<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=us-ascii">
<META content="MSHTML 6.00.2900.2722" name=GENERATOR></HEAD>
<BODY>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2>Just because you cannot get it to work does not mean that
IT does not work. </FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2>Just using the right motherboard is not enough. Did
you check for IRQ problems? You don't mention whether you have checked for
this.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2>Look for a thread called "Asterisk-Users Small office
setupusing analog lines w Asterisk" in the archive via
Google.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2>use site:lists.digium.com</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2>Try all the things listed in that
thread.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2>Do you have a network that is capable of VoIP? Are
you using hubs when you should be using switches?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2>There is a major difference and hubs WILL NOT work reliably
with VoIP.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2>Are you using QoS on your switches if you have lots of
network traffic?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005></SPAN><SPAN
class=907554520-24082005><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2>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.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2>I am putting my money on an IRQ issue
myself.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2>W</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=907554520-24082005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of
</B>canuck15<BR><B>Sent:</B> Wednesday, August 24, 2005 1:38 PM<BR><B>To:</B>
asterisk-users@lists.digium.com<BR><B>Subject:</B> [Asterisk-Users] Will Echo
problems EVER be solved, I'm scared<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>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.
</FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>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. </FONT></SPAN><SPAN
class=593295119-24082005><FONT face=Arial size=2></FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>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 :(. </FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>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</FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>The things I have
tried include in no particular order and not limited to
are:</FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>*Buy latest TDM400P
with latest FXO module</FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>*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.</FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>*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.</FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>*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.)</FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>*Make sure
echotraining line is before FXO channel assignment in zapata.conf
file</FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>*Run fxotune which
did not find a need to adjust the FXO levels
(1=0,0,0,0,0,0,0,0)</FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>Based on all the
above testing the best settings were pretty much in line with what most people
are finding. </FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>echocancel=on.
echowhenbridged=on, echotraining=800, Mark 2 echo canceller, aggressive
cancellation OFF, bugs.digium.com #2820 patch, RX=8.0,
TX=-1.0.</FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>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.</FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>I have two SIP
phones. An Aastra 9133i and a Grandstream GXP2000. Echo problem is
the same on both phones.</FONT> </SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>I am located within
a metropolitan area in Canada.</FONT></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=593295119-24082005><SPAN class=593295119-24082005><FONT
face=Arial size=2>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.</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=593295119-24082005><FONT face=Arial size=2>
</FONT></SPAN></DIV></BODY></HTML>