Update CHANGES and UPGRADE.txt for 13.27.0

CHANGES

@@ -12,6 +12,43 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.26.0 to Asterisk 13.27.0 ----------
+------------------------------------------------------------------------------
+
+Dial
+------------------
+ * Add RINGTIME and RINGTIME_MS variables containing respectively seconds and
+   milliseconds between creation of the dialing channel and receiving the first
+   RINGING signal
+
+   Add PROGRESSTIME and PROGRESSTIME_MS variables analogous to the above with respect to
+   the PROGRESS signal. Shorter of these two times should be equivalent to
+   the PDD (Post Dial Delay) value
+
+   Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get millisecond resolution
+   versions of DIALEDTIME and ANSWEREDTIME
+
+RTP/ICE
+------------------
+ * You can now indicate that you'd like an ice_host_candidate's local address
+   to be published as well as the mapped address.  See the sample rtp.conf
+   for more information.
+
+res_pjsip
+------------------
+ * Added a new PJSIP global setting called norefersub.
+   Default is true to keep support working as before.
+
+   res_pjsip_refer configures PJSIP norefersub capability accordingly.
+
+   Checks the PJSIP global setting value.
+   If it is true (default) it adds the norefersub capability to PJSIP.
+   If it is false (disabled) it does not add the norefersub capability
+   to PJSIP.
+
+   This is useful for Cisco switches that do not follow RFC4488.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.25.0 to Asterisk 13.26.0 ----------
 ------------------------------------------------------------------------------

doc/CHANGES-staging/app_dial_ringtime_progresstime.txt

@@ -1,12 +0,0 @@
-Subject: Dial
-
-Add RINGTIME and RINGTIME_MS variables containing respectively seconds and
-milliseconds between creation of the dialing channel and receiving the first
-RINGING signal
-
-Add PROGRESSTIME and PROGRESSTIME_MS variables analogous to the above with respect to
-the PROGRESS signal. Shorter of these two times should be equivalent to
-the PDD (Post Dial Delay) value
-
-Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get millisecond resolution
-versions of DIALEDTIME and ANSWEREDTIME

doc/CHANGES-staging/res_pjsip_add_norefersub_global_config.txt

@@ -1,13 +0,0 @@
-Subject: res_pjsip
-
-Added a new PJSIP global setting called norefersub.
-Default is true to keep support working as before.
-
-res_pjsip_refer configures PJSIP norefersub capability accordingly.
-
-Checks the PJSIP global setting value.
-If it is true (default) it adds the norefersub capability to PJSIP.
-If it is false (disabled) it does not add the norefersub capability
-to PJSIP.
-
-This is useful for Cisco switches that do not follow RFC4488.

doc/CHANGES-staging/rtp_ice_include_local_address.txt

@@ -1,5 +0,0 @@
-Subject: RTP/ICE
-
-You can now indicate that you'd like an ice_host_candidate's local address
-to be published as well as the mapped address.  See the sample rtp.conf
-for more information.