asterisk users - Jun 2011 - Question regarding progressinband

Hello,
I have question regarding the changes that are made in the sip
protocol in Asterisk - the option progressinband.
When this option is set to yes in asterisk version 1.4.21.1 - the call flow is:

sip.conf:
progressinband=yes

Device                    Asterisk
-----------INVITE SDP--------->
<---------100 Trying------------
<-----183 Session Prgoress--

After version 1.4.2X+ (tested with 1.4.36/1.4.41.1) the call flow changes to:

Device                    Asterisk
-----------INVITE SDP--------->
<---------100 Trying------------
<---------180 Ringing----------
<-----183 Session Prgoress--