[asterisk-users] Meetme conference room duplex issue

[jamespev]

  We are using only SIP trunks for our provider.(we have no POTS hardware)  Is there an aggressive echo cancellation setting in this case?
  Could this be related to the audio buffers setting in meetme.conf?
  Thanks for the ideas!
James> Are you using zap channels with 'aggressive' echo suppression enabled?  
> That will make calls pretty half-duplex.
> 
> Moj
> 
> jamespev wrote:>>>    Hello.  We are very successfully using asterisk (in the form of >> trixbox 2.2/asterisk 1.2).  We run a few conference lines for customer >> teleconferences which mostly work well but they seem to operate at >> half duplex.  If a person starts talking they will cut off others on >> the call.  Is this normal behavior?  Are there any options I can >> change to change this?>>>>    Thanks!>>>> James
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