[svn-commits] oej: branch 1.4 r45329 - /branches/1.4/configs/sip.conf.sample

[svn-commits at lists.digium.com]

Author: oej
Date: Tue Oct 17 12:39:18 2006
New Revision: 45329

URL: http://svn.digium.com/view/asterisk?rev=45329&view=rev
Log:
Adding information about Marks direct-RTP hack to the docs...

Modified:
    branches/1.4/configs/sip.conf.sample

Modified: branches/1.4/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/branches/1.4/configs/sip.conf.sample?rev=45329&r1=45328&r2=45329&view=diff
==============================================================================
--- branches/1.4/configs/sip.conf.sample (original)
+++ branches/1.4/configs/sip.conf.sample Tue Oct 17 12:39:18 2006
@@ -255,6 +255,11 @@
 				; 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.
+
+				; In Asterisk 1.4 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