From afceb3e4aa15c8763935e4a4e89075ccacd55060 Mon Sep 17 00:00:00 2001
From: Joshua Colp <jcolp@digium.com>
Date: Wed, 8 Aug 2007 13:52:13 +0000
Subject: [PATCH] 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.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
 configs/sip.conf.sample | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 842c7b513a..c80ca87ccf 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -403,7 +403,8 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 				; 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
-- 
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