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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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||||
|
||||
Build System
|
||||
----
|
||||
* 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
|
||||
event documentation. Note that your system must have Python in order for
|
||||
this make target to succeed.
|
||||
|
||||
Core
|
||||
----
|
||||
* The expression parser now recognizes the ABS() absolute value function,
|
||||
which will convert negative floating point values to positive values.
|
||||
-------------------
|
||||
* The Asterisk build system will now build and install a shared library
|
||||
(libasteriskssl.so) used to wrap various initialization and shutdown functions
|
||||
from the libssl and libcrypto libraries provided by OpenSSL. This is done so
|
||||
@@ -31,147 +21,112 @@ Core
|
||||
modules that are loaded into Asterisk, since they should only be called once
|
||||
in any single process. If desired, this feature can be disabled by supplying
|
||||
the "--disable-asteriskssl" option to the configure script.
|
||||
* 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.
|
||||
* 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)
|
||||
* 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.
|
||||
* 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 informational and configuration reload capabilities to
|
||||
this feature ('acl show [named acl]' and 'reload acl').
|
||||
* Hangup handlers can be attached to channels using the CHANNEL(hangup_handler_xxx)
|
||||
options. Hangup handlers will run when the channel is hung up similar to the
|
||||
h extension.
|
||||
* The AMI Hangup event now includes the AccountCode header so you can easily
|
||||
correlate with AMI Newchannel events.
|
||||
|
||||
CLI Changes
|
||||
-------------------
|
||||
* mixmonitor list <channel> command will now show MixMonitor ID, and the filenames
|
||||
of all running mixmonitors on a channel.
|
||||
* The debuglevel 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.
|
||||
* 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
|
||||
event documentation. Note that your system must have Python in order for
|
||||
this make target to succeed.
|
||||
|
||||
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.
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||||
|
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Voicemail
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||||
------------------
|
||||
* Addition of the VM_INFO function - see Dialplan function changes
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||||
* The imapserver, imapport, and imapflags configuration options can now be
|
||||
overriden on a user by user basis.
|
||||
|
||||
SIP Changes
|
||||
-----------
|
||||
* Asterisk will no longer substitute CID number for CID name into 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.
|
||||
* 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.
|
||||
* 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.
|
||||
* 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.
|
||||
* Adds an option send_diversion which can be disabled to prevent
|
||||
diversion headers from automatically being added to invites.
|
||||
* 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 support for WebSocket transport. This can be configured using 'ws' or 'wss'
|
||||
as the transport.
|
||||
|
||||
Chan_local changes
|
||||
------------------
|
||||
* Added a manager event "LocalBridge" for local channel call bridges between
|
||||
the two pseudo-channels created.
|
||||
|
||||
Chan_dahdi changes
|
||||
------------------
|
||||
* Added dialtone_detect option for analog ports to disconnect incoming
|
||||
calls when dialtone is detected.
|
||||
|
||||
------------------------------------------------------------------------------
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||||
--- Functionality changes since Asterisk 10.4.0 ------------------------------
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||||
------------------------------------------------------------------------------
|
||||
|
||||
Build System
|
||||
------------
|
||||
* The optimization portion of the build system has been reworked to avoid
|
||||
broken builds on certain architectures. All architecture-specific
|
||||
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
|
||||
be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
|
||||
|
||||
------------------------------------------------------------------------------
|
||||
--- Functionality changes since Asterisk 10.3.0 ------------------------------
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||||
------------------------------------------------------------------------------
|
||||
* BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
|
||||
make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
|
||||
|
||||
Chan_unistim changes
|
||||
--------------------
|
||||
* Added ability to use multiple lines on phone, so for one device in
|
||||
configuration multiple lines can be defined, it allows to have multiple calls
|
||||
on one phone, callwaiting and switching between calls.
|
||||
* Added option 'sharpdial' allowing end dialing by pressing # key
|
||||
* Added option 'interdigit_timer' for controll phone dial timeout
|
||||
* Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
|
||||
* Added global 'debug' option, that enables debug in channel driver
|
||||
* Added ability for translation on-screen menu to 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
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||||
menu of phone
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||||
* 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 for pickup a call using features.conf defined value and
|
||||
on-screen key
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||||
* Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
|
||||
previously parsed the header file to obtain the version of Asterisk, you
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||||
will now have to go through Asterisk to get the version information.
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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
|
||||
requirement to use two different keywords. For example, to specify all
|
||||
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
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||||
---------------------
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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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||||
Applications
|
||||
-------------------
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||||
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||||
Queue changes
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||||
-------------
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||||
Bridge
|
||||
-------------------
|
||||
* 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
|
||||
-------------------
|
||||
* Added support for IPv6.
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||||
|
||||
* Add interrupt ('I') command to ExternalIVR. Sending this command from an
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||||
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
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||||
caller.
|
||||
|
||||
|
||||
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
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||||
indicated mailboxes.
|
||||
|
||||
|
||||
OSP Applications
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||||
-------------------
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||||
* Increased the default number of allowed destinations from 5 to 12.
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||||
|
||||
|
||||
Page
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||||
-------------------
|
||||
* 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
|
||||
pausing a queue member when their device reports busy or congestion.
|
||||
|
||||
* The 'ignorebusy' option for queue members has been deprecated in favor of
|
||||
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
|
||||
@@ -181,59 +136,281 @@ Queue changes
|
||||
'ringinuse' setting and does not override per member settings like it does
|
||||
in earlier versions.
|
||||
|
||||
Voicemail 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 base name
|
||||
as the sender's callerid if there is a similarly named file in
|
||||
<astspooldir>/recordings/callerids/
|
||||
* 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.
|
||||
|
||||
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
|
||||
when receiving DTMF. Use the 'j' option to enable extension jumping. Also
|
||||
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
|
||||
manually specify timezone and format) There are other beneftis eg. format 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.
|
||||
manually specify timezone and format) There are other benefits, e.g., format
|
||||
can now be used without specifying time zone as well.
|
||||
|
||||
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.
|
||||
Voicemail
|
||||
------------------
|
||||
* Addition of the VM_INFO function - see Function changes.
|
||||
|
||||
* 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.
|
||||
* The imapserver, imapport, and imapflags configuration options can now be
|
||||
overriden on a user by user basis.
|
||||
|
||||
CDR postgresql driver changes
|
||||
-----------------------------
|
||||
* Added command "cdr show pgsql status" to check connection status
|
||||
* 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
|
||||
base name as the sender's callerid if there is a similarly named file in
|
||||
<astspooldir>/recordings/callerids/
|
||||
|
||||
AMI (Asterisk Manager Interface) changes
|
||||
----------------------------------------
|
||||
* Originate now generates an error response if the extension given
|
||||
is not found in the dialplan
|
||||
* 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
|
||||
the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
|
||||
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
|
||||
if the i(variable) option is used. StopMixMonitor will accept MixMonitorID as
|
||||
on option to close specific MixMonitors.
|
||||
* Added VoiceMailPlayMsg application. This application will play a single
|
||||
voicemail message from a mailbox. The result of the application, SUCCESS or
|
||||
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
|
||||
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.
|
||||
|
||||
Functions
|
||||
------------------
|
||||
* Hangup handlers can be attached to channels using the CHANNEL() function.
|
||||
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
|
||||
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.
|
||||
|
||||
* AMI Events can now be documented in the Asterisk source. Two new CLI
|
||||
commands have been added to display information about AMI events at run time:
|
||||
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. Note that AMI event documentation is only
|
||||
generated when Asterisk is compiled using 'make full'.
|
||||
* AMI Events can now be documented in the Asterisk source. Note that AMI event
|
||||
documentation is only generated when Asterisk is compiled using 'make full'.
|
||||
See the CLI section for commands to display AMI event information.
|
||||
|
||||
FAX changes
|
||||
-----------
|
||||
* FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
|
||||
control of faxdetect.
|
||||
* The AMI Hangup event now includes the AccountCode header so you can easily
|
||||
correlate with AMI Newchannel events.
|
||||
|
||||
DUNDi changes
|
||||
-------------
|
||||
* Allow the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to be
|
||||
used within the dynamic weight attribute when specifying a mapping.
|
||||
* The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
|
||||
the StateInterface of the queue member.
|
||||
|
||||
Dialplan functions
|
||||
------------------
|
||||
* 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.
|
||||
* 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
|
||||
expiration.
|
||||
|
||||
Followme changes
|
||||
-------------
|
||||
* 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.
|
||||
* CEL events can now contain a user defined header UserDefType. See core
|
||||
changes for more information.
|
||||
|
||||
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
|
||||
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
|
||||
@@ -294,14 +574,13 @@ RTP changes
|
||||
mode has successfully exited. These changes are based on how pjmedia handles
|
||||
media sources and source changes.
|
||||
|
||||
Text Messaging
|
||||
--------------
|
||||
* 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.
|
||||
* 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
|
||||
settings have been introduced to configure STUN/TURN connections.
|
||||
|
||||
|
||||
res_corosync
|
||||
------------
|
||||
-------------------
|
||||
* A new module, res_corosync, has been introduced. This module uses the
|
||||
Corosync cluster enginer (http://www.corosync.org) to allow a local cluster
|
||||
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
|
||||
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
|
||||
-----------
|
||||
* Direct media functionality has been added.
|
||||
Options in config are: directmedia (directrtp) and directrtpsetup (earlydirect)
|
||||
res_xmpp
|
||||
-------------------
|
||||
* This module adds a cleaned up, drop-in replacement for res_jabber called
|
||||
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 -------------------
|
||||
|
||||
Reference in New Issue
Block a user