|
|
|
|
@ -78,7 +78,7 @@ ExternalIVR
|
|
|
|
|
-------------------
|
|
|
|
|
* Added support for IPv6.
|
|
|
|
|
|
|
|
|
|
* Add interrupt ('I') command to ExternalIVR. Sending this command from an
|
|
|
|
|
* 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
|
|
|
|
|
@ -254,7 +254,7 @@ 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:
|
|
|
|
|
* Direct media functionality has been added. Options in config are:
|
|
|
|
|
directmedia (directrtp) and directrtpsetup (earlydirect)
|
|
|
|
|
|
|
|
|
|
* ChannelUpdate events now contain a CallRef header.
|
|
|
|
|
@ -354,16 +354,16 @@ chan_unistim
|
|
|
|
|
* 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
|
|
|
|
|
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
|
|
|
|
|
* 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
|
|
|
|
|
* Added ability to pickup a call using features.conf defined value and
|
|
|
|
|
on-screen key
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
@ -421,7 +421,7 @@ AMI (Asterisk Manager Interface)
|
|
|
|
|
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
|
|
|
|
|
to hangup multiple channels, use /regex/ as the Channel option. Existing
|
|
|
|
|
behavior to hanging up a single channel is unchanged, but if you pass a regex,
|
|
|
|
|
the manager will send you a list of channels back that were hung up.
|
|
|
|
|
|
|
|
|
|
@ -509,6 +509,8 @@ Codecs
|
|
|
|
|
and CELT. You are able to set up a call and have attribute information pass.
|
|
|
|
|
This should help considerably with video calls.
|
|
|
|
|
|
|
|
|
|
* The iLBC codec can now use a system-provided iLBC library if one is installed,
|
|
|
|
|
just like the GSM codec.
|
|
|
|
|
|
|
|
|
|
Logging
|
|
|
|
|
-------------------
|
|
|
|
|
@ -545,7 +547,7 @@ Parking
|
|
|
|
|
* 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
|
|
|
|
|
* Channel variable PARKEDCALL is now set with the name of the parking lot
|
|
|
|
|
when a timeout occurs.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
@ -567,7 +569,7 @@ Resource Modules
|
|
|
|
|
|
|
|
|
|
Calendars
|
|
|
|
|
-------------------
|
|
|
|
|
* A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
|
|
|
|
|
* A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
|
|
|
|
|
CALENDAR_WRITE has completed successfully.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
@ -711,7 +713,7 @@ ConfBridge
|
|
|
|
|
mixing audio at sample rates ranging from 8khz-96khz.
|
|
|
|
|
* CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
|
|
|
|
|
and bridge profiles on a channel.
|
|
|
|
|
* CONFBRIDGE_INFO dialplan function capable of retrieving information
|
|
|
|
|
* CONFBRIDGE_INFO dialplan function capable of retrieving information
|
|
|
|
|
about a conference such as locked status and number of parties, admins,
|
|
|
|
|
and marked users.
|
|
|
|
|
* Addition of video_mode option in confbridge.conf for adding video support
|
|
|
|
|
@ -759,7 +761,7 @@ Calendaring
|
|
|
|
|
|
|
|
|
|
MixMonitor
|
|
|
|
|
--------------------------
|
|
|
|
|
* Added two new options, r and t with file name arguments to record
|
|
|
|
|
* Added two new options, r and t with file name arguments to record
|
|
|
|
|
single direction (unmixed) audio recording separate from the bidirectional
|
|
|
|
|
(mixed) recording. The mixed file name argument is optional now as long
|
|
|
|
|
as at least one recording option is used.
|
|
|
|
|
@ -1023,7 +1025,7 @@ Applications
|
|
|
|
|
* Added 'y' option to app_record. This option enables a mode where any DTMF digit
|
|
|
|
|
received will terminate recording.
|
|
|
|
|
* Voicemail now supports per mailbox settings for folders when using IMAP storage.
|
|
|
|
|
Previously the folder could only be set per context, but has now been extended
|
|
|
|
|
Previously the folder could only be set per context, but has now been extended
|
|
|
|
|
using the imapfolder option.
|
|
|
|
|
* Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
|
|
|
|
|
* Voicemail now allows the pager date format to be specified separately from the
|
|
|
|
|
@ -1274,7 +1276,7 @@ libpri channel driver (chan_dahdi) DAHDI changes
|
|
|
|
|
to eliminate tromboned calls. A tromboned call goes out an interface and comes
|
|
|
|
|
back into the same interface. Tromboned calls happen because of call routing,
|
|
|
|
|
call deflection, call forwarding, and call transfer.
|
|
|
|
|
* Added the ability to send and receive ETSI Advice-Of-Charge messages.
|
|
|
|
|
* Added the ability to send and receive ETSI Advice-Of-Charge messages.
|
|
|
|
|
* Added the ability to support call waiting calls. (The SETUP has no B channel
|
|
|
|
|
assigned.)
|
|
|
|
|
* Added Malicious Call ID (MCID) event to the AMI call event class.
|
|
|
|
|
@ -1298,7 +1300,7 @@ Asterisk Manager Interface
|
|
|
|
|
* The configuration file manager.conf now supports a channelvars option, which
|
|
|
|
|
specifies a list of channel variables to include in each channel-oriented
|
|
|
|
|
event.
|
|
|
|
|
* The redirect command now has new parameters ExtraContext, ExtraExtension,
|
|
|
|
|
* The redirect command now has new parameters ExtraContext, ExtraExtension,
|
|
|
|
|
and ExtraPriority to allow redirecting the second channel to a different
|
|
|
|
|
location than the first.
|
|
|
|
|
* Added new event "JabberStatus" in the Jabber module to monitor buddies
|
|
|
|
|
@ -1407,7 +1409,7 @@ Miscellaneous
|
|
|
|
|
of unit tests with the purpose of verifying the operation of C functions.
|
|
|
|
|
* SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
|
|
|
|
|
XMPP text messages to the remote JID.
|
|
|
|
|
* Modules.conf has a new option - "require" - that marks a module as critical for
|
|
|
|
|
* Modules.conf has a new option - "require" - that marks a module as critical for
|
|
|
|
|
the execution of Asterisk.
|
|
|
|
|
If one of the required modules fail to load, Asterisk will exit with a return
|
|
|
|
|
code set to 2.
|
|
|
|
|
@ -1454,7 +1456,7 @@ CLI Changes
|
|
|
|
|
which applies the setting to the entire module specified, regardless of which source
|
|
|
|
|
files it was built from.
|
|
|
|
|
* New 'manager show settings' command showing the current settings loaded from
|
|
|
|
|
manager.conf.
|
|
|
|
|
manager.conf.
|
|
|
|
|
* Added 'all' keyword to the CLI command "channel request hangup" so that you can send
|
|
|
|
|
the channel hangup request to all channels.
|
|
|
|
|
* Added a "core reload" CLI command that executes a global reload of Asterisk.
|
|
|
|
|
@ -1474,7 +1476,7 @@ SIP Changes
|
|
|
|
|
remote services. For backwards compatibility, "secret" still has the
|
|
|
|
|
same function as before, but now you can configure both a remote secret and a
|
|
|
|
|
local secret for mutual authentication.
|
|
|
|
|
* If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
|
|
|
|
|
* If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
|
|
|
|
|
the sound will be played to the target of an attended transfer
|
|
|
|
|
* Added two new configuration options, "qualifygap" and "qualifypeers", which allow
|
|
|
|
|
finer control over how many peers Asterisk will qualify and the gap between them
|
|
|
|
|
@ -1494,7 +1496,7 @@ SIP Changes
|
|
|
|
|
as a mailbox. Please see the sip.conf.sample file for more information.
|
|
|
|
|
* Added a function to remove SIP headers added in the dialplan before the
|
|
|
|
|
first INVITE is generated - SIPRemoveHeader()
|
|
|
|
|
* Channel variables set with setvar= in a device configuration is now
|
|
|
|
|
* Channel variables set with setvar= in a device configuration is now
|
|
|
|
|
set both for inbound and outbound calls.
|
|
|
|
|
* Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
|
|
|
|
|
|
|
|
|
|
@ -1650,7 +1652,7 @@ LDAP Schema File Additions
|
|
|
|
|
- Gives more configuration Flags for SIP-Users available (tested)
|
|
|
|
|
- Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
|
|
|
|
|
without extensibleObject (which really should be the last resort); gives
|
|
|
|
|
also additional possibilities for LDAP-filter
|
|
|
|
|
also additional possibilities for LDAP-filter
|
|
|
|
|
|
|
|
|
|
------------------------------------------------------------------------------
|
|
|
|
|
--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
|
|
|
|
|
@ -1673,8 +1675,8 @@ Dialplan Functions
|
|
|
|
|
* Added a new dialplan function, AST_CONFIG(), which allows you to access
|
|
|
|
|
variables from an Asterisk configuration file.
|
|
|
|
|
* The JACK_HOOK function now has a c() option to supply a custom client name.
|
|
|
|
|
* Added two new dialplan functions from libspeex for audio gain control and
|
|
|
|
|
denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
|
|
|
|
|
* Added two new dialplan functions from libspeex for audio gain control and
|
|
|
|
|
denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
|
|
|
|
|
rx directions of a channel from the dialplan.
|
|
|
|
|
* The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
|
|
|
|
|
based on other parameters. The default is still to search based on the
|
|
|
|
|
@ -1733,7 +1735,7 @@ Application Changes
|
|
|
|
|
participant on the bridged channel as well.
|
|
|
|
|
* Chanspy has a new option, 'n', which will allow for the spied-on party's name
|
|
|
|
|
to be spoken instead of the channel name or number. For more information on the
|
|
|
|
|
use of this option, issue the command "core show application ChanSpy" from the
|
|
|
|
|
use of this option, issue the command "core show application ChanSpy" from the
|
|
|
|
|
Asterisk CLI.
|
|
|
|
|
* Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
|
|
|
|
|
spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
|
|
|
|
|
@ -1743,11 +1745,11 @@ Application Changes
|
|
|
|
|
* ExternalIVR now takes several options that affect the way it performs, as
|
|
|
|
|
well as having several new commands. Please see the External IVR page on the Asterisk
|
|
|
|
|
wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
|
|
|
|
|
* Added ability to communicate over a TCP socket instead of forking a child process for the
|
|
|
|
|
* Added ability to communicate over a TCP socket instead of forking a child process for the
|
|
|
|
|
ExternalIVR application.
|
|
|
|
|
* ChanIsAvail has a new option, 'a', which will return all available channels instead
|
|
|
|
|
of just the first one if you give the function more then one channel to check.
|
|
|
|
|
* PrivacyManager now takes an option where you can specify a context where the
|
|
|
|
|
* PrivacyManager now takes an option where you can specify a context where the
|
|
|
|
|
given number will be matched. This way you have more control over who is allowed
|
|
|
|
|
and it stops the people who blindly enter 10 digits.
|
|
|
|
|
* ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
|
|
|
|
|
@ -1770,10 +1772,10 @@ Application Changes
|
|
|
|
|
SIP Changes
|
|
|
|
|
-----------
|
|
|
|
|
* Added DNS manager support to registrations for peers referencing peer entries.
|
|
|
|
|
DNS manager runs in the background which allows DNS lookups to be run asynchronously
|
|
|
|
|
DNS manager runs in the background which allows DNS lookups to be run asynchronously
|
|
|
|
|
as well as periodically updating the IP address. These properties allow for
|
|
|
|
|
better performance as well as recovery in the event of an IP change.
|
|
|
|
|
* Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
|
|
|
|
|
* Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
|
|
|
|
|
load/reload of large numbers of peers/users by ~40x (for large lists of peers).
|
|
|
|
|
These changes also provide performance improvements for call setup and tear down.
|
|
|
|
|
* Added ability to specify registration expiry time on a per registration basis in
|
|
|
|
|
@ -1783,8 +1785,8 @@ SIP Changes
|
|
|
|
|
* Added t38pt_usertpsource option. See sip.conf.sample for details.
|
|
|
|
|
* Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
|
|
|
|
|
* 'sip show peers' and 'sip show users' display their entries sorted in
|
|
|
|
|
alphabetical order, as opposed to the order they were in, in the config
|
|
|
|
|
file or database.
|
|
|
|
|
alphabetical order, as opposed to the order they were in, in the config
|
|
|
|
|
file or database.
|
|
|
|
|
* Videosupport now supports an additional option, "always", which always sets
|
|
|
|
|
up video RTP ports, even on clients that don't support it. This helps with
|
|
|
|
|
callfiles and certain transfers to ensure that if two video phones are
|
|
|
|
|
@ -1868,11 +1870,11 @@ Miscellaneous
|
|
|
|
|
operator. This is most helpful when working with long SQL queries in
|
|
|
|
|
func_odbc.conf, as the queries no longer need to be specified on a single
|
|
|
|
|
line.
|
|
|
|
|
* CDR config file, cdr.conf, has an added option, "initiatedseconds",
|
|
|
|
|
* CDR config file, cdr.conf, has an added option, "initiatedseconds",
|
|
|
|
|
which will add a second to the billsec when the ending
|
|
|
|
|
time is set, if the number in the microseconds field of the end time is
|
|
|
|
|
time is set, if the number in the microseconds field of the end time is
|
|
|
|
|
greater than the number of microseconds in the answer time. This allows
|
|
|
|
|
users to count the 'initiated' seconds in their billing records.
|
|
|
|
|
users to count the 'initiated' seconds in their billing records.
|
|
|
|
|
|
|
|
|
|
------------------------------------------------------------------------------
|
|
|
|
|
--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
|
|
|
|
|
@ -1884,7 +1886,7 @@ AMI - The manager (TCP/TLS/HTTP)
|
|
|
|
|
on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
|
|
|
|
|
* Manager version has changed to 1.1
|
|
|
|
|
* Added a new action 'CoreShowChannels' to list currently defined channels
|
|
|
|
|
and some information about them.
|
|
|
|
|
and some information about them.
|
|
|
|
|
* Added a new action 'SIPshowregistry' to list SIP registrations.
|
|
|
|
|
* Added TLS support for the manager interface and HTTP server
|
|
|
|
|
* Added the URI redirect option for the built-in HTTP server
|
|
|
|
|
@ -1896,7 +1898,7 @@ AMI - The manager (TCP/TLS/HTTP)
|
|
|
|
|
Asterisk configuration file in JSON format. This is intended to help
|
|
|
|
|
improve the performance of AJAX applications using the manager interface
|
|
|
|
|
over HTTP.
|
|
|
|
|
* SIP and IAX manager events now use "ChannelType" in all cases where we
|
|
|
|
|
* SIP and IAX manager events now use "ChannelType" in all cases where we
|
|
|
|
|
indicate channel driver. Previously, we used a mixture of "Channel"
|
|
|
|
|
and "ChannelDriver" headers.
|
|
|
|
|
* Added a "Bridge" action which allows you to bridge any two channels that
|
|
|
|
|
@ -1927,7 +1929,7 @@ AMI - The manager (TCP/TLS/HTTP)
|
|
|
|
|
* Originate now requires the Originate privilege and, if you want to call out
|
|
|
|
|
to a subshell, it requires the System privilege, as well. This was done to
|
|
|
|
|
enhance manager security.
|
|
|
|
|
* Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
|
|
|
|
|
* Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
|
|
|
|
|
* New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
|
|
|
|
|
or manager show command Atxfer from the CLI
|
|
|
|
|
* New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
|
|
|
|
|
@ -1949,7 +1951,7 @@ Dialplan functions
|
|
|
|
|
held for any given channel. Also, locks are automatically freed when a
|
|
|
|
|
channel is hung up.
|
|
|
|
|
* Added HINT() dialplan function that allows retrieving hint information.
|
|
|
|
|
Hints are mappings between extensions and devices for the sake of
|
|
|
|
|
Hints are mappings between extensions and devices for the sake of
|
|
|
|
|
determining the state of an extension. This function can retrieve the list
|
|
|
|
|
of devices or the name associated with a hint.
|
|
|
|
|
* Added EXTENSION_STATE() dialplan function which allows retrieving the state
|
|
|
|
|
@ -1999,7 +2001,7 @@ SIP changes
|
|
|
|
|
-----------
|
|
|
|
|
* Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
|
|
|
|
|
option is enabled, Asterisk will watch for a CNG tone in the incoming audio
|
|
|
|
|
for a received call. If it is detected, the channel will jump to the
|
|
|
|
|
for a received call. If it is detected, the channel will jump to the
|
|
|
|
|
'fax' extension in the dialplan.
|
|
|
|
|
* The default SIP useragent= identifier now includes the Asterisk version
|
|
|
|
|
* A new option, match_auth_username in sip.conf changes the matching of incoming requests.
|
|
|
|
|
@ -2020,8 +2022,8 @@ SIP changes
|
|
|
|
|
* The SIPPEER function have new options for port address, call and pickup groups
|
|
|
|
|
* Added support for T.140 realtime text in SIP/RTP
|
|
|
|
|
* The "checkmwi" option has been removed from sip.conf, as it is no longer
|
|
|
|
|
required due to the restructuring of how MWI is handled. See the descriptions
|
|
|
|
|
in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
|
|
|
|
|
required due to the restructuring of how MWI is handled. See the descriptions
|
|
|
|
|
in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
|
|
|
|
|
for more information.
|
|
|
|
|
* Added rtpdest option to CHANNEL() dialplan function.
|
|
|
|
|
* Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
|
|
|
|
|
@ -2029,11 +2031,11 @@ SIP changes
|
|
|
|
|
in the same dial command, or if the new c option in dial() is used.
|
|
|
|
|
* The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
|
|
|
|
|
states it is not needed. For phones, however, that do require it the "registertrying" option
|
|
|
|
|
has been added so it can be enabled.
|
|
|
|
|
has been added so it can be enabled.
|
|
|
|
|
* A new option called "callcounter" (global/peer/user level) enables call counters needed
|
|
|
|
|
for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
|
|
|
|
|
used to enable this functionality).
|
|
|
|
|
* New settings for timer T1 and timer B on a global level or per device. This makes it
|
|
|
|
|
* New settings for timer T1 and timer B on a global level or per device. This makes it
|
|
|
|
|
possible to force timeout faster on non-responsive SIP servers. These settings are
|
|
|
|
|
considered advanced, so don't use them unless you have a problem.
|
|
|
|
|
* Added a dial string option to be able to set the To: header in an INVITE to any
|
|
|
|
|
@ -2046,7 +2048,7 @@ SIP changes
|
|
|
|
|
* Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
|
|
|
|
|
and configs/sip.conf.sample for more information on how it is used.
|
|
|
|
|
* Added a new configuration option "authfailureevents" that enables manager events when
|
|
|
|
|
a peer can't authenticate properly.
|
|
|
|
|
a peer can't authenticate properly.
|
|
|
|
|
* Added DNS manager support to registrations for peers not referencing a peer entry.
|
|
|
|
|
|
|
|
|
|
IAX2 changes
|
|
|
|
|
@ -2143,7 +2145,7 @@ New Channel Drivers
|
|
|
|
|
work with Mac CoreAudio, but portaudio supports a number of other audio
|
|
|
|
|
interfaces, as well. Note that this channel driver requires v19 or higher
|
|
|
|
|
of portaudio; older versions have a different API.
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
DUNDi changes
|
|
|
|
|
-------------
|
|
|
|
|
* Added the ability to specify arguments to the Dial application when using
|
|
|
|
|
@ -2210,15 +2212,15 @@ Queue changes
|
|
|
|
|
-------------
|
|
|
|
|
* Added the general option 'shared_lastcall' so that member's wrapuptime may be
|
|
|
|
|
used across multiple queues.
|
|
|
|
|
* Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
|
|
|
|
|
* Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
|
|
|
|
|
setqueueentryvar options for each queue, see queues.conf.sample for details.
|
|
|
|
|
* Added keepstats option to queues.conf which will keep queue
|
|
|
|
|
statistics during a reload.
|
|
|
|
|
* setinterfacevar option in queues.conf also now sets a variable
|
|
|
|
|
called MEMBERNAME which contains the member's name.
|
|
|
|
|
* Added 'Strategy' field to manager event QueueParams which represents
|
|
|
|
|
the queue strategy in use.
|
|
|
|
|
* Added option to run macro when a queue member is connected to a caller,
|
|
|
|
|
the queue strategy in use.
|
|
|
|
|
* Added option to run macro when a queue member is connected to a caller,
|
|
|
|
|
see queues.conf.sample for details.
|
|
|
|
|
* app_queue now has a 'loose' option which is almost exactly like 'strict' except it
|
|
|
|
|
does not count paused queue members as unavailable.
|
|
|
|
|
@ -2264,7 +2266,7 @@ Queue changes
|
|
|
|
|
|
|
|
|
|
MeetMe Changes
|
|
|
|
|
--------------
|
|
|
|
|
* The 'o' option to provide an optimization has been removed and its functionality
|
|
|
|
|
* The 'o' option to provide an optimization has been removed and its functionality
|
|
|
|
|
has been enabled by default.
|
|
|
|
|
* When a conference is created, the UNIQUEID of the channel that caused it to be
|
|
|
|
|
created is stored. Then, every channel that joins the conference will have the
|
|
|
|
|
@ -2292,7 +2294,7 @@ MeetMe Changes
|
|
|
|
|
conference when there is only one member and the M option is used.
|
|
|
|
|
* Added MEETME_INFO dialplan function which provides a way to query
|
|
|
|
|
various properties of a Meetme conference.
|
|
|
|
|
* Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
|
|
|
|
|
* Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
|
|
|
|
|
and *84: record in-conf
|
|
|
|
|
|
|
|
|
|
Other Dialplan Application Changes
|
|
|
|
|
@ -2331,7 +2333,7 @@ Other Dialplan Application Changes
|
|
|
|
|
|
|
|
|
|
Music On Hold Changes
|
|
|
|
|
---------------------
|
|
|
|
|
* A new option, "digit", has been added for music on hold classes in
|
|
|
|
|
* A new option, "digit", has been added for music on hold classes in
|
|
|
|
|
musiconhold.conf. If this is set for a music on hold class, a caller
|
|
|
|
|
listening to music on hold can press this digit to switch to listening
|
|
|
|
|
to this music on hold class.
|
|
|
|
|
@ -2344,7 +2346,7 @@ AEL Changes
|
|
|
|
|
-----------
|
|
|
|
|
* AEL upgraded to use the Gosub with Arguments instead
|
|
|
|
|
of Macro application, to hopefully reduce the problems
|
|
|
|
|
seen with the artificially low stack ceiling that
|
|
|
|
|
seen with the artificially low stack ceiling that
|
|
|
|
|
Macro bumps into. Macros can only call other Macros
|
|
|
|
|
to a depth of 7. Tests run using gosub, show depths
|
|
|
|
|
limited only by virtual memory. A small test demonstrated
|
|
|
|
|
@ -2360,17 +2362,17 @@ AEL Changes
|
|
|
|
|
fashion: Set(LOCAL(myvar)=someval); ("local" is now
|
|
|
|
|
an AEL keyword).
|
|
|
|
|
* utils/conf2ael introduced. Will convert an extensions.conf
|
|
|
|
|
file into extensions.ael. Very crude and unfinished, but
|
|
|
|
|
file into extensions.ael. Very crude and unfinished, but
|
|
|
|
|
will be improved as time goes by. Should be useful for a
|
|
|
|
|
first pass at conversion.
|
|
|
|
|
* aelparse will now read extensions.conf to see if a referenced
|
|
|
|
|
macro or context is there before issueing a warning.
|
|
|
|
|
* AEL parser sets a local channel variable ~~EXTEN~~, to
|
|
|
|
|
* AEL parser sets a local channel variable ~~EXTEN~~, to
|
|
|
|
|
preserve the value of ${EXTEN} thru switch statements.
|
|
|
|
|
* New operator in $[...] expressions: the ~~ operator serves
|
|
|
|
|
as a concatenation operator. AT THE MOMENT, it is really only
|
|
|
|
|
necessary and useful in AEL, especially in if() expressions.
|
|
|
|
|
Operation: ${a} ~~ ${b| with force both a and b to strings, strip
|
|
|
|
|
Operation: ${a} ~~ ${b| with force both a and b to strings, strip
|
|
|
|
|
any enclosing double-quotes, and evaluate to the value of a
|
|
|
|
|
concatenated with the value of b. For example if a is set to
|
|
|
|
|
"xyz" and b has the value "abc", then ${a} ~~ ${b| would
|
|
|
|
|
@ -2438,7 +2440,7 @@ Logger changes
|
|
|
|
|
and to ensure that the oldest log file gets deleted.
|
|
|
|
|
* Added realtime support for the queue log
|
|
|
|
|
|
|
|
|
|
Call Detail Records
|
|
|
|
|
Call Detail Records
|
|
|
|
|
-------------------
|
|
|
|
|
* The cdr_manager module has a [mappings] feature, like cdr_custom,
|
|
|
|
|
to add fields to the manager event from the CDR variables.
|
|
|
|
|
@ -2499,7 +2501,7 @@ Miscellaneous New Modules
|
|
|
|
|
* Added support for writing and running your dialplan in lua using the pbx_lua
|
|
|
|
|
module. See configs/extensions.lua.sample for examples of how to do this.
|
|
|
|
|
|
|
|
|
|
Miscellaneous
|
|
|
|
|
Miscellaneous
|
|
|
|
|
-------------
|
|
|
|
|
* Ability to use libcap to set high ToS bits when non-root
|
|
|
|
|
on Linux. If configure is unable to find libcap then you
|
|
|
|
|
@ -2547,7 +2549,7 @@ Miscellaneous
|
|
|
|
|
turned on, via the CHANNEL(trace) dialplan function. Could be useful for
|
|
|
|
|
dialplan debugging.
|
|
|
|
|
* iLBC source code no longer included (see UPGRADE.txt for details)
|
|
|
|
|
* If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
|
|
|
|
|
* If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
|
|
|
|
|
deadlock is detected, a backtrace of the stack which led to the lock calls
|
|
|
|
|
will be output to the CLI.
|
|
|
|
|
* If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
|
|
|
|
|
|
Kevin P. Fleming
13 years ago