Tack on ${eventextra} to the sample cel_custom.conf.

git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@283207 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 10 insertions, 4 deletions

diff --git a/CHANGES b/CHANGES
index c4fd00ec2..681d7f0f5 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
 ------------
@@ -535,6 +540,10 @@ Miscellaneous
  * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
    have less than 3 lines on the LCD.
  * 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
 -----------
@@ -550,6 +559,7 @@ CLI Changes
    manager.conf. 
  * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
    the channel hangup request to all channels.
+ * Added a "core reload" CLI command that executes a global reload of Asterisk.
 
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2  -------------
@@ -1091,10 +1101,6 @@ SIP changes
     option is enabled, Asterisk will watch for a CNG tone in the incoming audio
     for a received call.  If it is detected, the channel will jump to the 
     'fax' extension in the dialplan.
-  * Improved NAT and STUN support.
-     chan_sip now can use port numbers in bindaddr, externip and externhost
-     options, as well as contact a STUN server to detect its external address
-     for the SIP socket. See sip.conf.sample, 'NAT' section.
   * The default SIP useragent= identifier now includes the Asterisk version
   * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
      If set, and the incoming request carries authentication info,