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Update UPGRADE-13.txt file

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
changes/42/42/1
Matthew Jordan 11 years ago
parent 3e452fa4d9
commit 455243cdd4
1 changed files with 150 additions and 140 deletions
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UPGRADE-13.txt
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@ -21,15 +21,7 @@
=== UPGRADE-12.txt -- Upgrade info for 11 to 12
===========================================================
From 12 to 13:
- Sample config files have been moved from configs/ to a subfolder of that
directory, 'samples'.
- The menuselect utility has been pulled into the Asterisk repository. As a
result, the libxml2 development library is now a required dependency for
Asterisk.
General Asterisk Changes:
- The asterisk command line -I option and the asterisk.conf internal_timing
option are removed and always enabled if any timing module is loaded.
@ -42,14 +34,6 @@ From 12 to 13:
an older version of rasterisk connects to the new version then the
"core set verbose" command will have no effect.
- Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
objects will emit additional debug information to the refs log file located
in the standard Asterisk log file directory. This log file is useful in
tracking down object leaks and other reference counting issues. Prior to
this version, this option was only available by modifying the source code
directly. This change also includes a new script, refcounter.py, in the
contrib folder that will process the refs log file.
- The asterisk compatibility options in asterisk.conf have been removed.
These options enabled certain backwards compatibility features for
pbx_realtime, res_agi, and app_set that made their behaviour similar to
@ -57,99 +41,47 @@ From 12 to 13:
update their dialplans to use ',' instead of '|' as a delimiter, and should
use the Set dialplan application instead of the MSet dialplan application.
accountcode:
- Accountcode behavior changed somewhat to add functional peeraccount
support. The main change is that local channels now cross accountcode
and peeraccount across the special bridge between the ;1 and ;2 channels
just like channels between normal bridges. See the CHANGES file for
more information.
ARI:
- The ARI version has been changed from 1.0.0 to 1.1.0. This is to reflect
the backwards compatible changes listed below.
- Added a new ARI resource 'mailboxes' which allows the creation and
modification of mailboxes managed by external MWI. Modules res_mwi_external
and res_stasis_mailbox must be enabled to use this resource.
- Added new events for externally initiated transfers. The event
BridgeBlindTransfer is now raised when a channel initiates a blind transfer
of a bridge in the ARI controlled application to the dialplan; the
BridgeAttendedTransfer event is raised when a channel initiates an
attended transfer of a bridge in the ARI controlled application to the
dialplan.
- Channel variables may now be specified as a body parameter to the
POST /channels operation. The 'variables' key in the JSON is interpreted
as a sequence of key/value pairs that will be added to the created channel
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.
- The DialStatus field in the DialEnd event can now have additional values.
This includes ABORT, CONTINUE, and GOTO.
- The res_mwi_external_ami module can, if loaded, provide additional AMI
actions and events that convey MWI state within Asterisk. This includes
the MWIGet, MWIUpdate, and MWIDelete actions, as well as the MWIGet and
MWIGetComplete events that occur in response to an MWIGet action.
Build System:
- Sample config files have been moved from configs/ to a subfolder of that
directory, 'samples'.
- AMI now contains a new class authorization, 'security'. This is used with
the following new events: FailedACL, InvalidAccountID, SessionLimit,
MemoryLimit, LoadAverageLimit, RequestNotAllowed, AuthMethodNotAllowed,
RequestBadFormat, SuccessfulAuth, UnexpectedAddress, ChallengeResponseFailed,
InvalidPassword, ChallengeSent, and InvalidTransport.
- The menuselect utility has been pulled into the Asterisk repository. As a
result, the libxml2 development library is now a required dependency for
Asterisk.
- Bridge related events now have two additional fields: BridgeName and
BridgeCreator. BridgeName is a descriptive name for the bridge;
BridgeCreator is the name of the entity that created the bridge. This
affects the following events: ConfbridgeStart, ConfbridgeEnd,
ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
- Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
objects will emit additional debug information to the refs log file located
in the standard Asterisk log file directory. This log file is useful in
tracking down object leaks and other reference counting issues. Prior to
this version, this option was only available by modifying the source code
directly. This change also includes a new script, refcounter.py, in the
contrib folder that will process the refs log file.
- MixMonitor AMI actions now require users to have authorization classes.
* MixMonitor - system
* MixMonitorMute - call or system
* StopMixMonitor - call or system
Applications:
- Removed the undocumented manager.conf block-sockets option. It interferes with
TCP/TLS inactivity timeouts.
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.
- The response to the PresenceState AMI action has historically contained two
Message keys. The first of these is used as an informative message regarding
the success/failure of the action; the second contains a Presence state
specific message. Having two keys with the same unique name in an AMI
message is cumbersome for some client; hence, the Presence specific Message
has been deprecated. The message will now contain a PresenceMessage key
for the presence specific information; the Message key containing presence
information will be removed in the next major version of AMI.
SetMusicOnHold:
- The SetMusicOnHold dialplan application was deprecated and has been removed.
Users of the application should use the CHANNEL function's musicclass
setting instead.
CDRs:
- The "endbeforehexten" setting now defaults to "yes", instead of "no".
When set to "no", yhis setting will cause a new CDR to be generated when a
channel enters into hangup logic (either the 'h' extension or a hangup
handler subroutine). In general, this is not the preferred default: this
causes extra CDRs to be generated for a channel in many common dialplans.
WaitMusicOnHold:
- The WaitMusicOnHold dialplan application was deprecated and has been
removed. Users of the application should use MusicOnHold with a duration
parameter instead.
CDR Backends:
- The cdr_sqlite module was deprecated and has been removed. Users of this
module should use the cdr_sqlite3_custom module instead.
Channel Drivers:
chan_dahdi:
- SS7 support now requires libss7 v2.0 or later.
@ -241,6 +173,108 @@ chan_sip:
(3) All other codecs that are not the preferred codec and not a joint
codec offered by the inbound offer
chan_unistim:
- The unistim.conf 'dateformat' has changed meaning of options values to conform
values used inside Unistim protocol
- Added 'dtmf_duration' option with changing default operation to disable
receivied dtmf playback on unistim phone
Core:
Account Codes:
- accountcode behavior changed somewhat to add functional peeraccount
support. The main change is that local channels now cross accountcode
and peeraccount across the special bridge between the ;1 and ;2 channels
just like channels between normal bridges. See the CHANGES file for
more information.
ARI:
- The ARI version has been changed to 1.5.0. This is to reflect backwards
compatible changes made since 12.0.0 was released.
- Added a new ARI resource 'mailboxes' which allows the creation and
modification of mailboxes managed by external MWI. Modules res_mwi_external
and res_stasis_mailbox must be enabled to use this resource.
- Added new events for externally initiated transfers. The event
BridgeBlindTransfer is now raised when a channel initiates a blind transfer
of a bridge in the ARI controlled application to the dialplan; the
BridgeAttendedTransfer event is raised when a channel initiates an
attended transfer of a bridge in the ARI controlled application to the
dialplan.
- Channel variables may now be specified as a body parameter to the
POST /channels operation. The 'variables' key in the JSON is interpreted
as a sequence of key/value pairs that will be added to the created channel
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 to 2.5.0. This is to reflect backwards
compatible changes made since 12.0.0 was released.
- The DialStatus field in the DialEnd event can now have additional values.
This includes ABORT, CONTINUE, and GOTO.
- The res_mwi_external_ami module can, if loaded, provide additional AMI
actions and events that convey MWI state within Asterisk. This includes
the MWIGet, MWIUpdate, and MWIDelete actions, as well as the MWIGet and
MWIGetComplete events that occur in response to an MWIGet action.
- AMI now contains a new class authorization, 'security'. This is used with
the following new events: FailedACL, InvalidAccountID, SessionLimit,
MemoryLimit, LoadAverageLimit, RequestNotAllowed, AuthMethodNotAllowed,
RequestBadFormat, SuccessfulAuth, UnexpectedAddress, ChallengeResponseFailed,
InvalidPassword, ChallengeSent, and InvalidTransport.
- Bridge related events now have two additional fields: BridgeName and
BridgeCreator. BridgeName is a descriptive name for the bridge;
BridgeCreator is the name of the entity that created the bridge. This
affects the following events: ConfbridgeStart, ConfbridgeEnd,
ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
- MixMonitor AMI actions now require users to have authorization classes.
* MixMonitor - system
* MixMonitorMute - call or system
* StopMixMonitor - call or system
- Removed the undocumented manager.conf block-sockets option. It interferes with
TCP/TLS inactivity timeouts.
- The response to the PresenceState AMI action has historically contained two
Message keys. The first of these is used as an informative message regarding
the success/failure of the action; the second contains a Presence state
specific message. Having two keys with the same unique name in an AMI
message is cumbersome for some client; hence, the Presence specific Message
has been deprecated. The message will now contain a PresenceMessage key
for the presence specific information; the Message key containing presence
information will be removed in the next major version of AMI.
- The manager.conf 'eventfilter' now takes an "extended" regular expression
instead of a "basic" one.
CDRs:
- The "endbeforehexten" setting now defaults to "yes", instead of "no".
When set to "no", yhis setting will cause a new CDR to be generated when a
channel enters into hangup logic (either the 'h' extension or a hangup
handler subroutine). In general, this is not the preferred default: this
causes extra CDRs to be generated for a channel in many common dialplans.
CLI commands:
- "core show settings" now lists the current console verbosity in addition
to the root console verbosity.
@ -249,15 +283,7 @@ 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:
Logging:
- The 'verbose' setting in logger.conf still takes an optional argument,
specifying the verbosity level for each logging destination. However,
the default is now to once again follow the current root console level.
@ -265,12 +291,6 @@ Configuration Files:
again set the root console verbose level and affect the verbose level
logged.
- The manager.conf 'eventfilter' now takes an "extended" regular expression
instead of a "basic" one.
- The unistim.conf 'dateformat' has changed meaning of options values to conform
values used inside Unistim protocol
HTTP:
- Added http.conf session_inactivity timer option to close HTTP connections
that aren't doing anything.
@ -280,19 +300,6 @@ HTTP:
keep alive time between HTTP requests is configured in http.conf with the
session_keep_alive parameter.
MusicOnHold
- The SetMusicOnHold dialplan application was deprecated and has been removed.
Users of the application should use the CHANNEL function's musicclass
setting instead.
- The WaitMusicOnHold dialplan application was deprecated and has been
removed. Users of the application should use MusicOnHold with a duration
parameter instead.
ODBC:
- The compatibility setting, allow_empty_string_in_nontext, has been removed.
Empty column values will be stored as empty strings during realtime updates.
Realtime Configuration:
- WARNING: The database migration script that adds the 'extensions' table for
realtime had to be modified due to an error when installing for MySQL. The
@ -335,10 +342,25 @@ Realtime Configuration:
- 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.
Resources:
res_odbc:
- The compatibility setting, allow_empty_string_in_nontext, has been removed.
Empty column values will be stored as empty strings during realtime updates.
res_jabber:
- This module was deprecated and has been removed. Users of this module should
use res_xmpp instead.
res_http_websocket:
- Added a compatibility option to ari.conf, sip.conf, and pjsip.conf
'websocket_write_timeout'. When a websocket connection exists where Asterisk
writes a substantial amount of data to the connected client, and the connected
client is slow to process the received data, the socket may be disconnected.
In such cases, it may be necessary to adjust this value.
Default is 100 ms.
Scripts:
safe_asterisk:
- The safe_asterisk script was previously not installed on top of an existing
version. This caused bug-fixes in that script not to be deployed. If your
@ -352,21 +374,9 @@ safe_asterisk:
- "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
- "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"
Unistim:
- Added 'dtmf_duration' option with changing default operation to disable
receivied dtmf playback on unistim phone
Utilities:
- The refcounter program has been removed in favor of the refcounter.py script
in contrib/scripts.
WebSockets:
- Added a compatibility option to ari.conf, sip.conf, and pjsip.conf
'websocket_write_timeout'. When a websocket connection exists where Asterisk
writes a substantial amount of data to the connected client, and the connected
client is slow to process the received data, the socket may be disconnected.
In such cases, it may be necessary to adjust this value.
Default is 100 ms.
===========================================================
===========================================================

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