Alessandro Russo wrote: > Unfortunately echo is not due to speakerphone. Each participant calls > a geographical number that is redirected from the PBX to a call > manager which pass the flow to the asterisk machine which creates a > meetme voice conference, so user calls via traditional either fixed or > mobile phone. Therefore they cannot mute their phone while they aren't > speak :( > Moreover the echo problem occurs when we do tests within the same > phone-cloud, in our organization phones are connected through some > cisco call managers, so when a phone calls the internal number ABCD > the flow arrives to the call manger which forward it to the asterisk, > this is the path done: phone <=> call manager <=> asterisk > and also in internal cloud we experienced echo problems with more than > 2 participants, not all the conversation is affected by echo, > sometimes there is echo and sometimes not. > > I performed the zttest and I obtained the following results: > > asterisk:~# zttest > Opened pseudo zap interface, measuring accuracy... > 99.966690% 99.971863% 99.936729% 99.967766% 99.936913% 99.968163% > 99.967667% > 99.936623% 99.969818% 99.937019% 99.967972% 99.937012% 99.968063% > 99.967865% 99.936440% > 99.967766% 99.935356% 99.967667% 99.937401% 99.968460% 99.967667% > 99.936333% > --- Results after 22 passes --- > Best: 99.972 -- Worst: 99.935 -- Average: 99.955330, Difference: 99.992836
Alessandro, I'm sorry to hear that your problem isn't acoustic echo. I'll be following this thread to see if anyone offers you any suggestions and I'll let you know if I discover anything that improves the echo problem in my installation. What is the timing source in the conference server? In general, it will be either a Zaptel/DAHDI hardware device or the ztdummy/dahdi-dummy module. See this page <http://www.voip-info.org/wiki/view/Asterisk+timer> for details. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
