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