OK comments on echo and levels.
I made a living doing this in a central office so take it for what it's
worth.
Milliwatt is 0dbm0 or 0dbm at a 0 reference point.
At the point where the phone line get's to your demarc the is supposed
to ba a -2 to 3db reference point, sometimes called a -2 or -3 test
level point (TLP). So that milliwatt tone at that point should read in
the range of -2 to -3 dbm.
Voice BTW, is considered to be a nominal -15dbm0.
The digital stream of a T1/E1 is considered to be a 0 reference point.
When I worked on telephone switches (NorTel DMS250) the entire switch,
because it was all digital was considered to be a 0 TLP.
If the milliwatt is arriving at the demarc at the nominal -2 to -3dbm
and getting into the asterisk to be measured at 8dBm (+8dbm0), I'd say
something is grossly mal-adjusted. You're seeing 8db of gain!
Fix that and your echo should go away.
P.S.
With that much gain, there is no echo cancellor that I know that can
cope, hard or soft.
canuck15 wrote:
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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