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------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
------------------------------------------------------------------------------
Device State Handling
---------------------
* The event infrastructure in Asterisk got another big update to help support
distributed events. It currently supports distributed device state and
distributed Voicemail MWI (Message Waiting Indication). A new module has
been merged, res_ais, which facilitates communicating events between servers.
It uses the SAForum AIS (Service Availability Forum Application Interface
Specification) CLM (Cluster Management) and EVT (Event) services to maintain
a cluster of Asterisk servers, and to share events between them. For more
information on setting this up, see doc/distributed_devstate.txt.
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
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
forwarding station ID. However, there are new options that allow you to search
based on the message desk terminal ID, or the message desk number.
* TIMEOUT() has been modified to be accurate down to the millisecond.
* ENUM*() functions now include the following new options:
- 'u' returns the full URI and does not strip off the URI-scheme.
- 's' triggers ISN specific rewriting
- 'i' looks for branches into an Infrastructure ENUM tree
- 'd' for a direct DNS lookup without any flipping of digits.
* TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
* CHANNEL() now has options for the maximum, minimum, and standard or normal
deviation of jitter, rtt, and loss for a call using chan_sip.
DAHDI channel driver (chan_dahdi) Changes
----------------------------------------
* Channels can now be configured using named sections in chan_dahdi.conf, just
like other channel drivers, including the use of templates.
* The default for pridialplan has changed from 'national' to 'unknown'.
PBX Changes
-----------
* It is now possible to specify a pattern match as a hint. Once a phone subscribes
to something that matches the pattern a hint will be created using the contents
and variables evaluated.
* Dialplan matching has been extended to allow an extension to return to the
PBX core to wait for more digits. This is done by using the new dialplan
application called "Incomplete". This will permit a whole new level of
extension control, by giving the administrator more control over early
matches employing one of the short-circuit pattern match operators. Note
that custom applications can trigger this same behavior by returning the
special value AST_PBX_INCOMPLETE.
Application Changes
-------------------
* Directory now permits both first and last names to be matched at the same
time. In addition, the number of digits to enter of the name can be set in
the arguments to Directory; previously, you could enter only 3, regardless
of how many names are in your company. For large companies, this should be
quite helpful.
* Voicemail now permits a mailbox setting to wrap around from first to last
messages, if the "messagewrap" option is set to a true value.
* Voicemail now permits an external script to be run, for password validation.
The script should output "VALID" or "INVALID" on stdout, depending upon the
wish to validate or invalidate the password given. Arguments are:
"mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
more details
* Dial has a new option: F(context^extension^pri), which permits a callee to
continue in the dialplan, at the specified label, if the caller hangs up.
* ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
technology name (e.g. SIP, IAX, etc) of the channel being spied on.
* The Jack application now has a c() option to supply a custom client name.
* Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
like the pre-existing whisper mode, except that the spy can also talk to the
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
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
words, if using the 'd' option, it is not possible to enter a number to append to
the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
change to whisper mode, and pressing 6 will change to barge mode.
* ExternalIVR now takes several options that affect the way it performs, as
well as having several new commands. Please see doc/externalivr.txt for the
complete documentation.
* 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
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
answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
* The Dial() application no longer copies the language used by the caller to the callee's
channel. If you desire for the caller's channel's language to be used for file playback
to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
SIP Changes
-----------
* The ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using setvar to cause a given
audio file to be played upon completion of an attended transfer.
* 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
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
load/reload of large numbers of peers/users by ~40x (for large lists of peers.
Initially, we saw 4x improvement in call setup/destruction, but at the time
of merging, this gain has disappeared; further research will be done to try
and restore this performance improvement. Astobj2 refcounting is now used
for users, peers, and dialogs. Users are encouraged to assist in regression
testing and problem reporting!
* Added ability to specify registration expiry time on a per registration basis in
the register line.
* Added support for T140 RED - redundancy in T.140 to prevent text loss due to
lost packets.
* 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.
IAX Changes
-----------
* Existing DNS manager lookups extended to check for SRV records.
CLI Changes
-----------
* New CLI command, "config reload <file.conf>" which reloads any module that
references that particular configuration file. Also added "config list"
which shows which configuration files are in use.
* New CLI commands, "pri show version" and "ss7 show version" that will
display which version of libpri and libss7 are being used, respectively.
A new API call was added so trunk will now have to be compiled against
a versions of libpri and libss7 that have them or it will not know that
these libraries exist.
DNS manager changes
-------------------
* Addresses managed by DNS manager now can check to see if there is a DNS
SRV record for a given domain and will use that hostname/port if present.
AMI - The manager (TCP/TLS/HTTP)
--------------------------------
* The Status command now takes an optional list of variables to display
along with channel status.
ODBC Changes
------------
* res_odbc no longer has a limit of 1023 total possible unshared connections,
as some people were running into this limit. This limit has been increased
to 4.2 billion.
Queue changes
-------------
* The TRANSFER queue log entry now includes the the caller's original
position in the transferred-from queue.
* A new configuration option, "timeoutpriority" has been added. Please see the section labeled
"QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
as well as an explanation about timeout options in general
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
------------------------------------------------------------------------------
AMI - The manager (TCP/TLS/HTTP)
--------------------------------
* Manager has undergone a lot of changes, all of them documented
in doc/manager_1_1.txt
* Manager version has changed to 1.1
* Added a new action 'CoreShowChannels' to list currently defined channels
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
* The output of CallerID in Manager events is now more consistent.
CallerIDNum is used for number and CallerIDName for name.
* Enable https support for builtin web server.
See configs/http.conf.sample for details.
* Added a new action, GetConfigJSON, which can return the contents of an
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
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
are currently active on the system.
* Added a "ListAllVoicemailUsers" action that allows you to get a list of all
the voicemail users setup.
* Added 'DBDel' and 'DBDelTree' manager commands.
* cdr_manager now reports events via the "cdr" level, separating it from
the very verbose "call" level.
* Manager users are now stored in memory. If you change the manager account
list (delete or add accounts) you need to reload manager.
* Added Masquerade manager event for when a masquerade happens between
two channels.
* Added "manager reload" command for the CLI
* Lots of commands that only provided information are now allowed under the
Reporting privilege, instead of only under Call or System.
* The IAX* commands now require either System or Reporting privilege, to
mirror the privileges of the SIP* commands.
* Added ability to retrieve list of categories in a config file.
* Added ability to retrieve the content of a particular category.
* Added ability to empty a context.
* Created new action to create a new file.
* Updated delete action to allow deletion by line number with respect to category.
* Added new action insert to add new variable to category at specified line.
* Updated action newcat to allow new category to be inserted in file above another
existing category.
* Added new event "JitterBufStats" in the IAX2 channel
* 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"
* New command: Atxfer. See doc/manager_1_1.txt for more details or
manager show command Atxfer from the CLI
Dialplan functions
------------------
* Added the DEVICE_STATE() dialplan function which allows retrieving any device
state in the dialplan, as well as creating custom device states that are
controllable from the dialplan.
* Extend CALLERID() function with "pres" and "ton" parameters to
fetch string representation of calling number presentation indicator
and numeric representation of type of calling number value.
* MailboxExists converted to dialplan function
* A new option to Dial() for telling IP phones not to count the call
as "missed" when dial times out and cancels.
* Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
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
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
of any extension.
* Added SYSINFO() dialplan function which allows retrieval of system information
* Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
the existence of a dialplan target.
* Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
upper and lower case, respectively.
* When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
ID for the call (not the Asterisk call ID or unique ID), provided that the
channel driver supports this. For SIP, you get the SIP call-ID for the
bridged channel which you can store in the CDR with a custom field.
CLI Changes
-----------
* New CLI command "core show hint" (usage: core show hint <exten>)
* New CLI command "core show settings"
* Added 'core show channels count' CLI command.
* Added the ability to set the core debug and verbose values on a per-file basis.
* Added 'queue pause member' and 'queue unpause member' CLI commands
* Ability to set process limits ("ulimit") without restarting Asterisk
* Enhanced "agi debug" to print the channel name as a prefix to the debug
output to make debugging on busy systems much easier.
* New CLI commands "dialplan set extenpatternmatching true/false"
* New CLI command: "core set chanvar" to set a channel variable from the CLI.
* Added an easy way to execute Asterisk CLI commands at startup. Any commands
listed in the startup_commands section of cli.conf will get executed.
* Added a CLI command, "devstate change", which allows you to set custom device
states from the func_devstate module that provides the DEVICE_STATE() function
and handling of the "Custom:" devices.
* New CLI command: "sip show sched" which shows all ast_sched entries for sip,
sorted into the different possible callbacks, with the number of entries
currently scheduled for each. Gives you a feel for how busy the sip channel
driver is.
SIP changes
-----------
* Improved NAT and STUN support.
chan_sip now can use port numbers in bindaddr, externip and externhost
options, as well as contact a STUN server to detect its external address
for the SIP socket. See sip.conf.sample, 'NAT' section.
* 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.
If set, and the incoming request carries authentication info,
the username to match in the users list is taken from the Digest header
rather than from the From: field. This feature is considered experimental.
* The "musiconhold" and "musicclass" settings in sip.conf are now removed,
since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
* The "localmask" setting was removed in version 1.2 and the reminder about it
being removed is now also removed.
* A new option "busylevel" for setting a level of calls where asterisk reports
a device as busy, to separate it from call-limit. This value is also added
to the SIP_PEER dialplan function.
* A new realtime family called "sipregs" is now supported to store SIP registration
data. If this family is defined, "sippeers" will be used for configuration and
"sipregs" for registrations. If it's not defined, "sippeers" will be used for
registration data, as before.
* 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
for more information.
* Added rtpdest option to CHANNEL() dialplan function.
* Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
* SIP now adds a header to the CANCEL if the call was answered by another phone
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.
* 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
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
SIP uri.
* Added a new global and per-peer option, qualifyfreq, which allows you to configure
the qualify frequency.
* Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
were not properly torn down due to network or endpoint failures during an established
SIP session.
* Added experimental TCP and TLS support for SIP. See doc/siptls.txt 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.
* Added DNS manager support to registrations for peers not referencing a peer entry.
IAX2 changes
------------
* Added the trunkmaxsize configuration option to chan_iax2.
* Added the srvlookup option to iax.conf
* Added support for OSP. The token is set and retrieved through the CHANNEL()
dialplan function.
XMPP Google Talk/Jingle changes
-------------------------------
* Added the bindaddr option to gtalk.conf.
Skinny changes
-------------
* Added skinny show device, skinny show line, and skinny show settings CLI commands.
* Proper codec support in chan_skinny.
* Added settings for IP and Ethernet QoS requests
MGCP changes
------------
* Added separate settings for media QoS in mgcp.conf
Console Channel Driver changes
------------------------------
* Added experimental support for video send & receive to chan_oss.
This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
a video source.
Phone channel changes (chan_phone)
----------------------------------
* Added G729 passthrough support to chan_phone for Sigma Designs boards.
H.323 channel Changes
---------------------
* H323 remote hold notification support added (by NOTIFY message
and/or H.450 supplementary service)
Local channel changes
---------------------
* The device state functionality in the Local channel driver has been updated
to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
to just UNKNOWN if the extension exists.
* Added jitterbuffer support for chan_local. This allows you to use the
generic jitterbuffer on incoming calls going to Asterisk applications.
For example, this would allow you to use a jitterbuffer for an incoming
SIP call to Voicemail by putting a Local channel in the middle. This
feature is enabled by using the 'j' option in the Dial string to the Local
channel in conjunction with the existing 'n' option for local channels.
* A 'b' option has been added which causes chan_local to return the actual channel
that is behind it when queried. This is useful for transfer scenarios as the
actual channel will be transferred, not the Local channel.
Agent channel changes
----------------------
* The ackcall and endcall options are now supplemented with options acceptdtmf
and enddtmf. These allow for the DTMF keypress to be configurable. The options
default to their old hard-coded values ('#' and '*' respectively) so this should
not break any existing agent installations.
DAHDI channel driver (chan_dahdi) Changes
----------------------------------------
* SS7 support (via libss7 library)
* In India, some carriers transmit CID via dtmf. Some code has been added
that will handle some situations. The cidstart=polarity_IN choice has been added for
those carriers that transmit CID via dtmf after a polarity change.
* CID matching information is now shown when doing 'dialplan show'.
* Added dahdi show version CLI command.
* Added setvar support to chan_dahdi.conf channel entries.
* Added two new options: mwimonitor and mwimonitornotify. These options allow
you to enable MWI monitoring on FXO lines. When the MWI state changes,
the script specified in the mwimonitornotify option is executed. An internal
event indicating the new state of the mailbox is also generated, so that
the normal MWI facilities in Asterisk work as usual.
* Added signalling type 'auto', which attempts to use the same signalling type
for a channel as configured in DAHDI. This is primarily designed for analog
ports, but will also work for digital ports that are configured for FXS or FXO
signalling types. This mode is also the default now, so if your chan_dahdi.conf
does not specify signalling for a channel (which is unlikely as the sample
configuration file has always recommended specifying it for every channel) then
the 'auto' mode will be used for that channel if possible.
* Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
state for a channel; also ensured that the DNDState Manager event is
emitted no matter how the DND state is set or cleared.
New Channel Drivers
-------------------
* Added a new channel driver, chan_unistim. See doc/unistim.txt and
configs/unistim.conf.sample for details. This new channel driver allows
you to use Nortel i2002, i2004, and i2050 phones with Asterisk.
* Added a new channel driver, chan_console, which uses portaudio as a cross
platform audio interface. It was written as a channel driver that would
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
the DUNDi switch in the dialplan.
* Added the ability to set weights for responses dynamically. This can be
done using a global variable or a dialplan function. Using the SHELL()
function would allow you to have an external script set the weight for
each response.
* Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
functions will allow you to initiate a DUNDi query from the dialplan,
find out how many results there are, and access each one.
ENUM changes
------------
* Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
functions will allow you to initiate an ENUM lookup from the dialplan,
and Asterisk will cache the results. ENUMRESULT can be used to access
the results without doing multiple DNS queries.
Voicemail Changes
-----------------
* Added the ability to customize which sound files are used for some of the
prompts within the Voicemail application by changing them in voicemail.conf
* Added the ability for the "voicemail show users" CLI command to show users
configured by the dynamic realtime configuration method.
* MWI (Message Waiting Indication) handling has been significantly
restructured internally to Asterisk. It is now totally event based
instead of polling based. The voicemail application will notify other
modules that have subscribed to MWI events when something in the mailbox
changes.
This also means that if any other entity outside of Asterisk is changing
the contents of mailboxes, then the voicemail application still needs to
poll for changes. Examples of situations that would require this option
are web interfaces to voicemail or an email client in the case of using
IMAP storage. So, two new options have been added to voicemail.conf
to account for this: "pollmailboxes" and "pollfreq". See the sample
configuration file for details.
* Added "tw" language support
* Added support for storage of greetings using an IMAP server
* Added ability to customize forward, reverse, stop, and pause keys for message playback
* SMDI is now enabled in voicemail using the smdienable option.
* A "lockmode" option has been added to asterisk.conf to configure the file
locking method used for voicemail, and potentially other things in the
future. The default is the old behavior, lockfile. However, there is a
new method, "flock", that uses a different method for situations where the
lockfile will not work, such as on SMB/CIFS mounts.
* Added the ability to backup deleted messages, to ease recovery in the case
that a user accidentally deletes a message, and discovers that they need it.
* Reworked the SMDI interface in Asterisk. The new way to access SMDI information
is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
voicemail boxes. The SMDI interface can also poll for MWI changes when some
outside entity is modifying the state of the mailbox (such as IMAP storage or
a web interface of some kind).
* Added the support for marking messages as "urgent." There are two methods to accomplish
this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
the message as urgent after he has recorded a voicemail by following the voice instructions.
When listening to voicemails using VoiceMailMain urgent messages will be presented before other
messages
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
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,
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.
* Added min-announce-frequency option to queues.conf which allows you to control the
minimum amount of time between queue announcements for use when the caller's queue
position changes frequently.
* Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
queue log.
* Added ability for non-realtime queues to have realtime members
* Added the "linear" strategy to queues.
* Added the "wrandom" strategy to queues.
* Added new channel variable QUEUE_MIN_PENALTY
* QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
rules in queuerules.conf. See configs/queuerules.conf.sample for details
* Added a new parameter for member definition, called state_interface. This may be
used so that a member may be called via one interface but have a different interface's
device state reported.
* New configuration option: randomperiodicannounce. If a list of periodic announcements is
specified by the periodic-announce option, then one will be chosen randomly when it is time
to play a periodic announcment
* New configuration options: announce-position now takes two more values in addition to "yes" and
"no." Two new options, "limit" and "more," are allowed. These are tied to another option,
announce-position-limit. By setting announce-position to "limit" callers will only have their
position announced if their position is less than what is specified by announce-position-limit.
If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
will be told that their are more than announce-position-limit callers waiting.
* Two new queue log events have been added. An ADDMEMBER event will be logged
when a realtime queue member is added and a REMOVEMEMBER event will be logged
when a realtime queue member is removed. Since there is no calling channel associated
with these events, the string "REALTIME" is placed where the channel's unique id
is typically placed.
MeetMe Changes
--------------
* 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
MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
callers that come and go from long standing conferences.
* Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
except it does operations on a channel by name, instead of number in a conference.
This is a very useful feature in combination with the 'X' option to ChanSpy.
* Added 'C' option to Meetme which causes a caller to continue in the dialplan
when kicked out.
* Added new RealTime functionality to provide support for scheduled conferencing.
This includes optional messages to the caller if they attempt to join before
the schedule start time, or to allow the caller to join the conference early.
Also included is optional support for limiting the number of callers per
RealTime conference.
* Added the S() and L() options to the MeetMe application. These are pretty
much identical to the S() and L() options to Dial(). They let you set
timeouts for the conference, as well as have warning sounds played to
let the caller know how much time is left, and when it is running out.
* Added the ability to do "meetme concise" with the "meetme" CLI command.
This extends the concise capabilities of this CLI command to include
listing all conferences, instead of an addition to the other sub commands
for the "meetme" command.
* Added the ability to specify the music on hold class used to play into the
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.
Other Dialplan Application Changes
----------------------------------
* Argument support for Gosub application
* From the to-do lists: straighten out the app timeout args:
Wait() app now really does 0.3 seconds- was truncating arg to an int.
WaitExten() same as Wait().
Congestion() - Now takes floating pt. argument.
Busy() - now takes floating pt. argument.
Read() - timeout now can be floating pt.
WaitForRing() now takes floating pt timeout arg.
SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
* Added 's' option to Page application.
* Added 'E' and 'V' commands to ExternalIVR.
* Added 'o' and 'X' options to Chanspy.
* Added a new dialplan application, Bridge, which allows you to bridge the
calling channel to any other active channel on the system.
* Added the ability to specify a music on hold class to play instead of ringing
for the SLATrunk application.
* The Read application no longer exits the dialplan on error. Instead, it sets
READSTATUS to ERROR, which you can catch and handle separately.
* Added 'm' option to Directory, which lists out names, 8 at a time, instead
of asking for verification of each name, one at a time.
* Privacy() no longer uses privacy.conf, as all options are specifyable as
direct options to the app.
* AMD() has a new "maximum word length" option. "show application AMD" from the CLI
for more details
* GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
* The ChannelRedirect application no longer exits the dialplan if the given channel
does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
or NOCHANNEL if the given channel was not found.
* The silencethreshold setting that was previously configurable in multiple
applications is now settable globally via dsp.conf.
* Added ability to communicate over a TCP socket instead of forking a child process for the
ExternalIVR application.
Music On Hold Changes
---------------------
* 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.
* Support for realtime music on hold has been added.
* In conjunction with the realtime music on hold, a general section has
been added to musiconhold.conf, its sole variable is cachertclasses. If this
is set, then music on hold classes found in realtime will be cached in memory.
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
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
recursive call depths of 100,000 without problems.
-- in addition to this, all apps that allowed a macro
to be called, as in Dial, queues, etc, are now allowing
a gosub call in similar fashion.
* AEL now generates LOCAL(argname) declarations when it
Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
etc. That makes the arguments local in scope. The user
can define their own local variables in macros, now,
by saying "local myvar=someval;" or using Set() in this
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
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
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
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
evaluate to xyzabc .
Call Features (res_features) Changes
------------------------------------
* Added the parkedcalltransfers option to features.conf
* The built-in method for doing attended transfers has been updated to
include some new options that allow you to have the transferee sent
back to the person that did the transfer if the transfer is not successful.
See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
in features.conf.sample.
* Added support for configuring named groups of custom call features in
features.conf. This means that features can be written a single time, and
then mapped into groups of features for different key mappings or easier
access control.
* Updated the ParkedCall application to allow you to not specify a parking
extension. If you don't specify a parking space to pick up, it will grab
the first one available.
* Added cli command 'features reload' to reload call features from features.conf
* Moved into core asterisk binary.
Language Support Changes
------------------------
* Brazilian Portuguese (pt-BR) in VM, and say.c was added
* Added support for the Hungarian language for saying numbers, dates, and times.
AGI Changes
-----------
* Added SPEECH commands for speech recognition. A complete listing can be found
using agi show.
* If app_stack is loaded, GOSUB is a native AGI command that may be used to
invoke subroutines in the dialplan. Note that calling EXEC with Gosub
does not behave as expected; the native command needs to be used, instead.
Logger changes
--------------
* Added rotatestrategy option to logger.conf, along with two new options:
"timestamp" which will use the time to name the logger files instead of
sequence number; and "rotate", which rotates the names of the logfiles,
similar to the way syslog rotates files.
* Added exec_after_rotate option to logger.conf, which allows a system
command to be run after rotation. This is primarily useful with
rotatestrategry=rotate, to allow a limit on the number of logfiles kept
and to ensure that the oldest log file gets deleted.
* Added realtime support for the queue log
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.
* Added cdr_adaptive_odbc, a new module that adapts to the structure of your
backend database CDR table. Specifically, additional, non-standard
columns are supported, merely by setting the corresponding CDR variable in
your dialplan. In addition, you may alias any column to another name (for
example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
simply "alias src => ANI" in the configuration file). Records may be
posted to more than one backend, simply by specifying multiple categories
in the configuration file. And finally, you may filter which CDRs get
posted to each backend, by specifying a filter (which the record must
match) for the particular category. Filters are additive (meaning all
rules must match to post that CDR).
* The Postgres CDR module now supports some features of the cdr_adaptive_odbc
module. Specifically, you may add additional columns into the table and
they will be set, if you set the corresponding CDR variable name. Also,
if you omit columns in your database table, they will be silently skipped
(but a record will still be inserted, based on what columns remain). Note
that the other two features from cdr_adaptive_odbc (alias and filter) are
not currently supported.
* The ResetCDR application now has an 'e' option that re-enables a CDR if it
has been disabled using the NoCDR application.
Miscellaneous New Modules
-------------------------
* Added a new CDR module, cdr_sqlite3_custom.
* Added a new realtime configuration module, res_config_sqlite
* Added a new codec translation module, codec_resample, which re-samples
signed linear audio between 8 kHz and 16 kHz to help support wideband
codecs.
* Added a new module, res_phoneprov, which allows auto-provisioning of phones
based on configuration templates that use Asterisk dialplan function and
variable substitution. It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.
* Added a new module, app_jack, which provides interfaces to JACK, the Jack
Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
provided; there is a JACK() application, and a JACK_HOOK() function. Both
interfaces create an input and output JACK port. The application makes
these ports the endpoint of the call. The audio coming from the channel
goes out the output port and whatever comes back in on the input port is
what gets sent to the channel. The JACK_HOOK() function turns on a JACK
audiohook on the channel. This lets you run the audio coming from a
channel through JACK, and whatever comes back in is what gets forwarded
on as the channel's audio. This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
application.
* Added a new module, res_config_curl, which permits using a HTTP POST url
to retrieve, create, update, and delete realtime information from a remote
web server. Note that this module requires func_curl.so to be loaded for
backend functionality.
* Added a new module, res_config_ldap, which permits the use of an LDAP
server for realtime data access.
* 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
-------------
* Ability to use libcap to set high ToS bits when non-root
on Linux. If configure is unable to find libcap then you
can use --with-cap to specify the path.
* Added maxfiles option to options section of asterisk.conf which allows you to specify
what Asterisk should set as the maximum number of open files when it loads.
* Added the jittertargetextra configuration option.
* Added support for setting the CoS for VLAN traffic (802.1p). See the sample
configuration files for the IP channel drivers. The new option is "cos".
This information is also documented in doc/qos.tex, or the IP Quality of Service
section of asterisk.pdf.
* When originating a call using AMI or pbx_spool that fails the reason for failure
will now be available in the failed extension using the REASON dialplan variable.
* Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
It allows you to configure a prefix for auto-monitor recordings.
* A new extension pattern matching algorithm, based on a trie, is introduced
here, that could noticeably speed up mid-sized to large dialplans.
It is NOT used by default, as duplicating the behaviour of the old pattern
matcher is still under development. A config file option, in extensions.conf,
in the [general] section, called "extenpatternmatchingnew", is by default
set to false; setting that to true will force the use of the new algorithm.
Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
be used to switch the algorithms at run time.
* A new option when starting a remote asterisk (rasterisk, asterisk -r) for
specifying which socket to use to connect to the running Asterisk daemon
(-s)
* Performance enhancements to the sched facility, which is used in
the channel drivers, etc. Added hashtabs and doubly-linked lists
to speed up deletion; start at the beginning or end of list to
speed up insertion.
* Added Doubly-linked lists after the fashion of linkedlists.h. They are in
dlinkedlists.h. Doubly-linked lists feature fast deletion times.
Added regression tests to the tests/ dir, also.
* Added a refcount trace feature to astobj2 for those trying to balance
object creation, deletion; work, play; space and time. See the
notes in astobj2.h. Also, see utils/refcounter as well, as a
quick way to find unbalanced refcounts in what could be a sea
of objects that were balanced.
* Added logging to 'make update' command. See update.log
* Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
do not come from the remote party.
* Added the 'n' option to the SpeechBackground application to tell it to not
answer the channel if it has not already been answered.
* Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
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
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
the "core show locks" CLI command will give lock information output as well
as a backtrace of the stack which led to the lock calls.
* users.conf now sports an optional alternateexts property, which permits
allocation of additional extensions which will reach the specified user.
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