remove a couple of entries that got duplicated and snuck into the SIP section. Also, align the NAT/STUN entry with the others.

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76985 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 4 insertions, 7 deletions

diff --git a/CHANGES b/CHANGES
index 9a03a3da0..b43b2442d 100644
--- a/CHANGES
+++ b/CHANGES
@@ -42,13 +42,10 @@ CLI Changes
 
 SIP changes
 -----------
-   * 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.
-   * Added the bindaddr option to gtalk.conf.
-   * Added the ability to specify arguments to the Dial application when using
-     the DUNDi switch 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,