In Asterisk 1.4.0 all was OK I have voice beetwen SIP phone and SIP operator. Under 1.4.1 and 1.4.2 i don’t hear any voice. Configuration files is that same.
I see in my 3CX Phone that codec called “In” is not established, codec “Out” is established ass Ulaw or Alaw depend on what I choose.
Any ideas what was changed ?