[Asterisk-Users] Will Echo problems EVER be solved, I'm scared

[Wiley Siler]

Just because you cannot get it to work does not mean that IT does not
work.  
 
Just using the right motherboard is not enough.  Did you check for IRQ
problems?  You don't mention whether you have checked for this.
Look for a thread called "Asterisk-Users Small office setupusing analog
lines w Asterisk" in the archive via Google.
use site:lists.digium.com
Try all the things listed in that thread.
 
Do you have a network that is capable of VoIP?  Are you using hubs when
you should be using switches?
There is a major difference and hubs WILL NOT work reliably with VoIP.
Are you using QoS on your switches if you have lots of network traffic?
 
If you are using your own Distro and installing from scratch, try to use
Asterisk at Home just to see if you still have the same problem.
 
I am putting my money on an IRQ issue myself.
 
W
 
 
 
 
 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of canuck15
Sent: Wednesday, August 24, 2005 1:38 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared


 
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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