The single most usefull tool that anyone outside telco consultants is likely to have is ztmonitor.
If you do a ztmonitor [channel number] -v you will get a visual of the sound strengths and it's pretty easy to see when rx or tx are out of balance... Now, if only that would help fix the low-level static noise I have on the x100p, that would be great. Chris. On Thu, 22 Apr 2004, Rich Adamson wrote: > > I do feel the echo cancellation does need some work. > > > > Currently, other than listening to users, there is no way to benchmark or > > trouble shoot echo problems. > > Sure there are, it's just that 99% of the asterisk implementors don't > have the test equipment to do it, and a good share probably wouldn't > know how to do it if they had access to the equipment. > > > We find that 2 to 3 out of every 20 calls will experience echo. While > > echo is a problem that naturally occurs from SIP - PSTN and vice versa, I > > am still baffled by the fact that the cancellation works randomly. > > > > When doing a zap show channel X, it will also report that the cancellation > > is still on. We experience the most echo with a T100P --> Adtran TA 750 > > FXO modules. While I understand these do not have impedence matching, I > > wonder to myself why echo cancellation works sometimes, and others not. > > > > Looking at Network util, processor util, and memory utilization, they do > > not provide any clear indication as to why /when it occurs. > > Not likely to have any impact whatsoever. > > > Is there anything more that can be done to debug echo cancellation, and > > further are other users experiencing this random echo. I know it was > > discussed before, but the support folks at digium aren't able to offer > > anymore help. > > You've probably read enough from previous postings to know there are > several different locations within an end-to-end voice call where echo can > creap into a system. In very general terms, any place where an end-to-end > channel incures a two-wire to four-wire conversion (whether done in hardware > or software), echo can creap in due to lots of different reasons. Since > asterisk provides us with lots of configuration choices, hardly any two > systems are alike. Therefore, don't know that anyone is going to write > * code anytime soon to correct something that can't be pointed to, etc. > > Someone mentioned they have echo on sip to sip calls (presumably the call > was processed by a single * system). If they do, the problem is likely > in the sip phone as there are no echo cancallation needs in that four-wire > end-to-end call from an * perspective. > > I've got a fair amount of test equipment (and 20+ years telephony > background), and am planning to assemble a document identifying some of > the pstn issues, level settings, and other things impacting a reasonable > system implementation. Unless someone wants to UPS a transmission test > set to me quickly, the document won't be completed for several weeks. > (The only test set I have access to will not be released for a couple > of weeks due to classes, etc.) > > I'm also expecting these tests to point out a number of other transmission > issues within asterisk that we'll get documented with real numbers, etc. > > Rich > > > -- chris maresca senior partner - www.olliancegroup.com linux, up 17 days _______________________________________________ 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