Merged revisions 337219 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/10

........
  r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13 lines
  
  Make ast_pbx_run() not default to s@default if extension is not found
  
  Review: https://reviewboard.asterisk.org/r/1446/
  
  This is a bug - or architecture mistake - that has been in Asterisk for a 
  very long time. It was exposed by the AMI originate action and possibly
  some other applications. Most channel drivers checks if an extension
  exists BEFORE starting a pbx on an inbound call, so most calls will
  not depend on this issue.
  
  Thanks everyone involved in the review and on IRC and the mailing list
  for a quick review and all the feedback.

  (closes issue ASTERISK-18578)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337220 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 16 insertions, 1 deletions

diff --git a/CHANGES b/CHANGES
index 32b2719b9..c068de2d7 100644
--- a/CHANGES
+++ b/CHANGES
@@ -249,7 +249,7 @@ Asterisk Modules
 
 IAX2 Changes
 ------------
-* authdebug is now disabled by default. To enable this functionaility again
+ * authdebug is now disabled by default. To enable this functionaility again
    set authdebug = yes in iax.conf.
 
 RTP Changes
@@ -257,6 +257,21 @@ RTP Changes
  * The rtp.conf setting "strictrtp" is now enabled by default. In previous
    releases it was disabled.
 
+PBX Core
+--------
+ * The PBX core previously made a call with a non-existing extension test for
+   extension s@default and jump there if the extension existed.
+   This was a bad default behaviour and violated the principle of least surprise.
+   It has therefore been changed in this release. It may affect some
+   applications and configurations that rely on this behaviour. Most channel
+   drivers have avoided this for many releases by testing whether the extension
+   called exists before starting the PBX and generating a local error.
+   This behaviour still exists and works as before.
+
+   Extension "s" is used when no extension is given in a channel driver,
+   like immediate answer in DAHDI or calling to a domain with no user part
+   in a SIP uri.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
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