Asterisk 1.4.2 no Voice on SIP users


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 ?

Try canreinvite = no. I had that same problem. From version 1.4.1 the direct bridging is doing that on SIP. I can hear only something in the beginning of the connection (less than a second) and then silence - the connection is on but no sound.