diff options
author | Matthew Jordan <mjordan@digium.com> | 2014-03-28 17:41:23 +0000 |
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committer | Matthew Jordan <mjordan@digium.com> | 2014-03-28 17:41:23 +0000 |
commit | 597f25db69c71d342e8661983df47e457625b5e2 (patch) | |
tree | d39f3071d46b4f2c3cf8082e4b0c2503747b47e2 /UPGRADE.txt | |
parent | a438a0e65fa183860ed04eb1487ecd991db57225 (diff) |
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
Diffstat (limited to 'UPGRADE.txt')
-rw-r--r-- | UPGRADE.txt | 68 |
1 files changed, 35 insertions, 33 deletions
diff --git a/UPGRADE.txt b/UPGRADE.txt index b139fec5c..41f21f2ec 100644 --- a/UPGRADE.txt +++ b/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. =========================================================== =========================================================== |