[asterisk-commits] oej: trunk r45333 - in /trunk: ./ configs/sip.conf.sample

[asterisk-commits at lists.digium.com]

Author: oej
Date: Tue Oct 17 12:51:34 2006
New Revision: 45333

URL: http://svn.digium.com/view/asterisk?rev=45333&view=rev
Log:
Update of docs

Modified:
    trunk/   (props changed)
    trunk/configs/sip.conf.sample

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?rev=45333&r1=45332&r2=45333&view=diff
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Tue Oct 17 12:51:34 2006
@@ -263,6 +263,11 @@
 				; The default setting is YES. If you have all clients
 				; 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