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

[Bruce Ferrell]

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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