IP Phone A <------> Asterisk IP PBX <---> Analog Phone B In my tests, echos were generated by IP Phone A when I turned on the speaker. As was pointed out, zaptel could not cancel these echos because the echo was not received at the zaptel interface, rather on the SIP interface.
I have a question : Does it mean that the SIP interface has to cancel the echo from IP Phone A ? This should be a real problem since it is possible that echo could be generated by IP Phones when they are in the handsfree mode (Speaker mode) or echo could be generated from an old IP Phone. > On Mon, 2006-07-17 at 17:04 +0800, Chan Kwang Mien wrote: > > Hi Paul, > > > > Thanks for your reply. > > > > The purpose of my tests is to verify that Echo Cancellation is up and > > running in the zaptel-1.2.6 that I have installed. > > > > In my test-bed, > > > > IP Phone A <------> Asterisk IP PBX <---> Analog Phone B > > Hopefully to put one more explination out here to make sure it is well > known why your problem isn't one that is solveable in Asterisk. > > Take your example above and break it down the way asterisk will see it. > > IP phone <-----> Asterisk > > IP phones are 4 wire equivalent so no echo should be introduced here. No > echo cancellation is needed > > Analog Phone <---- Asterisk > > Asterisk monitors this to gather data for echo cancel. > > Asterisk ----> Analog Phone > > Now asterisk knows what it sent and can try to remove it from the > returning audio stream. > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
