[Asterisk-Users] Will Echo problems EVER be solved, I'm scared
canuck15
canuck15 at hotmail.com
Wed Aug 24 17:06:52 MST 2005
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 at ip-flow.com.ar]
Sent: Wednesday, August 24, 2005 3:18 PM
To: asterisk-users at 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 at 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
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