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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
This commit is contained in:
Matthew Jordan
35
CHANGES
35
CHANGES
@@ -133,11 +133,46 @@ AMI
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second channel when dialing LOCAL, or defaults to appending ;2 if only
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the single Id is given.
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RealTime
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------------------
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* A new set of Alembic scripts has been added for CDR tables. This will create
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a 'cdr' table with the default schema that Asterisk expects.
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res_pjsip
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------------------
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* transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
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be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
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* Added the following new CLI commands:
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- "pjsip show contacts" - list all current PJSIP contacts.
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- "pjsip show contact" - show specific information about a current PJSIP
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contact.
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- "pjsip show channel" - show detailed information about a PJSIP channel.
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res_pjsip_multihomed
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------------------
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* A new module, res_pjsip_multihomed handles situations where the system
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Asterisk is running out has multiple interfaces. res_pjsip_multihomed
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determines which interface should be used during message sending.
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res_pjsip_pidf_digium_body_supplement
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------------------
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* A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
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request body formatting for presence support in Digium phones.
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res_pjsip_send_to_voicemail
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------------------
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* A new module, res_pjsip_send_to_voicemail allows for REFER requests with
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particular headers to transfer a PJSIP channel directly to a particular
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extension that has VoiceMail. This is intended to be used with Digium
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phones that support this feature.
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res_pjsip_outbound_registration
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------------------
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* A new CLI command has been added: "pjsip show registrations", which lists
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all configured PJSIP registrations
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------------------------------------------------------------------------------
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--- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
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------------------------------------------------------------------------------
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68
UPGRADE.txt
68
UPGRADE.txt
@@ -20,37 +20,6 @@
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=== UPGRADE-11.txt -- Upgrade info for 10 to 11
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=== UPGRADE-12.txt -- Upgrade info for 11 to 12
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===========================================================
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From 12.1.0 to 12.2.0:
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PJSIP:
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- The PJSIP registrar now stores the contents of the User-Agent header of incoming
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REGISTER requests for each contact that is registered. If using realtime for
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PJSIP contacts, this means that the schema has been updated to add a user_agent
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column. An alembic revision has been added to facilitate this update.
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- PJSIP endpoints now have a "message_context" option that can be used to determine
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where to route incoming MESSAGE requests from the endpoint.
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IAX2:
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- When communicating with a peer on an Asterisk 1.4 or earlier system, the
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chan_iax2 parameter 'connectedline' must be set to "no" in iax.conf. This
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prevents an incompatible connected line frame from an Astersik 1.8 or later
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system from causing a hangup in an Asterisk 1.4 or earlier system. Note that
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this particular incompatibility has always existed between 1.4 and 1.8 and
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later versions; this upgrade note is simply informing users of its existance.
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Realtime Configuration:
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- PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from yes/no
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enumerators to string values. 'cos_audio' and 'cos_video' have been changed from
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yes/no enumerators to integer values. PJSIP transport column 'tos' has been
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changed from a yes/no enumerator to a string value. 'cos' has been changed from
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a yes/no enumerator to an integer value.
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From 12.0.0 to 12.1.0:
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* The sound_place_into_conference sound used in Confbridge is now deprecated
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and is no longer functional since it has been broken since its inception
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and the fix involved using a different method to achieve the same goal. The
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new method to achieve this functionality is by using sound_begin to play
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a sound to the conference when waitmarked users are moved into the conference.
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From 12 to 13:
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@@ -84,6 +53,18 @@ ARI:
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as channel variables. Other parameters in the JSON body are treated as
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query parameters of the same name.
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- A bug fix in bridge creation has caused a behavioural change in how
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subscriptions are created for bridges. A bridge created through ARI, does
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not, by itself, have a subscription created for any particular Stasis
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application. When a channel in a Stasis application joins a bridge, an
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implicit event subscription is created for that bridge as well. Previously,
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when a channel left such a bridge, the subscription was leaked; this allowed
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for later bridge events to continue to be pushed to the subscribed
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applications. That leak has been fixed; as a result, bridge events that were
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delivered after a channel left the bridge are no longer delivered. An
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application must subscribe to a bridge through the applications resource if
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it wishes to receive all events related to a bridge.
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AMI:
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- The AMI version has been changed from 2.0.0 to 2.1.0. This is to reflect
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the backwards compatible changes listed below.
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@@ -125,6 +106,14 @@ CLI commands:
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logging levels since verbose logging levels were made per console. That
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syntax is now removed and a silence option added in its place.
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ConfBridge:
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- The sound_place_into_conference sound used in Confbridge is now deprecated
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and is no longer functional since it has been broken since its inception
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and the fix involved using a different method to achieve the same goal. The
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new method to achieve this functionality is by using sound_begin to play
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a sound to the conference when waitmarked users are moved into the conference.
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Configuration Files:
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- The 'verbose' setting in logger.conf still takes an optional argument,
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specifying the verbosity level for each logging destination. However,
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@@ -159,15 +148,28 @@ Realtime Configuration:
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'maximum_expiration', 'outbound_proxy', and 'support_path'.
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- The following columns were added to the 'ps_contacts' realtime table:
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'outbound_proxy' and 'path'.
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'outbound_proxy', 'user_agent', and 'path'.
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- New columns have been added to the ps_endpoints realtime table for the
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'media_address', 'redirect_method' and 'set_var' options. Also the
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'mwi_fromuser' column was renamed to 'mwi_from_user'.
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'mwi_fromuser' column was renamed to 'mwi_from_user'. A new column
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'message_context' was added to let users configure how MESSAGE requests are
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routed to the dialplan.
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- A new column was added to the 'ps_globals' realtime table for the 'debug'
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option.
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- PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from
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yes/no enumerators to string values. 'cos_audio' and 'cos_video' have been
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changed from yes/no enumerators to integer values. PJSIP transport column
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'tos' has been changed from a yes/no enumerator to a string value. 'cos' has
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been changed from a yes/no enumerator to an integer value.
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- The 'queues' and 'queue_members' realtime tables have been added to the
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config Alembic scripts.
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- A new set of Alembic scripts has been added for CDR tables. This will create
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a 'cdr' table with the default schema that Asterisk expects.
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===========================================================
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===========================================================
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@@ -54,7 +54,7 @@
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- \ref manager.c Main manager code file
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*/
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#define AMI_VERSION "2.1.0"
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#define AMI_VERSION "2.2.0"
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#define DEFAULT_MANAGER_PORT 5038 /* Default port for Asterisk management via TCP */
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#define DEFAULT_MANAGER_TLS_PORT 5039 /* Default port for Asterisk management via TCP */
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@@ -2,7 +2,7 @@
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"_copyright": "Copyright (C) 2013, Digium, Inc.",
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"_author": "David M. Lee, II <dlee@digium.com>",
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"_svn_revision": "$Revision$",
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"apiVersion": "1.1.0",
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"apiVersion": "1.2.0",
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"swaggerVersion": "1.1",
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"basePath": "http://localhost:8088/stasis",
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"resourcePath": "/api-docs/applications.{format}",
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@@ -2,7 +2,7 @@
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"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
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"_author": "David M. Lee, II <dlee@digium.com>",
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"_svn_revision": "$Revision$",
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"apiVersion": "1.1.0",
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"apiVersion": "1.2.0",
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"swaggerVersion": "1.1",
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"basePath": "http://localhost:8088/stasis",
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"resourcePath": "/api-docs/asterisk.{format}",
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@@ -2,7 +2,7 @@
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"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
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"_author": "David M. Lee, II <dlee@digium.com>",
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"_svn_revision": "$Revision$",
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"apiVersion": "1.1.0",
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"apiVersion": "1.2.0",
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"swaggerVersion": "1.1",
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"basePath": "http://localhost:8088/stasis",
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"resourcePath": "/api-docs/bridges.{format}",
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@@ -2,7 +2,7 @@
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"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
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"_author": "David M. Lee, II <dlee@digium.com>",
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"_svn_revision": "$Revision$",
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"apiVersion": "1.1.0",
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"apiVersion": "1.2.0",
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"swaggerVersion": "1.1",
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"basePath": "http://localhost:8088/stasis",
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"resourcePath": "/api-docs/channels.{format}",
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@@ -2,7 +2,7 @@
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"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
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"_author": "Kevin Harwell <kharwell@digium.com>",
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"_svn_revision": "$Revision$",
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"apiVersion": "1.1.0",
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"apiVersion": "1.2.0",
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"swaggerVersion": "1.1",
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"basePath": "http://localhost:8088/stasis",
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"resourcePath": "/api-docs/deviceStates.{format}",
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@@ -2,7 +2,7 @@
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"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
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"_author": "David M. Lee, II <dlee@digium.com>",
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"_svn_revision": "$Revision$",
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"apiVersion": "1.1.0",
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"apiVersion": "1.2.0",
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"swaggerVersion": "1.1",
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"basePath": "http://localhost:8088/stasis",
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"resourcePath": "/api-docs/endpoints.{format}",
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@@ -2,7 +2,7 @@
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"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
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"_author": "David M. Lee, II <dlee@digium.com>",
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"_svn_revision": "$Revision$",
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"apiVersion": "1.1.0",
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"apiVersion": "1.2.0",
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"swaggerVersion": "1.2",
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"basePath": "http://localhost:8088/stasis",
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"resourcePath": "/api-docs/events.{format}",
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@@ -2,7 +2,7 @@
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"_copyright": "Copyright (C) 2013, Digium, Inc.",
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"_author": "Jonathan Rose <jrose@digium.com>",
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"_svn_revision": "$Revision$",
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"apiVersion": "1.1.0",
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"apiVersion": "1.2.0",
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"swaggerVersion": "1.1",
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"basePath": "http://localhost:8088/stasis",
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"resourcePath": "/api-docs/mailboxes.{format}",
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@@ -2,7 +2,7 @@
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"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
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"_author": "David M. Lee, II <dlee@digium.com>",
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"_svn_revision": "$Revision$",
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"apiVersion": "1.1.0",
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"apiVersion": "1.2.0",
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"swaggerVersion": "1.1",
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"basePath": "http://localhost:8088/stasis",
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"resourcePath": "/api-docs/playbacks.{format}",
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@@ -2,7 +2,7 @@
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"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
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"_author": "David M. Lee, II <dlee@digium.com>",
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"_svn_revision": "$Revision$",
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"apiVersion": "1.1.0",
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"apiVersion": "1.2.0",
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"swaggerVersion": "1.1",
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"basePath": "http://localhost:8088/stasis",
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"resourcePath": "/api-docs/recordings.{format}",
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@@ -2,7 +2,7 @@
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"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
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"_author": "David M. Lee, II <dlee@digium.com>",
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"_svn_revision": "$Revision$",
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"apiVersion": "1.1.0",
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"apiVersion": "1.2.0",
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"swaggerVersion": "1.1",
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"basePath": "http://localhost:8088/stasis",
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"resourcePath": "/api-docs/sounds.{format}",
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@@ -2,7 +2,7 @@
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"_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.",
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"_author": "David M. Lee, II <dlee@digium.com>",
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"_svn_revision": "$Revision$",
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"apiVersion": "1.1.0",
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"apiVersion": "1.2.0",
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"swaggerVersion": "1.1",
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"basePath": "http://localhost:8088/ari",
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"apis": [
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