]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
(closes issue #10335)
authorJoshua Colp <jcolp@digium.com>
Wed, 8 Aug 2007 13:51:01 +0000 (13:51 +0000)
committerJoshua Colp <jcolp@digium.com>
Wed, 8 Aug 2007 13:51:01 +0000 (13:51 +0000)
Reported by: adamgundy
Update sip.conf to include another scenario where directrtpsetup will fail.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@78569 65c4cc65-6c06-0410-ace0-fbb531ad65f3

configs/sip.conf.sample

index a533f64478427504253dba19112a0cdd1521cfcc..10c38b195eb70d7926782a60aac8aa22ad129ab9 100644 (file)
@@ -315,7 +315,8 @@ 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