From a8a26ad38999d88613248835b4b88e80622f007a Mon Sep 17 00:00:00 2001 From: Olle Johansson Date: Tue, 17 Oct 2006 17:51:34 +0000 Subject: Update of docs git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45333 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- configs/sip.conf.sample | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'configs') diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index b6e84fa91..504b7c7c9 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -264,6 +264,11 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; behind a NAT, or for some other reason wants Asterisk to ; stay in the audio path, you may want to turn this off. + ; 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 ; sent is known to not be behind a NAT (as the RTP core can -- cgit v1.2.3