Update of docs

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45333 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index b6e84fa..504b7c7 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -264,6 +264,11 @@ srvlookup=yes                      ; Enable DNS SRV lookups on outbound calls
                                ; behind a NAT, or for some other reason wants Asterisk to
                                ; stay in the audio path, you may want to turn this off.
 
+                               ; This setting also affect direct RTP
+                               ; at call setup (a new feature in 1.4 - setting up the
+                               ; call directly between the endpoints instead of sending
+                               ; a re-INVITE).
+
 ;canreinvite=nonat             ; An additional option is to allow media path redirection
                                ; (reinvite) but only when the peer where the media is being
                                ; sent is known to not be behind a NAT (as the RTP core can