I
came into this with my eyes wide open. I have read ABSOLUTELY
EVERYTHING there is to be found on the net about avoiding echo problems
BEFORE I even attempted to create a production system. Since lots of
people are apparently using this in production environments now I just
assumed that echo IS avoidable.
As
others have recommended, I created a test system with the proposed
production parts. I bought a couple different SIP phones to try and a
Digium TDM01B card. I am using an older PIII 1Ghz system with
815chipset (PCI Rev2.2) with 256MB for my test system. The only thing
that will be different on a production system is that I will be using a
newer chipset PC with faster processor and 512MB. Probably Intel 7505,
7210, or 7211 chipsets which seem to be the most compatible with
Asterisk.
My
problem is that I cannot eliminate echo no matter what I try. I
seriously doubt that a newer chipset faster PC with more memory will
eliminate or even reduce my echo problems based on what I have
read. I am not about to drop more cash to try and find out.
Essentially, my findings are that Asterisk is NOT production capable
for my configuration which is via FXO and PSTN. That is probably THE
most common configuration so if it is not production capable like that
it isn't production capable period as far as I'm concerned. What a
disappointment :(.
Unless
I am missing something I am sure that many many people with a similar
configuration in a production environment have the same problem.
Perhaps they are just living with it?? For me it is just as
unacceptable on an Asterisk system as it is on a traditional PBX. Some
calls are ok and some are not. No correlation to local, long distance,
time of day. There always seems to be some echo. Sometimes it is
worse than other times. Again, no correlation to local, long distance,
time of day. Tried connecting to ATA adapter and using VoIP provider
instead to see if the telco was causing the problem. That did not
change anything. Still the same general echo problem
The
things I have tried include in no particular order and not limited to
are:
*Buy
latest TDM400P with latest FXO module
*Ensure copper
connection to analog telco lines and telco are not causing problems
including running a separate shielded line to the demarc AND having the
telco guy come out and test the levels, impedance etc.
*Adjust
RX/TX levels as per Asterisk Wiki using the quick Ztmonitor method and
by using the detailed Ztmonitor method via a Telco 102milliwatt test
phone #. The end result was RX=8.0, TX=-1.0. Since I still have echo
problems I have tried all sort of other settings without success.
*After
ALL of the above, try every possible combination of all of the
following on Asterisk v1.0.9: echocancel (off, on, 128, 256, 16, 32,
64), echowhenbridged (on, off), echotraining (off, on, 800), Mark
2 (default, aggressive, CVS head developments, bugs.digium.com patches,
adjust threshold level as per wiki etc. etc.)
*Make
sure echotraining line is before FXO channel assignment in zapata.conf
file
*Run
fxotune which did not find a need to adjust the FXO levels
(1=0,0,0,0,0,0,0,0)
Based
on all the above testing the best settings were pretty much in line
with what most people are finding.
echocancel=on.
echowhenbridged=on, echotraining=800, Mark 2 echo canceller, aggressive
cancellation OFF, bugs.digium.com #2820 patch, RX=8.0, TX=-1.0.
Still
have echo. Aggressive mode helps a bit but then the other persons
voice get's cut off a lot especially when I talk and the cutting in and
out of the canceller is more noticeable and objectionable in general
than if Aggressive is turned off.
I have
two SIP phones. An Aastra 9133i and a Grandstream GXP2000. Echo
problem is the same on both phones.
I
am located within a metropolitan area in Canada.
Any comments and/or suggestions would be greatly
appreciated as I am pretty much out of ideas and ready to give up on
Asterisk as a suitable traditional small business phone system
replacement.
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