Update API versions and UPGRADE/CHANGES for 12.2.0

This patch does the following:
 * It updates the AMI version to 2.2.0 to indicate backwards compatible
   changes have been made since the last release
 * It updates the ARI version to 1.2.0 to indicate backwards compatible
   changes have been made since the last release
 * It updates the UPGRADE/CHANGES files with changes that were not
   mentioned
........

Merged revisions 411529 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411530 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -133,11 +133,46 @@ AMI
    second channel when dialing LOCAL, or defaults to appending ;2 if only
    the single Id is given.
 
+RealTime
+------------------
+ * A new set of Alembic scripts has been added for CDR tables. This will create
+   a 'cdr' table with the default schema that Asterisk expects.
+
 res_pjsip
 ------------------
  * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
    be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
 
+ * Added the following new CLI commands:
+   - "pjsip show contacts" - list all current PJSIP contacts.
+   - "pjsip show contact" - show specific information about a current PJSIP
+     contact.
+   - "pjsip show channel" - show detailed information about a PJSIP channel.
+
+res_pjsip_multihomed
+------------------
+ * A new module, res_pjsip_multihomed handles situations where the system
+   Asterisk is running out has multiple interfaces. res_pjsip_multihomed
+   determines which interface should be used during message sending.
+
+res_pjsip_pidf_digium_body_supplement
+------------------
+ * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
+   request body formatting for presence support in Digium phones.
+
+res_pjsip_send_to_voicemail
+------------------
+ * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
+   particular headers to transfer a PJSIP channel directly to a particular
+   extension that has VoiceMail. This is intended to be used with Digium
+   phones that support this feature.
+
+res_pjsip_outbound_registration
+------------------
+ * A new CLI command has been added: "pjsip show registrations", which lists
+   all configured PJSIP registrations
+
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
 ------------------------------------------------------------------------------

UPGRADE.txt

@@ -20,37 +20,6 @@
 === UPGRADE-11.txt  -- Upgrade info for 10 to 11
 === UPGRADE-12.txt  -- Upgrade info for 11 to 12
 ===========================================================
-From 12.1.0 to 12.2.0:
-PJSIP:
- - The PJSIP registrar now stores the contents of the User-Agent header of incoming
-   REGISTER requests for each contact that is registered. If using realtime for
-   PJSIP contacts, this means that the schema has been updated to add a user_agent
-   column. An alembic revision has been added to facilitate this update.
- 
- - PJSIP endpoints now have a "message_context" option that can be used to determine
-   where to route incoming MESSAGE requests from the endpoint.
-
-IAX2:
- - When communicating with a peer on an Asterisk 1.4 or earlier system, the
-   chan_iax2 parameter 'connectedline' must be set to "no" in iax.conf. This
-   prevents an incompatible connected line frame from an Astersik 1.8 or later
-   system from causing a hangup in an Asterisk 1.4 or earlier system. Note that
-   this particular incompatibility has always existed between 1.4 and 1.8 and
-   later versions; this upgrade note is simply informing users of its existance.
-
-Realtime Configuration:
- - PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from yes/no
-   enumerators to string values. 'cos_audio' and 'cos_video' have been changed from
-   yes/no enumerators to integer values. PJSIP transport column 'tos' has been
-   changed from a yes/no enumerator to a string value. 'cos' has been changed from
-   a yes/no enumerator to an integer value.
-
-From 12.0.0 to 12.1.0:
-* The sound_place_into_conference sound used in Confbridge is now deprecated
-  and is no longer functional since it has been broken since its inception
-  and the fix involved using a different method to achieve the same goal. The
-  new method to achieve this functionality is by using sound_begin to play
-  a sound to the conference when waitmarked users are moved into the conference.
 
 From 12 to 13:
 
@@ -84,6 +53,18 @@ ARI:
    as channel variables. Other parameters in the JSON body are treated as
    query parameters of the same name.
 
+ - A bug fix in bridge creation has caused a behavioural change in how
+   subscriptions are created for bridges. A bridge created through ARI, does
+   not, by itself, have a subscription created for any particular Stasis
+   application. When a channel in a Stasis application joins a bridge, an
+   implicit event subscription is created for that bridge as well. Previously,
+   when a channel left such a bridge, the subscription was leaked; this allowed
+   for later bridge events to continue to be pushed to the subscribed
+   applications. That leak has been fixed; as a result, bridge events that were
+   delivered after a channel left the bridge are no longer delivered. An
+   application must subscribe to a bridge through the applications resource if
+   it wishes to receive all events related to a bridge.
+
 AMI:
  - The AMI version has been changed from 2.0.0 to 2.1.0. This is to reflect
    the backwards compatible changes listed below.
@@ -125,6 +106,14 @@ CLI commands:
    logging levels since verbose logging levels were made per console.  That
    syntax is now removed and a silence option added in its place.
 
+ConfBridge:
+- The sound_place_into_conference sound used in Confbridge is now deprecated
+  and is no longer functional since it has been broken since its inception
+  and the fix involved using a different method to achieve the same goal. The
+  new method to achieve this functionality is by using sound_begin to play
+  a sound to the conference when waitmarked users are moved into the conference.
+
+
 Configuration Files:
  - The 'verbose' setting in logger.conf still takes an optional argument,
    specifying the verbosity level for each logging destination.  However,
@@ -159,15 +148,28 @@ Realtime Configuration:
    'maximum_expiration', 'outbound_proxy', and 'support_path'.
 
  - The following columns were added to the 'ps_contacts' realtime table:
-   'outbound_proxy' and 'path'.
+   'outbound_proxy', 'user_agent', and 'path'.
 
  - New columns have been added to the ps_endpoints realtime table for the
    'media_address', 'redirect_method' and 'set_var' options.  Also the
-   'mwi_fromuser' column was renamed to 'mwi_from_user'.
+   'mwi_fromuser' column was renamed to 'mwi_from_user'. A new column
+   'message_context' was added to let users configure how MESSAGE requests are
+   routed to the dialplan.
 
  - A new column was added to the 'ps_globals' realtime table for the 'debug'
    option.
 
+ - PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from
+   yes/no enumerators to string values. 'cos_audio' and 'cos_video' have been
+   changed from yes/no enumerators to integer values. PJSIP transport column
+   'tos' has been changed from a yes/no enumerator to a string value. 'cos' has
+   been changed from a yes/no enumerator to an integer value.
+
+ - The 'queues' and 'queue_members' realtime tables have been added to the
+   config Alembic scripts.
+
+ - A new set of Alembic scripts has been added for CDR tables. This will create
+   a 'cdr' table with the default schema that Asterisk expects.
 
 ===========================================================
 ===========================================================

include/asterisk/manager.h

@@ -54,7 +54,7 @@
 - \ref manager.c Main manager code file
  */
 
-#define AMI_VERSION                     "2.1.0"
+#define AMI_VERSION                     "2.2.0"
 #define DEFAULT_MANAGER_PORT 5038	/* Default port for Asterisk management via TCP */
 #define DEFAULT_MANAGER_TLS_PORT 5039	/* Default port for Asterisk management via TCP */
 

rest-api/api-docs/applications.json

@@ -2,7 +2,7 @@
 	"_copyright": "Copyright (C) 2013, Digium, Inc.",
 	"_author": "David M. Lee, II <dlee@digium.com>",
 	"_svn_revision": "$Revision$",
-	"apiVersion": "1.1.0",
+	"apiVersion": "1.2.0",
 	"swaggerVersion": "1.1",
 	"basePath": "http://localhost:8088/stasis",
 	"resourcePath": "/api-docs/applications.{format}",

rest-api/api-docs/asterisk.json

@@ -2,7 +2,7 @@
 	"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
 	"_author": "David M. Lee, II <dlee@digium.com>",
 	"_svn_revision": "$Revision$",
-	"apiVersion": "1.1.0",
+	"apiVersion": "1.2.0",
 	"swaggerVersion": "1.1",
 	"basePath": "http://localhost:8088/stasis",
 	"resourcePath": "/api-docs/asterisk.{format}",

rest-api/api-docs/bridges.json

@@ -2,7 +2,7 @@
 	"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
 	"_author": "David M. Lee, II <dlee@digium.com>",
 	"_svn_revision": "$Revision$",
-	"apiVersion": "1.1.0",
+	"apiVersion": "1.2.0",
 	"swaggerVersion": "1.1",
 	"basePath": "http://localhost:8088/stasis",
 	"resourcePath": "/api-docs/bridges.{format}",

rest-api/api-docs/channels.json

@@ -2,7 +2,7 @@
 	"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
 	"_author": "David M. Lee, II <dlee@digium.com>",
 	"_svn_revision": "$Revision$",
-	"apiVersion": "1.1.0",
+	"apiVersion": "1.2.0",
 	"swaggerVersion": "1.1",
 	"basePath": "http://localhost:8088/stasis",
 	"resourcePath": "/api-docs/channels.{format}",

rest-api/api-docs/deviceStates.json

@@ -2,7 +2,7 @@
 	"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
 	"_author": "Kevin Harwell <kharwell@digium.com>",
 	"_svn_revision": "$Revision$",
-	"apiVersion": "1.1.0",
+	"apiVersion": "1.2.0",
 	"swaggerVersion": "1.1",
 	"basePath": "http://localhost:8088/stasis",
 	"resourcePath": "/api-docs/deviceStates.{format}",

rest-api/api-docs/endpoints.json

@@ -2,7 +2,7 @@
 	"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
 	"_author": "David M. Lee, II <dlee@digium.com>",
 	"_svn_revision": "$Revision$",
-	"apiVersion": "1.1.0",
+	"apiVersion": "1.2.0",
 	"swaggerVersion": "1.1",
 	"basePath": "http://localhost:8088/stasis",
 	"resourcePath": "/api-docs/endpoints.{format}",

rest-api/api-docs/events.json

@@ -2,7 +2,7 @@
 	"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
 	"_author": "David M. Lee, II <dlee@digium.com>",
 	"_svn_revision": "$Revision$",
-	"apiVersion": "1.1.0",
+	"apiVersion": "1.2.0",
 	"swaggerVersion": "1.2",
 	"basePath": "http://localhost:8088/stasis",
 	"resourcePath": "/api-docs/events.{format}",

rest-api/api-docs/mailboxes.json

@@ -2,7 +2,7 @@
 	"_copyright": "Copyright (C) 2013, Digium, Inc.",
 	"_author": "Jonathan Rose <jrose@digium.com>",
 	"_svn_revision": "$Revision$",
-	"apiVersion": "1.1.0",
+	"apiVersion": "1.2.0",
 	"swaggerVersion": "1.1",
 	"basePath": "http://localhost:8088/stasis",
 	"resourcePath": "/api-docs/mailboxes.{format}",

rest-api/api-docs/playbacks.json

@@ -2,7 +2,7 @@
 	"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
 	"_author": "David M. Lee, II <dlee@digium.com>",
 	"_svn_revision": "$Revision$",
-	"apiVersion": "1.1.0",
+	"apiVersion": "1.2.0",
 	"swaggerVersion": "1.1",
 	"basePath": "http://localhost:8088/stasis",
 	"resourcePath": "/api-docs/playbacks.{format}",

rest-api/api-docs/recordings.json

@@ -2,7 +2,7 @@
 	"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
 	"_author": "David M. Lee, II <dlee@digium.com>",
 	"_svn_revision": "$Revision$",
-	"apiVersion": "1.1.0",
+	"apiVersion": "1.2.0",
 	"swaggerVersion": "1.1",
 	"basePath": "http://localhost:8088/stasis",
 	"resourcePath": "/api-docs/recordings.{format}",

rest-api/api-docs/sounds.json

@@ -2,7 +2,7 @@
 	"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
 	"_author": "David M. Lee, II <dlee@digium.com>",
 	"_svn_revision": "$Revision$",
-	"apiVersion": "1.1.0",
+	"apiVersion": "1.2.0",
 	"swaggerVersion": "1.1",
 	"basePath": "http://localhost:8088/stasis",
 	"resourcePath": "/api-docs/sounds.{format}",

rest-api/resources.json

@@ -2,7 +2,7 @@
 	"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
 	"_author": "David M. Lee, II <dlee@digium.com>",
 	"_svn_revision": "$Revision$",
-	"apiVersion": "1.1.0",
+	"apiVersion": "1.2.0",
 	"swaggerVersion": "1.1",
 	"basePath": "http://localhost:8088/ari",
 	"apis": [