From: Leif Madsen <leif@leifmadsen.com>
Date: Wed, 28 Oct 2009 20:06:13 +0000 (+0000)
Subject: Update documentation in sip.conf.sample.
X-Git-Tag: 1.4.27-rc3~26
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=ff7b512bcc348acdad08cfc7ccf873bc35622a99;p=thirdparty%2Fasterisk.git

Update documentation in sip.conf.sample.

Update the documentation in sip.conf.sample in order to make it more clear
that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It
is only used to stop Asterisk from generating a reINVITE, but does not stop
it from accepting them if necessary.

(closes issue #15644)
Reported by: lmadsen

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@226382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 186ebd8f65..cd329b6111 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -361,6 +361,13 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; call directly between the endpoints instead of sending
                                 ; a re-INVITE).
 
+                                ; Additionally this option does not disable all reINVITE operations.
+                                ; It only controls Asterisk generating reINVITEs for the specific
+                                ; purpose of setting up a direct media path. If a reINVITE is
+                                ; needed to switch a media stream to inactive (when placed on
+                                ; hold) or to T.38, it will still be done, regardless of this
+                                ; setting.
+
 ;directrtpsetup=yes             ; Enable the new experimental direct RTP setup. This sets up
                                 ; the call directly with media peer-2-peer without re-invites.
                                 ; Will not work for video and cases where the callee sends