This commit introduces COLP/CONP and Redirecting party information into Asterisk.

The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.

A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.

This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.

(closes issue #8824)
Reported by: gareth

Review: http://reviewboard.digium.com/r/201



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 39 insertions, 3 deletions

diff --git a/CHANGES b/CHANGES
index 88ae3f515..a9205024f 100644
--- a/CHANGES
+++ b/CHANGES
@@ -7,7 +7,6 @@
 === and the other UPGRADE files for older releases.
 ===
 ======================================================================
-
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.6.3  -------------
 ------------------------------------------------------------------------------
@@ -23,10 +22,47 @@ Applications
    present, those values are sent immediatly upon receiving a PROGRESS message
    regardless if the call has been answered or not.
 
-Functions
----------
+Dialplan Functions
+------------------
+ * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
+   setting various connected line and redirecting party information.
  * The CHANNEL() function now supports the "name" option.
 
+Queue changes
+-------------
+  * A new option, 'I' has been added to both app_queue and app_dial.
+    By setting this option, Asterisk will not update the caller with
+    connected line changes or redirecting party changes when they occur.
+
+mISDN channel driver (chan_misdn) changes
+----------------------------------------
+  * Added display_connected parameter to misdn.conf to put a display string
+    in the CONNECT message containing the connected name and/or number if
+    the presentation setting permits it.
+  * Added display_setup parameter to misdn.conf to put a display string
+    in the SETUP message containing the caller name and/or number if the
+    presentation setting permits it.
+  * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
+    indicate the dialplan settings are to be obtained from the asterisk
+    channel.
+  * Made misdn.conf parameter callerid accept the "name" <number> format
+    used by the rest of the system.
+  * Made use the nationalprefix and internationalprefix misdn.conf
+    parameters to prefix any received number from the ISDN link if that
+    number has the corresponding Type-Of-Number.
+  * Added the following new parameters: unknownprefix, netspecificprefix,
+    subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
+    received number from the ISDN link if that number has the corresponding
+    Type-Of-Number.
+
+
+SIP channel driver (chan_sip) changes
+-------------------------------------------
+  * The sendrpid parameter has been expanded to include the options
+    'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
+    header to be sent (equivalent to setting sendrpid=yes) and setting
+    sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
+
 Asterisk Manager Interface
 --------------------------
  * The Hangup action now accepts a Cause header which may be used to