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-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:
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.
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,
'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.
===========================================================
===========================================================
- \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 */
"_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}",
"_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}",
"_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}",
"_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}",
"_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}",
"_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}",
"_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}",
"_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}",
"_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}",
"_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}",
"_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}",
"_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": [
projects / thirdparty / asterisk.git / commitdiff
