From: David Vossel <dvossel@digium.com>
Date: Fri, 13 Aug 2010 20:11:58 +0000 (+0000)
Subject: res_stun_monitor and corresponding options CHANGES documentation
X-Git-Tag: 1.8.0-beta4~34
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=5b3270acc20fa19b0f34fea430f8f4de65e43282;p=thirdparty%2Fasterisk.git

res_stun_monitor and corresponding options CHANGES documentation


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/CHANGES b/CHANGES
index 52dc4e64e4..df6f220abd 100644
--- a/CHANGES
+++ b/CHANGES
@@ -73,6 +73,8 @@ SIP Changes
    RTP has been outfitted with the same abilities.
  * Added support for setting the Max-Forwards: header in SIP requests. Setting is
    available in device configurations as well as in the dial plan.
+ * Addition of the 'subscribe_network_change' option for turning on and off
+   res_stun_monitor module support in chan_sip.
 
 IAX2 Changes
 -----------
@@ -82,6 +84,9 @@ IAX2 Changes
    encryption is being used. This interoperates with the SIP SRTP implementation
    so that a secure SIP call can be bridged to a secure IAX call when the
    dialplan requires bridged channels to be "secure".
+ * Addition of the 'subscribe_network_change' option for turning on and off
+   res_stun_monitor module support in chan_iax.
+
 
 MGCP Changes
 ------------
@@ -537,6 +542,8 @@ Miscellaneous
  * Realtime now supports database failover.  See the sample extconfig.conf for details.
  * The addition of improved translation path building for wideband codecs.  Sample
    rate changes during translation are now avoided unless absolutely necessary.
+ * The addition of the res_stun_monitor module for monitoring and reacting to network
+   changes while behind a NAT.
 
 CLI Changes
 -----------