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Update API versions and UPGRADE/CHANGES for 12.2.0

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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


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@411529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Matthew Jordan
2014-03-28 17:35:48 +00:00
parent 44e73556bc
commit c6e15a8e34
15 changed files with 96 additions and 29 deletions
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36
CHANGES
View File

@@ -21,6 +21,7 @@ ARI
------------------
* The live recording object on recording events now contains a target_uri
field which contains the URI of what is being recorded.
* The bridge type used when creating a bridge is now a comma separated list of
bridge properties. Valid options are: mixing, holding, dtmf_events, and
proxy_media.
@@ -45,11 +46,46 @@ 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 ------------
------------------------------------------------------------------------------

63
UPGRADE.txt
View File

@@ -21,6 +21,35 @@
===
===========================================================
From 12.1.0 to 12.2.0:
AMI:
- The AMI version has been changed from 2.1.0 to 2.2.0. This is to reflect
the backwards compatible changes listed in the CHANGES file.
ARI:
- The ARI version has been changed from 1.1.0 to 1.2.0. This is to reflect
the backwards compatible changes listed in the CHANGES file.
- 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.
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.
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
@@ -40,27 +69,29 @@ ODBC:
(sipppeers,sipregs)
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.
- 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.
- PJSIP endpoints now have a "message_context" option that can be used to
determine where to route incoming MESSAGE requests from the endpoint.
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.
- 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.
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.
- The per console verbose level feature as previously implemented caused a
large performance penalty. The fix required some minor incompatibilities

2
include/asterisk/manager.h
View File

@@ -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 */

2
rest-api/api-docs/applications.json
View File

@@ -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}",

2
rest-api/api-docs/asterisk.json
View File

@@ -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}",

2
rest-api/api-docs/bridges.json
View File

@@ -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}",

2
rest-api/api-docs/channels.json
View File

@@ -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}",

2
rest-api/api-docs/deviceStates.json
View File

@@ -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}",

2
rest-api/api-docs/endpoints.json
View File

@@ -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}",

2
rest-api/api-docs/events.json
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@@ -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}",

2
rest-api/api-docs/mailboxes.json
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@@ -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}",

2
rest-api/api-docs/playbacks.json
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@@ -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}",

2
rest-api/api-docs/recordings.json
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@@ -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}",

2
rest-api/api-docs/sounds.json
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@@ -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}",

2
rest-api/resources.json
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@@ -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": [