asterisk users - Aug 2005 - 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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On Wednesday 24 August 2005 16:37, canuck15 wrote:
> 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.
So in other words, everything will be changing on your production system.  Not 
a good way to start.
> 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 :(.
Most of us don't have any trouble.
> *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.
I'd be damn curious to know what you got out of this -- most telco guys will
do a basic metallic check, throw on a butt-set and say "yup, I got
dialtone."
-- hardly a real check but that's neither here nor there.  I'm also in
Canada
(1.5hrs from Toronto, ON) so I'm *really* curious who you got on the line to
do a real line test with you.  I have resorted to buying my own telco test 
equipment off ebay and using that, even though our techs here are excellent.
> *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.
Ok good.  Can you detail exactly what you did to reach these numbers?  I'm 
curious.
> *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.)
I'd posted something earlier that basically says this:  Without measured, 
controlled tests, you're just pissing up a rope.  Wildly changing settings 
and hoping for the best does nothing but cost you time and energy.
> *Run fxotune which did not find a need to adjust the FXO levels
> (1=0,0,0,0,0,0,0,0)
fxotune doesn't adjust FXO levels, it adjusts a very simple FIR filter which
is part of the DAA in the FXO module.  IMO it helps with audio quality but 
not much with echo.
> 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.
Agressive mode turns the phone line into a half-duplex environment.  When your 
voice energy is detected it mutes the receive audio.
> I have two SIP phones.  An Aastra 9133i and a Grandstream GXP2000.  Echo
> problem is the same on both phones.
Do you have echo between the two phones?  What about when calling out to a 
VOIP provider, dialing a DID you own that comes back in and hits the other 
phone?
> 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.
I haven't seen your zconfig.h nor your zaptel Makefile, and you didn't
tell us
anything about your network (network card, switch, etc.).

My general advice for zaptel is to do the following:
zaptel Makefile: underneath the comments about zconfig.h add
KFLAGS+=-march=pentium4 (or pentium3 or pentiumpro, use the exact proc)
CFLAGS+=-march=pentium4 (or pentium3 or pentiumpro, use the exact proc)

and in zconfig.h
- enable XLAW (optimize for small # of zap channels)
- enable MMX
- MARK2, no agressive mode.

Whenever I've done that my echo has largely disappeared.  

Have you also tried flipping tip and ring going into the TDM card?

-A.

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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of canuck15
Sent: Wednesday, August 24, 2005 1:38 PM
To: asterisk-users@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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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.
>  
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

The Asterisk Software is not the problem.  I'm thinking and I could be 
wrong that your having a total line balance mismatch with the card your 
using.  Check the line impedance and the card's.   Most people using 
Asterisk don't have that much echo. Anyway It would be nice to see a 
manual Hybrid adjustment on analog cards.

Don't give up.



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.
>  
>
>------------------------------------------------------------------------
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Try to use another land line and test the echo problem again.
Do you have any DSL service running in that line?


Quoting canuck15 <canuck15@hotmail.com>:
>
> 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.
>
>


-- 
Alfredo J. Fabretti
IPFLOW :: La inteligencia en sus comunicaciones
Argentina: (5411) 4294-8897
USA: (1) 914 301 8268
www.ip-flow.com.ar

Alfredo,

I tried a regular telco PSTN and a VoIP provider (webcall.ca using their
Nortel ATA connected to the TDM01B).  Both have very similar echo problems
using completely different wiring so I am quite convinced it has nothing to
do with the PSTN or wiring.

By the way, I sound just fine to the person on the other end.  They hear
absolutely no echo and say I sound crystal clear. 

I also want to say that I am encouraged at the optimistic responses so far.
It tells me that there is hope if so many people feel this can work today
with existing hardware.

-----Original Message-----
From: Alfredo J. Fabretti [mailto:ajf@ip-flow.com.ar]
Sent: Wednesday, August 24, 2005 3:18 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

Try to use another land line and test the echo problem again.
Do you have any DSL service running in that line?


Quoting canuck15 <canuck15@hotmail.com>:
>
> 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.
>
>


--
Alfredo J. Fabretti
IPFLOW :: La inteligencia en sus comunicaciones
Argentina: (5411) 4294-8897
USA: (1) 914 301 8268
www.ip-flow.com.ar

Ok, fxotune is a work in progress so to speak.  I fixed something in it about
a week ago that may help it adjust to the line better (whereas before I'm
not
sure that it was at all).  Try the latest CVS-HEAD version of fxotune as your
first step.  (oh, after you use fxotune you should turn off your gain settings
in zapata.conf).

Second step is to try the new echo canceller that was added to CVS-HEAD.  Look
in zconfig.h and try the KB1 echo canceller.  I have received many good reports
that it has cured practically all echo on all of the systems that I have heard
feedback from.

If all of this doesn't work, you probably have a serious hardware line issue
that you should resolve with your telco.

---
Matthew Fredrickson

Colin,

Your suggestions about identical hardware and BIOS revisions sound like good
advice.  If I ever get past this it is something I will definitely be
careful about.

Well I am starting to think that maybe it is my PC. I have a newer, faster
AMD system with PCI2.2 so I might give that one a shot.

You are using FXS modules with analog phones.  I am using SIP phones.  It is
my understanding that there is a LOT more delay when using SIP phones and of
course a lot of other things going on.

zttest yields 100% at least half the time and  99.987793% the rest of the
time.

Someone else suggested enabling MMX and CONFIG CALC XLAW in zconfig.h as
well as adding the following to the Zaptel Makefile:
KFLAGS+=-march=pentium3
CFLAGS+=-march=pentium3

I did all this, recompiled zaptel and rebooted but it didn't seem to make
any difference.

-----Original Message-----
From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com]
Sent: Wednesday, August 24, 2005 8:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

dude it's gotta be something with your system. Im using same setup at home
with a TDM22 with no probs. Asterisk@Home 1.5, Compaq Deskpro EN P3 500,
cordless phones.  You did a ton more than I did, I basically plugged
everything in and installed a@h.Worked first try. 

Hate to say it, but would it be possible to try a different system? Even an
old system as long as it is PCI 2.2.

The big clue here is that you say that echo is the same regardless of phone
or PSTN source. That pretty much narrows it down to the system. What's your
score on /usr/src/zaptel/zttest?

This is the burn of Asterisk. It's extremely difficult to make random
hardware work good with it. Because it is designed for commodity hardware,
stability is a moving target. Even bios revs can make a difference. In the
deployment I have done for my work I have a master Asterisk server and 30
IAX servers in remote locations. All 30 boxen are exactly the same, down to
the BIOS rev, Linux version, network cards, everything. This is the way to
do it, to narrow down infinite variables to a controllable few.  You need
patience or luck but preferably both. You have an advantage, though. From
your post it looks like you've done your homework, you aren't a dumbass,
and
you know how to arrive at a conclusion through process of elimination.
You're on the right track. Don't give up now, 'cause you are close.
> ----------
> From: 	canuck15
> Reply To: 	Asterisk Users Mailing List - Non-Commercial Discussion
> Sent: 	Wednesday, August 24, 2005 5:02 PM
> To: 	'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: 	RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm
> scared
> 
> <<File: ATT11994.txt>>
> Wiley,
>  
> The very first thing I checked was for IRQ problems.  I apoligize for 
> forgetting to mention that.
>  
> The only thing I found in the Google search you suggested is a thread 
> from January 2004 suggesting that businesses should ONLY use a T1 card 
> with a channel bank instead of X100P cards.  I understand the
"don't use
X100P"
> part but I just assumed the TDM400P with echo cancellation is working 
> fine now.
>  
> I am using a WRT54G (switch).  I am NOT connecting via wireless.  I 
> have no traffic to QoS.  Just VoIP and a WRT54G switch is quite 
> capable of that as far as I know.  No hubs!
>  
> I pretty much just use AAH now for it's ease of install but I have 
> rolled my own in the past.  The echo has persisted through AAH v1.3, 
> 1.4, 1.5.  I have recently (yesterday) installed the latest Asterisk 
> and Zaptel CVS head development tree just to see if it made a difference
and it DID NOT!
>  
>  
>  
>  
> 
>   _____
> 
> From: Wiley Siler [mailto:wsiler@education2020.com]
> Sent: Wednesday, August 24, 2005 2:00 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm 
> scared
> 
> 
> 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@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of canuck15
> Sent: Wednesday, August 24, 2005 1:38 PM
> To: asterisk-users@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.
>   
>

You did not specify anything about your network. If your network has a
big latency, echo cancellers can get into trouble. For instance, I
have echo problems just using wireless POTS phones on my sipura 2100
sip adapter/router on an otherwise unused 8Mbps ADSL internet
connection at home.

Lars Dybdahl.

On 8/24/05, canuck15 <canuck15@hotmail.com> wrote:
> My problem is that I cannot eliminate echo no matter what I try.

So bottom line please.

Have we decided that it is STILL correct to set RX/TX gain for 14800 with
ztmonitor quantitative using a telco 1004hz 0dbm test phone number?  If not,
what should we set it to with ztmonitor.  

-----Original Message-----
From: Rich Adamson [mailto:radamson@routers.com]
Sent: Thursday, August 25, 2005 8:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared
> I'll do my comments in line and hope I don't offend.
> 
> Rich Adamson wrote:
> >>First off, thank you *very* much for this unbelievably informative 
> >>post!  I've got it saved away now along with Kris
Boutilier's
> >>adjusting rxgain/txgain post.
> >>
> >>On Wednesday 24 August 2005 17:14, Bruce Ferrell wrote:
> >>
> >>>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.
> > 
> > 
> > If I read the above words exactly as written, the above is not true. 
> > Maybe there was a different intent that I'm missing, or, maybe
words
left out?
> 
> I'm a lousy typist :)
> 
> > I'm reading the words to say "if I put a transmission test
set on
> > the cable pair just before the pair leaves the central office, the 
> > reading should be in the -2 to -3 dbm range." If that is what you
> > meant, then its incorrect. Even the old analog step-by-step switch 
> > specs called for no more then .5db loss from the milliwatt generator 
> > to the cable pair (CO distribution frame).
> 
> > If you mean placing a transmission test set at the customer's
demarc
> > (at the customer's site), the -2 to -3 db is still incorrect for
"analog"
> > pstn circuits. That level _will be_ the 0db generator tone minus the 
> > cable loss from the CO to the customer's demarc. That cable loss
is
> > 100% predictable if you know the length and gauge of the copper 
> > wires between the central office and the customer's site. (That
"is"
> > exactly how the engineering spec is set for the less technical 
> > telephone installers to measure after installing a new pstn facility 
> > to a customer site.)
> 
> at the last point leaving the CO, the tone level should be a nominal 
> 0dbm.  By the time it get's to the customer demarc, -2 to -3 dbm.  The 
> loops are "suppposed" to be engineered that way.  On a brand
spanky
> new loop, yes 100% predictable.  Over time, all sorts of oddities 
> (corrosion, half taps, loading coils, and just general funkieness) are 
> introduced in the real world.
The -2 to -3 db is not correct for analog circuits. Copper wires have a loss
that is directly related to the length of the cable. (I don't have the chart
right here, but a 7,000 foot cable pair will have lets say 6db of loss and a
3,000 foot pair will be a 3db loss. You can't engineer something into a
copper pair to compensate for that loss.)

The only thing that I can think of that you might be talking about is using
an old analog carrier system on a copper pair. If that's what you're
thinking, then yes -2 to -3 db is very reasonable.

Assuming I am an average asterisk install an average distance from the CO
with average ears with current stable asterisk code (v1.0.9) looking for
'assistance' for optimal values knowing full well that it may not be
optimal...........I ask again, what should I set ztmonitor quantitative to. 

-----Original Message-----
From: Rich Adamson [mailto:radamson@routers.com]
Sent: Friday, August 26, 2005 1:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

Bottom line... ztmonitor can be used to 'assist' in setting some
starting
values, but the further your asterisk box is from the central office, the
more likely the gain values will have to be adjusted lower then what you
want, and may very well appear off-scale with ztmonitor.

Given the curent code and issues, using your ears instead of ztmonitor will
lead to better results, period. (Before lots of people jump on this and say
it does work, please reread the "further you are from the CO"
words again. Yes, ztmonitor can be used with low-loss pstn loops; no, it
will not provide anything close to an optimal circuit for higher-loss
loops.)

------------------------
> So bottom line please.
> 
> Have we decided that it is STILL correct to set RX/TX gain for 14800 
> with ztmonitor quantitative using a telco 1004hz 0dbm test phone 
> number?  If not, what should we set it to with ztmonitor.
> 
> -----Original Message-----
> From: Rich Adamson [mailto:radamson@routers.com]
> Sent: Thursday, August 25, 2005 8:20 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm 
> scared
> 
> > I'll do my comments in line and hope I don't offend.
> > 
> > Rich Adamson wrote:
> > >>First off, thank you *very* much for this unbelievably
informative
> > >>post!  I've got it saved away now along with Kris
Boutilier's
> > >>adjusting rxgain/txgain post.
> > >>
> > >>On Wednesday 24 August 2005 17:14, Bruce Ferrell wrote:
> > >>
> > >>>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.
> > > 
> > > 
> > > If I read the above words exactly as written, the above is not
true.
> > > Maybe there was a different intent that I'm missing, or,
maybe
> > > words
> left out?
> > 
> > I'm a lousy typist :)
> > 
> > > I'm reading the words to say "if I put a transmission
test set on
> > > the cable pair just before the pair leaves the central office,
the
> > > reading should be in the -2 to -3 dbm range." If that is
what you
> > > meant, then its incorrect. Even the old analog step-by-step
switch
> > > specs called for no more then .5db loss from the milliwatt 
> > > generator to the cable pair (CO distribution frame).
> > 
> > > If you mean placing a transmission test set at the customer's
> > > demarc (at the customer's site), the -2 to -3 db is still 
> > > incorrect for
> "analog"
> > > pstn circuits. That level _will be_ the 0db generator tone minus 
> > > the cable loss from the CO to the customer's demarc. That
cable
> > > loss is 100% predictable if you know the length and gauge of the 
> > > copper wires between the central office and the customer's
site. (That
"is"
> > > exactly how the engineering spec is set for the less technical 
> > > telephone installers to measure after installing a new pstn 
> > > facility to a customer site.)
> > 
> > at the last point leaving the CO, the tone level should be a nominal 
> > 0dbm.  By the time it get's to the customer demarc, -2 to -3 dbm.
> > The loops are "suppposed" to be engineered that way.  On a
brand
> > spanky new loop, yes 100% predictable.  Over time, all sorts of 
> > oddities (corrosion, half taps, loading coils, and just general
> > funkieness) are introduced in the real world.
> 
> The -2 to -3 db is not correct for analog circuits. Copper wires have 
> a loss that is directly related to the length of the cable. (I don't 
> have the chart right here, but a 7,000 foot cable pair will have lets 
> say 6db of loss and a 3,000 foot pair will be a 3db loss. You can't 
> engineer something into a copper pair to compensate for that loss.)
> 
> The only thing that I can think of that you might be talking about is 
> using an old analog carrier system on a copper pair. If that's what 
> you're thinking, then yes -2 to -3 db is very reasonable.
> 
> 
> 
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---------------End of Original Message-----------------