[asterisk-commits] file: trunk r78570 - in /trunk: ./ configs/sip.conf.sample

[SVN commits to the Asterisk project]

Author: file
Date: Wed Aug  8 08:52:13 2007
New Revision: 78570

URL: http://svn.digium.com/view/asterisk?view=rev&rev=78570
Log:
Merged revisions 78569 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r78569 | file | 2007-08-08 10:51:01 -0300 (Wed, 08 Aug 2007) | 4 lines

(closes issue #10335)
Reported by: adamgundy
Update sip.conf to include another scenario where directrtpsetup will fail.

........

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?view=diff&rev=78570&r1=78569&r2=78570
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Wed Aug  8 08:52:13 2007
@@ -403,7 +403,8 @@
 				; 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