[asterisk-commits] file: branch 1.4 r78569 - /branches/1.4/configs/sip.conf.sample

[SVN commits to the Asterisk project]

Author: file
Date: Wed Aug  8 08:51:01 2007
New Revision: 78569

URL: http://svn.digium.com/view/asterisk?view=rev&rev=78569
Log:
(closes issue #10335)
Reported by: adamgundy
Update sip.conf to include another scenario where directrtpsetup will fail.

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?view=diff&rev=78569&r1=78568&r2=78569
==============================================================================
--- branches/1.4/configs/sip.conf.sample (original)
+++ branches/1.4/configs/sip.conf.sample Wed Aug  8 08:51:01 2007
@@ -315,7 +315,8 @@
 				; the call directly with media peer-2-peer without re-invites.
 				; Will not work for video and cases where the callee sends 
 				; RTP payloads and fmtp headers in the 200 OK that does not match the
-				; callers INVITE.
+				; callers INVITE. This will also fail if canreinvite is enabled when
+				; the device is actually behind NAT.
 
 ;canreinvite=nonat		; An additional option is to allow media path redirection
 				; (reinvite) but only when the peer where the media is being