Requested changes documenting the fixed AEL functionality.

........

Merged revisions 358859 from http://svn.asterisk.org/svn/asterisk/branches/1.8

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@358860 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/CHANGES b/CHANGES
index ae351ecb985e9ead246e11ceabb15210a0bc9d31..d947b10c7d74378a0a307bebe0d683c78ee969c9 100644 (file)
--- a/CHANGES
+++ b/CHANGES
@@ -8,6 +8,16 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes since Asterisk 10.3.0 ------------------------------
+------------------------------------------------------------------------------
+
+Gosub changes
+-------------
+ * A new function, STACK_PEEK, has been added, to correlate for functionality
+   available in AEL in 1.4 that disappeared in 10.  STACK_PEEK permits the
+   user to see the location of the calling Gosub from within the subroutine.
+
 ------------------------------------------------------------------------------
 --- Functionality changes since Asterisk 10.1.0 ------------------------------
 ------------------------------------------------------------------------------

diff --git a/UPGRADE.txt b/UPGRADE.txt
index be641bfa2b634eb08a95f528686a5d68d988b564..565a0dd8088f60ef6838d141ae60f2fb687dcb59 100644 (file)
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -19,6 +19,16 @@
 ===
 ===========================================================
 
+From 10.3 to 10.4:
+
+* In AEL dialplans, the "h" extension will now be inherited from prior
+  calling contexts, just as it had in 1.4.  If you have created an AEL
+  dialplan from scratch in earlier versions of 10, you may want to check
+  that the execution of "h" extensions in earlier contexts is what you
+  want.  If you want to interrupt this functionality, simply placing an
+  "h" extension in the macro where you want no inheritance to take place
+  should be sufficient.
+
 From 10.2 to 10.3:
 
 * If no transport is specified in sip.conf, transport will default to UDP.