Try to use another land line and test the echo problem again. Do you have any DSL service running in that line?
Quoting canuck15 <[EMAIL PROTECTED]>:
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 _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- 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
