common approach). That's okay in many cases, but its
not okay in other more serious cases. The majority of the telco
people that would be dispatched for testing only know enough to
follow printed procedures using whatever testset they've been given;
they don't have the skills to actually interpret the readings for
cases they've never seen or been trained to recognize.

Its not hard to plug an ordinary phone into the same rj11 jack
used by asterisk. Do it and listen close. Given the problems that
you've stated, it should not be difficult to hear noise, hum,
low volume, etc, if it is in fact bad lines. Also, compare lines; it
is not very often four of four lines go bad in exactly the same way.
Can you hear any difference between lines?

Bridge an ordinary phone on the same pstn line as asterisk. Place
some calls from asterisk and listen to what's going on via the
analog phone. (Example: some central offices don't like dtmf tones
within xxx milliseconds after going off-hook. You'll get wrong
numbers, etc. Insert the 'w' option in your Dial statement to
delay those dtmf tones a little bit.) To be a little sneaky,
unscrew and remove the mouthpiece from the analog phone and you
can monitor calls all day long without impacting asterisk's ability
to handle calls. If asterisk is having an echo issue (as an example)
and you don't hear it with the bridged phone, you at least know
where to look.

If you messed with the txgain/rxgain for your analog lines, go
back to zero gain, use
 echocancel=yes
 echotraining=800
 rxgain=0.0
 txgain=0.0
on each pstn line, reboot the server, and test using some of
the above steps to verify problems.

If you're still not sure what's going on, transmission test sets
are sold by many different companies that you can use from the
asterisk rj11 jack to prove line quality. New sets run about
$400 to $600 for what you need; check ebay for used pricing.
The telco's have a telephone number for a "quiet termination" and
another one for their "milliwatt generator". Get those numbers and
use the test set to measure noise (quiet termination) and loss
(milliwatt generator). If those results are reaonable, then you've
got an asterisk configuration problem (and/or digium card problem).

Rich


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