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Update CHANGES for Asterisk 11
This updates the CHANGES file with things that were committed for Asterisk 11, but were not noted in that file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Matthew Jordan
763
CHANGES
763
CHANGES
@@ -13,17 +13,7 @@
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------------------------------------------------------------------------------
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------------------------------------------------------------------------------
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||||||
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Build System
|
Build System
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||||||
----
|
-------------------
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||||||
* A new make target, 'full', has been added to the Makefile. This performs
|
|
||||||
the same compilation actions as make all, but will also scan the entirety of
|
|
||||||
each source file for documentation. This option is needed to generate AMI
|
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||||||
event documentation. Note that your system must have Python in order for
|
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||||||
this make target to succeed.
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||||||
|
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||||||
Core
|
|
||||||
----
|
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||||||
* The expression parser now recognizes the ABS() absolute value function,
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||||||
which will convert negative floating point values to positive values.
|
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||||||
* The Asterisk build system will now build and install a shared library
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* The Asterisk build system will now build and install a shared library
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||||||
(libasteriskssl.so) used to wrap various initialization and shutdown functions
|
(libasteriskssl.so) used to wrap various initialization and shutdown functions
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||||||
from the libssl and libcrypto libraries provided by OpenSSL. This is done so
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from the libssl and libcrypto libraries provided by OpenSSL. This is done so
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@@ -31,147 +21,112 @@ Core
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|||||||
modules that are loaded into Asterisk, since they should only be called once
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modules that are loaded into Asterisk, since they should only be called once
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||||||
in any single process. If desired, this feature can be disabled by supplying
|
in any single process. If desired, this feature can be disabled by supplying
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||||||
the "--disable-asteriskssl" option to the configure script.
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the "--disable-asteriskssl" option to the configure script.
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* Threads belonging to a particular call are now linked with callids which get
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added to any log messages produced by those threads. Log messages can now be
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easily identified as involved with a certain call by looking at their call id.
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||||||
Call ids may also be attached to log messages for just about any case where
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||||||
it can be determined to be related to a particular call.
|
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||||||
* The minimum DTMF duration can now be configured in asterisk.conf
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||||||
as "mindtmfduration". The default value is (as before) set to 80 ms.
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(previously it was only available in source code)
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* Each logging destination and console now have an independent notion of the
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current verbosity level. Logger.conf now allows an optional argument to
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||||||
the 'verbose' specifier, indicating the level of verbosity sent to that
|
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particular logging destination. Additionally, remote consoles now each
|
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||||||
have their own verbosity level. The command 'core set verbose' will now set
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a separate level for each remote console without affecting any other
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console.
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* Named ACLs can now be specified in acl.conf and used in configurations that
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||||||
use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
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||||||
used to specify an ACL, a similar form of 'acl' will add a named ACL to the
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||||||
working ACL. In addition, some CLI commands have
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||||||
been added to provide informational and configuration reload capabilities to
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this feature ('acl show [named acl]' and 'reload acl').
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* Hangup handlers can be attached to channels using the CHANNEL(hangup_handler_xxx)
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||||||
options. Hangup handlers will run when the channel is hung up similar to the
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||||||
h extension.
|
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||||||
* The AMI Hangup event now includes the AccountCode header so you can easily
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||||||
correlate with AMI Newchannel events.
|
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||||||
|
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CLI Changes
|
* A new make target, 'full', has been added to the Makefile. This performs
|
||||||
-------------------
|
the same compilation actions as make all, but will also scan the entirety of
|
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* mixmonitor list <channel> command will now show MixMonitor ID, and the filenames
|
each source file for documentation. This option is needed to generate AMI
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||||||
of all running mixmonitors on a channel.
|
event documentation. Note that your system must have Python in order for
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* The debuglevel of "pri set debug" is now a bitmask ranging from 0 to 15 if
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this make target to succeed.
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numeric instead of 0, 1, or 2.
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* "stun show status" will show a table describing how the STUN client is behaving.
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|
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ConfBridge
|
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-------------------
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* Added menu action admin_toggle_mute_participants. This will mute / unmute
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all non-admin participants on a conference. The confbridge configuration file
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also allows for the default sounds played to all conference users when this
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||||||
occurs to be overriden using sound_participants_unmuted and sound_participants_muted.
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* Added menu action participant_count. This will playback the number of current
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||||||
participants in a conference.
|
|
||||||
* Added announcement configuration option to user profile. If set the sound file will
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||||||
be played to the user, and only the user, upon joining the conference bridge.
|
|
||||||
|
|
||||||
Voicemail
|
|
||||||
------------------
|
|
||||||
* Addition of the VM_INFO function - see Dialplan function changes
|
|
||||||
* The imapserver, imapport, and imapflags configuration options can now be
|
|
||||||
overriden on a user by user basis.
|
|
||||||
|
|
||||||
SIP Changes
|
|
||||||
-----------
|
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||||||
* Asterisk will no longer substitute CID number for CID name into display
|
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||||||
name field if CID number exists without a CID name. This change improves
|
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||||||
compatibility with certain device features such as Avaya IP500's directory
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|
||||||
lookup service.
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* A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
|
|
||||||
created using that setting to not be removed during SIP reload.
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||||||
* Add support to realtime for the 'callbackextension' option
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|
||||||
* When multiple peers exist with the same address, but differing
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||||||
callbackextension options, incoming requests that are matched by address
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will be matched to the peer with the matching callbackextension if it is
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available.
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* NAT settings are now a combinable list of options. The equivalent of the
|
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||||||
deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
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* Two new NAT options, auto_force_rport and auto_comedia, have been added
|
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||||||
which set the force_rport and comedia options automatically if Asterisk
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||||||
detects that an incoming SIP request crossed a NAT after being sent by
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the remote endpoint.
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* Adds an option send_diversion which can be disabled to prevent
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diversion headers from automatically being added to invites.
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* Add support for lightweight NAT keepalive. If enabled a blank packet will
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be sent to the remote host at a given interval to keep the NAT mapping open.
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This can be enabled using the keepalive configuration option.
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* Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
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as the transport.
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Chan_local changes
|
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------------------
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* Added a manager event "LocalBridge" for local channel call bridges between
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the two pseudo-channels created.
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Chan_dahdi changes
|
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------------------
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* Added dialtone_detect option for analog ports to disconnect incoming
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calls when dialtone is detected.
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------------------------------------------------------------------------------
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--- Functionality changes since Asterisk 10.4.0 ------------------------------
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------------------------------------------------------------------------------
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||||||
Build System
|
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||||||
------------
|
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||||||
* The optimization portion of the build system has been reworked to avoid
|
* The optimization portion of the build system has been reworked to avoid
|
||||||
broken builds on certain architectures. All architecture-specific
|
broken builds on certain architectures. All architecture-specific
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||||||
optimization has been removed in favor of using -march=native to allow gcc
|
optimization has been removed in favor of using -march=native to allow gcc
|
||||||
to detect the environment in which it is running when possible. This can
|
to detect the environment in which it is running when possible. This can
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||||||
be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
|
be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
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||||||
|
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------------------------------------------------------------------------------
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* BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
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--- Functionality changes since Asterisk 10.3.0 ------------------------------
|
make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
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------------------------------------------------------------------------------
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Chan_unistim changes
|
* Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
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--------------------
|
previously parsed the header file to obtain the version of Asterisk, you
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* Added ability to use multiple lines on phone, so for one device in
|
will now have to go through Asterisk to get the version information.
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||||||
configuration multiple lines can be defined, it allows to have multiple calls
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on one phone, callwaiting and switching between calls.
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||||||
* Added option 'sharpdial' allowing end dialing by pressing # key
|
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||||||
* Added option 'interdigit_timer' for controll phone dial timeout
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* Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
|
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* Added global 'debug' option, that enables debug in channel driver
|
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||||||
* Added ability for translation on-screen menu to multiple languages. Tested on
|
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Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
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ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
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menu of phone
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* In addition to English added French and Russian languages for on-screen menus
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* Reworked dialing number input: added dialing by timeout, immediate dial on
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||||||
on dialplan compare, phone number length now not limited by screen size
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* Added ability for pickup a call using features.conf defined value and
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on-screen key
|
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||||||
|
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Codec changes
|
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-------------
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* Codec lists may now be modified by the '!' character, to allow succinct
|
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specification of a list of codecs allowed and disallowed, without the
|
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||||||
requirement to use two different keywords. For example, to specify all
|
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||||||
codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
|
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||||||
|
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||||||
Music On Hold Changes
|
Applications
|
||||||
---------------------
|
-------------------
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||||||
* Added 'announcement' option which will play at the start of MOH and between
|
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songs in modes of MOH that can detect transitions between songs (eg.
|
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||||||
files, mp3, etc).
|
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||||||
|
|
||||||
Queue changes
|
Bridge
|
||||||
-------------
|
-------------------
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||||||
|
* Added 'F()' option. Similar to the dial option, this can be supplied with
|
||||||
|
arguments indicating where the callee should go after the caller is hung up,
|
||||||
|
or without options specified, the priority after the Queue will be used.
|
||||||
|
|
||||||
|
|
||||||
|
ConfBridge
|
||||||
|
-------------------
|
||||||
|
* Added menu action admin_toggle_mute_participants. This will mute / unmute
|
||||||
|
all non-admin participants on a conference. The confbridge configuration
|
||||||
|
file also allows for the default sounds played to all conference users when
|
||||||
|
this occurs to be overriden using sound_participants_unmuted and
|
||||||
|
sound_participants_muted.
|
||||||
|
|
||||||
|
* Added menu action participant_count. This will playback the number of
|
||||||
|
current participants in a conference.
|
||||||
|
|
||||||
|
* Added announcement configuration option to user profile. If set the sound
|
||||||
|
file will be played to the user, and only the user, upon joining the
|
||||||
|
conference bridge.
|
||||||
|
|
||||||
|
|
||||||
|
Dial
|
||||||
|
-------------------
|
||||||
|
* Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
|
||||||
|
channels respectively before the callee channels are called.
|
||||||
|
|
||||||
|
|
||||||
|
ExternalIVR
|
||||||
|
-------------------
|
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|
* Added support for IPv6.
|
||||||
|
|
||||||
|
* Add interrupt ('I') command to ExternalIVR. Sending this command from an
|
||||||
|
external process will cause the current playlist to be cleared, including
|
||||||
|
stopping any audio file that is currently playing. This is useful when you
|
||||||
|
want to interrupt audio playback only when specific DTMF is entered by the
|
||||||
|
caller.
|
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|
|
||||||
|
|
||||||
|
FollowMe
|
||||||
|
-------------------
|
||||||
|
* A new option, 'I' has been added to app_followme. By setting this option,
|
||||||
|
Asterisk will not update the caller with connected line changes when they
|
||||||
|
occur. This is similar to app_dial and app_queue.
|
||||||
|
|
||||||
|
* The 'N' option is now ignored if the call is already answered.
|
||||||
|
|
||||||
|
* Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
|
||||||
|
and caller channels respectively before the callee channels are called.
|
||||||
|
|
||||||
|
* The winning FollowMe outgoing call is now put on hold if the caller put it on
|
||||||
|
hold.
|
||||||
|
|
||||||
|
|
||||||
|
MixMonitor
|
||||||
|
------------------
|
||||||
|
* MixMonitor hooks now have IDs associated with them which can be used to
|
||||||
|
assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
|
||||||
|
will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
|
||||||
|
now accepts that ID as an argument.
|
||||||
|
|
||||||
|
* Added 'm' option, which stores a copy of the recording as a voicemail in the
|
||||||
|
indicated mailboxes.
|
||||||
|
|
||||||
|
|
||||||
|
OSP Applications
|
||||||
|
-------------------
|
||||||
|
* Increased the default number of allowed destinations from 5 to 12.
|
||||||
|
|
||||||
|
|
||||||
|
Page
|
||||||
|
-------------------
|
||||||
|
* The app_page application now no longer depends on DAHDI or app_meetme. It
|
||||||
|
has been re-architected to use app_confbridge internally.
|
||||||
|
|
||||||
|
|
||||||
|
Queue
|
||||||
|
-------------------
|
||||||
* Added queue options autopausebusy and autopauseunavail for automatically
|
* Added queue options autopausebusy and autopauseunavail for automatically
|
||||||
pausing a queue member when their device reports busy or congestion.
|
pausing a queue member when their device reports busy or congestion.
|
||||||
|
|
||||||
* The 'ignorebusy' option for queue members has been deprecated in favor of
|
* The 'ignorebusy' option for queue members has been deprecated in favor of
|
||||||
the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
|
the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
|
||||||
added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
|
added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
|
||||||
@@ -181,59 +136,281 @@ Queue changes
|
|||||||
'ringinuse' setting and does not override per member settings like it does
|
'ringinuse' setting and does not override per member settings like it does
|
||||||
in earlier versions.
|
in earlier versions.
|
||||||
|
|
||||||
Voicemail changes
|
* Added 'F()' option. Similar to the dial option, this can be supplied with
|
||||||
-----------------
|
arguments indicating where the callee should go after the caller is hung up,
|
||||||
* When voicemail plays a message's envelope with saycid set to yes, when reaching
|
or without options specified, the priority after the Queue will be used.
|
||||||
the caller id field it will play a recording of a file with the same base name
|
|
||||||
as the sender's callerid if there is a similarly named file in
|
|
||||||
<astspooldir>/recordings/callerids/
|
|
||||||
|
|
||||||
Applications
|
* Added new option log_member_name_as_agent, which will cause the membername to
|
||||||
------------
|
be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
|
||||||
|
state_interface has been set.
|
||||||
|
|
||||||
|
|
||||||
|
SayUnixTime
|
||||||
|
------------------
|
||||||
* Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
|
* Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
|
||||||
when receiving DTMF. Use the 'j' option to enable extension jumping. Also
|
when receiving DTMF. Use the 'j' option to enable extension jumping. Also
|
||||||
changed arguments to SayUnixTime so that every option is truly optional even
|
changed arguments to SayUnixTime so that every option is truly optional even
|
||||||
when using multiple options (so that j option could be used without having to
|
when using multiple options (so that j option could be used without having to
|
||||||
manually specify timezone and format) There are other beneftis eg. format can
|
manually specify timezone and format) There are other benefits, e.g., format
|
||||||
now be used without specifying time zone as well.
|
can now be used without specifying time zone as well.
|
||||||
* Added 'F()' option to Queue and Bridge. Similar to the dial option, these can
|
|
||||||
be supplied with arguments indicating where the callee should go after the caller
|
|
||||||
is hung up, or without options specified, the priority after the Queue/Bridge
|
|
||||||
will be used.
|
|
||||||
* Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
|
|
||||||
channels respectively before the callee channels are called.
|
|
||||||
|
|
||||||
Parking
|
|
||||||
------------
|
|
||||||
* New per parking lot options: comebackcontext and comebackdialtime. See
|
|
||||||
configs/features.conf.sample for more details.
|
|
||||||
|
|
||||||
* Channel variable PARKER is now set when comebacktoorigin is disabled in
|
Voicemail
|
||||||
a parking lot.
|
------------------
|
||||||
|
* Addition of the VM_INFO function - see Function changes.
|
||||||
|
|
||||||
* MixMonitor hooks now have IDs associated with them which can be used to assign
|
* The imapserver, imapport, and imapflags configuration options can now be
|
||||||
a target to StopMixMonitor. Use of MixMonitor's i(variable) option will allow
|
overriden on a user by user basis.
|
||||||
storage of the MixMontior ID in a channel variable. StopMixmonitor now accepts
|
|
||||||
that ID as an argument.
|
|
||||||
|
|
||||||
CDR postgresql driver changes
|
* When voicemail plays a message's envelope with saycid set to yes, when
|
||||||
-----------------------------
|
reaching the caller id field it will play a recording of a file with the same
|
||||||
* Added command "cdr show pgsql status" to check connection status
|
base name as the sender's callerid if there is a similarly named file in
|
||||||
|
<astspooldir>/recordings/callerids/
|
||||||
|
|
||||||
AMI (Asterisk Manager Interface) changes
|
* Voicemails now contains a unique message identifier "msg_id", which is stored
|
||||||
----------------------------------------
|
in the message envelope with the sound files. IMAP backends will now store
|
||||||
* Originate now generates an error response if the extension given
|
the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
|
||||||
is not found in the dialplan
|
backends will store the message identifier in a "msg_id" column. See
|
||||||
|
UPGRADE.txt for more information.
|
||||||
|
|
||||||
* MixMonitor will now show IDs associated with the mixmonitor upon creating them
|
* Added VoiceMailPlayMsg application. This application will play a single
|
||||||
if the i(variable) option is used. StopMixMonitor will accept MixMonitorID as
|
voicemail message from a mailbox. The result of the application, SUCCESS or
|
||||||
on option to close specific MixMonitors.
|
FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
|
||||||
|
|
||||||
* The SIPshowpeer manager action response field "SIP-Forcerport" has been updated
|
|
||||||
to include information about peers configured with nat=auto_force_rport by
|
Functions
|
||||||
returning "A" if auto_force_rport is set and nat is detected, and "a" if it is
|
------------------
|
||||||
set and nat is not detected. "Y" and "N" are still returned if auto_force_rport
|
* Hangup handlers can be attached to channels using the CHANNEL() function.
|
||||||
is not enabled.
|
Hangup handlers will run when the channel is hung up similar to the h
|
||||||
|
extension. The hangup_handler_push option will push a GoSub compatible
|
||||||
|
location in the dialplan onto the channel's hangup handler stack. The
|
||||||
|
hangup_handler_pop option will remove the last added location, and optionally
|
||||||
|
replace it with a new GoSub compatible location. The hangup_handler_wipe
|
||||||
|
option will remove all locations on the stack, and optionally add a new
|
||||||
|
location.
|
||||||
|
|
||||||
|
* The expression parser now recognizes the ABS() absolute value function,
|
||||||
|
which will convert negative floating point values to positive values.
|
||||||
|
|
||||||
|
* FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
|
||||||
|
control of faxdetect.
|
||||||
|
|
||||||
|
* Addition of the VM_INFO function that can be used to retrieve voicemail
|
||||||
|
user information, such as the email address and full name.
|
||||||
|
The MAILBOX_EXISTS dialplan function has been deprecated in favour of
|
||||||
|
VM_INFO.
|
||||||
|
|
||||||
|
* The REDIRECTING function now supports the redirecting original party id
|
||||||
|
and reason.
|
||||||
|
|
||||||
|
* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
|
||||||
|
lets you set some of the configuration options from the [general] section
|
||||||
|
of features.conf on a per-channel basis. FEATUREMAP() lets you customize
|
||||||
|
the key sequence used to activate built-in features, such as blindxfer,
|
||||||
|
and automon. See the built-in documentation for details.
|
||||||
|
|
||||||
|
* MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
|
||||||
|
instead of simply the uri. This is the format that MessageSend() can use
|
||||||
|
in the from parameter for outgoing SIP messages.
|
||||||
|
|
||||||
|
* Added the PRESENCE_STATE function. This allows retrieving presence state
|
||||||
|
information from any presence state provider. It also allows setting
|
||||||
|
presence state information from a CustomPresence presence state provider.
|
||||||
|
See AMI/CLI changes for related commands.
|
||||||
|
|
||||||
|
|
||||||
|
Channel Drivers
|
||||||
|
------------------
|
||||||
|
|
||||||
|
chan_local
|
||||||
|
------------------
|
||||||
|
* Added a manager event "LocalBridge" for local channel call bridges between
|
||||||
|
the two pseudo-channels created.
|
||||||
|
|
||||||
|
|
||||||
|
chan_dahdi
|
||||||
|
------------------
|
||||||
|
* Added dialtone_detect option for analog ports to disconnect incoming
|
||||||
|
calls when dialtone is detected.
|
||||||
|
|
||||||
|
* Added option colp_send to send ISDN connected line information. Allowed
|
||||||
|
settings are block, to not send any connected line information; connect, to
|
||||||
|
send connected line information on initial connect; and update, to send
|
||||||
|
information on any update during a call. Default is update.
|
||||||
|
|
||||||
|
|
||||||
|
chan_motif
|
||||||
|
------------------
|
||||||
|
* A new channel driver named chan_motif has been added which provides support for
|
||||||
|
Google Talk and Jingle in a single channel driver. This new channel driver includes
|
||||||
|
support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
|
||||||
|
hold, unhold, and ringing notification. It is also compliant with the current Jingle
|
||||||
|
specification, current Google Jingle specification, and the original Google Talk
|
||||||
|
protocol.
|
||||||
|
|
||||||
|
|
||||||
|
chan_ooh323
|
||||||
|
------------------
|
||||||
|
* Added NAT support for RTP. Setting in config is 'nat', which can be set
|
||||||
|
globally and overriden on a peer by peer basis.
|
||||||
|
|
||||||
|
* Direct media functionality has been added. Options in config are:
|
||||||
|
directmedia (directrtp) and directrtpsetup (earlydirect)
|
||||||
|
|
||||||
|
* ChannelUpdate events now contain a CallRef header.
|
||||||
|
|
||||||
|
|
||||||
|
chan_sip
|
||||||
|
------------------
|
||||||
|
* Asterisk will no longer substitute CID number for CID name in the display
|
||||||
|
name field if CID number exists without a CID name. This change improves
|
||||||
|
compatibility with certain device features such as Avaya IP500's directory
|
||||||
|
lookup service.
|
||||||
|
|
||||||
|
* A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
|
||||||
|
created using that setting to not be removed during SIP reload.
|
||||||
|
|
||||||
|
* Added settings recordonfeature and recordofffeature. When receiving an INFO
|
||||||
|
request with a "Record:" header, this will turn the requested feature on/off.
|
||||||
|
Allowed values are 'automon', 'automixmon', and blank to disable. Note that
|
||||||
|
dynamic features must be enabled and configured properly on the requesting
|
||||||
|
channel for this to function properly.
|
||||||
|
|
||||||
|
* Add support to realtime for the 'callbackextension' option.
|
||||||
|
|
||||||
|
* When multiple peers exist with the same address, but differing
|
||||||
|
callbackextension options, incoming requests that are matched by address
|
||||||
|
will be matched to the peer with the matching callbackextension if it is
|
||||||
|
available.
|
||||||
|
|
||||||
|
* Two new NAT options, auto_force_rport and auto_comedia, have been added
|
||||||
|
which set the force_rport and comedia options automatically if Asterisk
|
||||||
|
detects that an incoming SIP request crossed a NAT after being sent by
|
||||||
|
the remote endpoint.
|
||||||
|
|
||||||
|
* NAT settings are now a combinable list of options. The equivalent of the
|
||||||
|
deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
|
||||||
|
|
||||||
|
* Adds an option send_diversion which can be disabled to prevent
|
||||||
|
diversion headers from automatically being added to INVITE requests.
|
||||||
|
|
||||||
|
* Add support for lightweight NAT keepalive. If enabled a blank packet will
|
||||||
|
be sent to the remote host at a given interval to keep the NAT mapping open.
|
||||||
|
This can be enabled using the keepalive configuration option.
|
||||||
|
|
||||||
|
* Add option 'tonezone' to specify country code for indications. This option
|
||||||
|
can be set both globally and overridden for specific peers.
|
||||||
|
|
||||||
|
* The SIP Security Events Framework now supports IPv6.
|
||||||
|
|
||||||
|
* Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
|
||||||
|
between multiple user agents. When set, for directmedia reinvites,
|
||||||
|
Asterisk will not send an immediate reinvite on an incoming call leg. This
|
||||||
|
option is useful when peered with another SIP user agent that is known to
|
||||||
|
send immediate direct media reinvites upon call establishment.
|
||||||
|
|
||||||
|
* Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
|
||||||
|
as the transport.
|
||||||
|
|
||||||
|
* When a MESSAGE request is received, the address the request was received from
|
||||||
|
is now saved in the SIP_RECVADDR variable.
|
||||||
|
|
||||||
|
* Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
|
||||||
|
parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
|
||||||
|
the ANI2/OLI information is set on the channel, which can be retrieved using
|
||||||
|
the CALLERID function.
|
||||||
|
|
||||||
|
* Peers can now be configured to support negotiation of ICE candidates using
|
||||||
|
the setting icesupport. See res_rtp_asterisk changes for more information.
|
||||||
|
|
||||||
|
* Added support for format attribute negotiation. See the Codecs changes for
|
||||||
|
more information.
|
||||||
|
|
||||||
|
|
||||||
|
chan_skinny
|
||||||
|
------------------
|
||||||
|
* Added skinny version 17 protocol support.
|
||||||
|
|
||||||
|
|
||||||
|
chan_unistim
|
||||||
|
--------------------
|
||||||
|
* Added ability to use multiple lines for a single phone. This allows multiple
|
||||||
|
calls to occur on a single phone, using callwaiting and switching between calls.
|
||||||
|
|
||||||
|
* Added option 'sharpdial' allowing end dialing by pressing # key
|
||||||
|
|
||||||
|
* Added option 'interdigit_timer' to control phone dial timeout
|
||||||
|
|
||||||
|
* Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
|
||||||
|
|
||||||
|
* Added global 'debug' option, that enables debug in channel driver
|
||||||
|
|
||||||
|
* Added ability to translate on-screen menu in multiple languages. Tested on
|
||||||
|
Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
|
||||||
|
ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
|
||||||
|
menu of phone
|
||||||
|
|
||||||
|
* In addition to English added French and Russian languages for on-screen menus
|
||||||
|
|
||||||
|
* Reworked dialing number input: added dialing by timeout, immediate dial on
|
||||||
|
on dialplan compare, phone number length now not limited by screen size
|
||||||
|
|
||||||
|
* Added ability to pickup a call using features.conf defined value and
|
||||||
|
on-screen key
|
||||||
|
|
||||||
|
|
||||||
|
Core
|
||||||
|
------------------
|
||||||
|
* The minimum DTMF duration can now be configured in asterisk.conf
|
||||||
|
as "mindtmfduration". The default value is (as before) set to 80 ms.
|
||||||
|
(previously it was only available in source code)
|
||||||
|
|
||||||
|
* Named ACLs can now be specified in acl.conf and used in configurations that
|
||||||
|
use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
|
||||||
|
used to specify an ACL, a similar form of 'acl' will add a named ACL to the
|
||||||
|
working ACL. In addition, some CLI commands have been added to provide
|
||||||
|
show information and allow for module reloading - see CLI Changes.
|
||||||
|
|
||||||
|
* DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
|
||||||
|
be used within the dynamic weight attribute when specifying a mapping.
|
||||||
|
|
||||||
|
* CEL backends can now be configured to show "USER_DEFINED" in the EventName
|
||||||
|
header, instead of putting the user defined event name there. When enabled
|
||||||
|
the UserDefType header is added for user defined events. This feature is
|
||||||
|
enabled with the setting show_user_defined.
|
||||||
|
|
||||||
|
* Macro has been deprecated in favor of GoSub. For redirecting and connected
|
||||||
|
line purposes use the following variables instead of their macro equivalents:
|
||||||
|
REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
|
||||||
|
CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
|
||||||
|
cc_callback_macro in channel configurations.
|
||||||
|
|
||||||
|
|
||||||
|
AGI
|
||||||
|
------------------
|
||||||
|
* A new channel variable, AGIEXITONHANGUP, has been added which allows
|
||||||
|
Asterisk to behave like it did in Asterisk 1.4 and earlier where the
|
||||||
|
AGI application would exit immediately after a channel hangup is detected.
|
||||||
|
|
||||||
|
* IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
|
||||||
|
are resolved and each address is attempted in turn until one succeeds or
|
||||||
|
all fail.
|
||||||
|
|
||||||
|
|
||||||
|
AMI (Asterisk Manager Interface)
|
||||||
|
------------------
|
||||||
|
* Originate now generates an error response if the extension given is not found
|
||||||
|
in the dialplan
|
||||||
|
|
||||||
|
* MixMonitor will now show IDs associated with the mixmonitor upon creating
|
||||||
|
them if the i(variable) option is used. StopMixMonitor will accept
|
||||||
|
MixMonitorID as an option to close specific MixMonitors.
|
||||||
|
|
||||||
|
* The SIPshowpeer manager action response field "SIP-Forcerport" has been
|
||||||
|
updated to include information about peers configured with
|
||||||
|
nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
|
||||||
|
detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
|
||||||
|
returned if auto_force_rport is not enabled.
|
||||||
|
|
||||||
* Hangup now can take a regular expression as the Channel option. If you want
|
* Hangup now can take a regular expression as the Channel option. If you want
|
||||||
to hangup multiple channels, use /regex/ as the Channel option. Existing
|
to hangup multiple channels, use /regex/ as the Channel option. Existing
|
||||||
@@ -242,49 +419,152 @@ AMI (Asterisk Manager Interface) changes
|
|||||||
|
|
||||||
* Support for IPv6 addresses has been added.
|
* Support for IPv6 addresses has been added.
|
||||||
|
|
||||||
* AMI Events can now be documented in the Asterisk source. Two new CLI
|
* AMI Events can now be documented in the Asterisk source. Note that AMI event
|
||||||
commands have been added to display information about AMI events at run time:
|
documentation is only generated when Asterisk is compiled using 'make full'.
|
||||||
manager show events, which shows a list of all known and documented AMI
|
See the CLI section for commands to display AMI event information.
|
||||||
events, and manager show event [event name], which shows detail information
|
|
||||||
about a specific AMI event. Note that AMI event documentation is only
|
|
||||||
generated when Asterisk is compiled using 'make full'.
|
|
||||||
|
|
||||||
FAX changes
|
* The AMI Hangup event now includes the AccountCode header so you can easily
|
||||||
-----------
|
correlate with AMI Newchannel events.
|
||||||
* FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
|
|
||||||
control of faxdetect.
|
|
||||||
|
|
||||||
DUNDi changes
|
* The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
|
||||||
-------------
|
the StateInterface of the queue member.
|
||||||
* Allow the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to be
|
|
||||||
used within the dynamic weight attribute when specifying a mapping.
|
|
||||||
|
|
||||||
Dialplan functions
|
* Added AMI event SessionTimeout in the Call category that is issued when a
|
||||||
------------------
|
call is terminated due to either RTP stream inactivity or SIP session timer
|
||||||
* Addition of the VM_INFO function that can be used to retrieve voicemail
|
expiration.
|
||||||
user information, such as the email address and full name.
|
|
||||||
The MAILBOX_EXISTS dialplan function has been deprecated in favour of
|
|
||||||
VM_INFO.
|
|
||||||
* The REDIRECTING function now supports the redirecting original party id
|
|
||||||
and reason.
|
|
||||||
* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
|
|
||||||
lets you set some of the configuration options from the [general] section
|
|
||||||
of features.conf on a per-channel basis. FEATUREMAP() lets you customize
|
|
||||||
the key sequence used to activate built-in features, such as blindxfer,
|
|
||||||
and automon. See the built-in documentation for details.
|
|
||||||
|
|
||||||
Followme changes
|
* CEL events can now contain a user defined header UserDefType. See core
|
||||||
-------------
|
changes for more information.
|
||||||
* A new option, 'I' has been added to app_followme.
|
|
||||||
By setting this option, Asterisk will not update the caller with
|
|
||||||
connected line changes when they occur. This is similar to app_dial
|
|
||||||
and app_queue.
|
|
||||||
* The 'N' option is now ignored if the call is already answered.
|
|
||||||
* Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
|
|
||||||
and caller channels respectively before the callee channels are called.
|
|
||||||
|
|
||||||
RTP changes
|
* OOH323 ChannelUpdate events now contain a CallRef header.
|
||||||
-------------
|
|
||||||
|
* Added PresenceState command. This command will report the presence state for
|
||||||
|
the given presence provider.
|
||||||
|
|
||||||
|
* Added Parkinglots command. This will list all parking lots as a series of
|
||||||
|
AMI Parkinglot events.
|
||||||
|
|
||||||
|
* Added MessageSend command. This behaves in the same manner as the
|
||||||
|
MessageSend application, and is a technolgoy agnostic mechanism to send out
|
||||||
|
of call text messages.
|
||||||
|
|
||||||
|
* Added "message" class authorization. This grants an account permission to
|
||||||
|
send out of call messages. Write-only.
|
||||||
|
|
||||||
|
|
||||||
|
CLI
|
||||||
|
-------------------
|
||||||
|
* The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
|
||||||
|
filenames of all running mixmonitors on a channel.
|
||||||
|
|
||||||
|
* The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
|
||||||
|
numeric instead of 0, 1, or 2.
|
||||||
|
|
||||||
|
* "stun show status" will show a table describing how the STUN client is
|
||||||
|
behaving.
|
||||||
|
|
||||||
|
* "acl show [named acl]" will show information regarding a Named ACL. The
|
||||||
|
acl module can be reloaded with "reload acl".
|
||||||
|
|
||||||
|
* Added CLI command to display AMI event information - "manager show events",
|
||||||
|
which shows a list of all known and documented AMI events, and "manager show
|
||||||
|
event [event name]", which shows detail information about a specific AMI
|
||||||
|
event.
|
||||||
|
|
||||||
|
* The result of the CLI command "queue show" now includes the state interface
|
||||||
|
information of the queue member.
|
||||||
|
|
||||||
|
* The command "core set verbose" will now set a separate level of logging for
|
||||||
|
each remote console without affecting any other console.
|
||||||
|
|
||||||
|
* Added command "cdr show pgsql status" to check connection status
|
||||||
|
|
||||||
|
* "sip show channel" will now display the complete route set.
|
||||||
|
|
||||||
|
* Added "presencestate list" command. This command will list all custom
|
||||||
|
presence states that have been set by using the PRESENCE_STATE dialplan
|
||||||
|
function.
|
||||||
|
|
||||||
|
* Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
|
||||||
|
command. This changes a custom presence to a new state.
|
||||||
|
|
||||||
|
|
||||||
|
Codecs
|
||||||
|
-------------------
|
||||||
|
* Codec lists may now be modified by the '!' character, to allow succinct
|
||||||
|
specification of a list of codecs allowed and disallowed, without the
|
||||||
|
requirement to use two different keywords. For example, to specify all
|
||||||
|
codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
|
||||||
|
|
||||||
|
* Add support for parsing SDP attributes, generating SDP attributes, and
|
||||||
|
passing it through. This support includes codecs such as H.263, H.264, SILK,
|
||||||
|
and CELT. You are able to set up a call and have attribute information pass.
|
||||||
|
This should help considerably with video calls.
|
||||||
|
|
||||||
|
|
||||||
|
Logging
|
||||||
|
-------------------
|
||||||
|
* Asterisk version and build information is now logged at the beginning of a
|
||||||
|
log file.
|
||||||
|
|
||||||
|
* Threads belonging to a particular call are now linked with callids which get
|
||||||
|
added to any log messages produced by those threads. Log messages can now be
|
||||||
|
easily identified as involved with a certain call by looking at their call id.
|
||||||
|
Call ids may also be attached to log messages for just about any case where
|
||||||
|
it can be determined to be related to a particular call.
|
||||||
|
|
||||||
|
* Each logging destination and console now have an independent notion of the
|
||||||
|
current verbosity level. Logger.conf now allows an optional argument to
|
||||||
|
the 'verbose' specifier, indicating the level of verbosity sent to that
|
||||||
|
particular logging destination. Additionally, remote consoles now each
|
||||||
|
have their own verbosity level. The command 'core set verbose' will now set
|
||||||
|
a separate level for each remote console without affecting any other
|
||||||
|
console.
|
||||||
|
|
||||||
|
|
||||||
|
Music On Hold
|
||||||
|
-------------------
|
||||||
|
* Added 'announcement' option which will play at the start of MOH and between
|
||||||
|
songs in modes of MOH that can detect transitions between songs (eg.
|
||||||
|
files, mp3, etc).
|
||||||
|
|
||||||
|
|
||||||
|
Parking
|
||||||
|
-------------------
|
||||||
|
* New per parking lot options: comebackcontext and comebackdialtime. See
|
||||||
|
configs/features.conf.sample for more details.
|
||||||
|
|
||||||
|
* Channel variable PARKER is now set when comebacktoorigin is disabled in
|
||||||
|
a parking lot.
|
||||||
|
|
||||||
|
* Channel variable PARKEDCALL is now set with the name of the parking lot
|
||||||
|
when a timeout occurs.
|
||||||
|
|
||||||
|
|
||||||
|
CDRs
|
||||||
|
-------------------
|
||||||
|
|
||||||
|
CDR Postgresql Driver
|
||||||
|
-------------------
|
||||||
|
* Added command "cdr show pgsql status" to check connection status
|
||||||
|
|
||||||
|
|
||||||
|
CDR Adaptive ODBC Driver
|
||||||
|
-------------------
|
||||||
|
* Added schema option for databases that support specifying a schema.
|
||||||
|
|
||||||
|
|
||||||
|
Resource Modules
|
||||||
|
-------------------
|
||||||
|
|
||||||
|
Calendars
|
||||||
|
-------------------
|
||||||
|
* A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
|
||||||
|
CALENDAR_WRITE has completed successfully.
|
||||||
|
|
||||||
|
|
||||||
|
res_rtp_asterisk
|
||||||
|
-------------------
|
||||||
* A new option, 'probation' has been added to rtp.conf
|
* A new option, 'probation' has been added to rtp.conf
|
||||||
RTP in strictrtp mode can now require more than 1 packet to exit learning
|
RTP in strictrtp mode can now require more than 1 packet to exit learning
|
||||||
mode with a new source (and by default requires 4). The probation option
|
mode with a new source (and by default requires 4). The probation option
|
||||||
@@ -294,14 +574,13 @@ RTP changes
|
|||||||
mode has successfully exited. These changes are based on how pjmedia handles
|
mode has successfully exited. These changes are based on how pjmedia handles
|
||||||
media sources and source changes.
|
media sources and source changes.
|
||||||
|
|
||||||
Text Messaging
|
* Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
|
||||||
--------------
|
enabled or disabled using the icesupport setting. A variety of other
|
||||||
* MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
|
settings have been introduced to configure STUN/TURN connections.
|
||||||
instead of simply the uri. This is the format that MessageSend() can use
|
|
||||||
in the from parameter for outgoing SIP messages.
|
|
||||||
|
|
||||||
res_corosync
|
res_corosync
|
||||||
------------
|
-------------------
|
||||||
* A new module, res_corosync, has been introduced. This module uses the
|
* A new module, res_corosync, has been introduced. This module uses the
|
||||||
Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
|
Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
|
||||||
of Asterisk servers to both Message Waiting Indication (MWI) and/or
|
of Asterisk servers to both Message Waiting Indication (MWI) and/or
|
||||||
@@ -309,28 +588,26 @@ res_corosync
|
|||||||
is a replacement for the res_ais module that was in previous releases of
|
is a replacement for the res_ais module that was in previous releases of
|
||||||
Asterisk.
|
Asterisk.
|
||||||
|
|
||||||
AGI
|
|
||||||
---
|
|
||||||
* A new channel variable, AGIEXITONHANGUP, has been added which allows
|
|
||||||
Asterisk to behave like it did in Asterisk 1.4 and earlier where the
|
|
||||||
AGI application would exit immediately after a channel hangup is detected.
|
|
||||||
* IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
|
|
||||||
are resolved and each address is attempted in turn until one succeeds or
|
|
||||||
all fail.
|
|
||||||
|
|
||||||
chan_ooh323
|
res_xmpp
|
||||||
-----------
|
-------------------
|
||||||
* Direct media functionality has been added.
|
* This module adds a cleaned up, drop-in replacement for res_jabber called
|
||||||
Options in config are: directmedia (directrtp) and directrtpsetup (earlydirect)
|
res_xmpp. This provides the same externally facing functionality but is
|
||||||
|
implemented differently internally. res_jabber has been deprecated in favor
|
||||||
|
of res_xmpp; please see the UPGRADE.txt file for more information.
|
||||||
|
|
||||||
|
|
||||||
|
Scripts
|
||||||
|
-------------------
|
||||||
|
* The safe_asterisk script has been updated to allow several of its parameters
|
||||||
|
to be set from environment variables. This also enables a custom run
|
||||||
|
directory of Asterisk to be specified, instead of defaulting to /tmp.
|
||||||
|
|
||||||
|
* The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
|
||||||
|
its value to determine the directory to assume is the top-level directory of
|
||||||
|
the source tree. If the variable is not set, it defaults to the current
|
||||||
|
behavior and uses the current working directory.
|
||||||
|
|
||||||
chan_motif
|
|
||||||
----------
|
|
||||||
* A new channel driver named chan_motif has been added which provides support for
|
|
||||||
Google Talk and Jingle in a single channel driver. This new channel driver includes
|
|
||||||
support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
|
|
||||||
hold, unhold, and ringing notification. It is also compliant with the current Jingle
|
|
||||||
specification, current Google Jingle specification, and the original Google Talk
|
|
||||||
protocol.
|
|
||||||
|
|
||||||
------------------------------------------------------------------------------
|
------------------------------------------------------------------------------
|
||||||
--- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
|
--- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
|
||||||
|
|||||||
Reference in New Issue
Block a user