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

Paul digium-list at 9ux.com
Wed Aug 24 14:28:08 MST 2005


Has anyone tried this approach?

1) Install * on a PC(probably don't need much horsepower)

2) Setup a sipura spa-2000 ata so that it is not on the same lan you are 
troubleshooting. One way to do this is with a crossover cable to the 
above PC. Restrict both ata ports to ulaw only.

3) Port 1 of ata gets a good analog phone

4) Port 2 simulates a pots line. Run a quality short cable to the fxo 
you are testing.

This way you can also test things like caller ID without paying a telco.

I suppose you could also use the ata that comes with vonage and others 
to test an fxo. As long as you get good call quality it should work.


Wiley Siler wrote:

> 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.
>  
>
>------------------------------------------------------------------------
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at 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
>




More information about the asterisk-users mailing list