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Merge "build: Revise CHANGES and UPGRADE.txt handling."

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Benjamin Keith Ford 6 years ago committed by Gerrit Code Review
parent adc8aa8125 a4ab7f5f80
commit 7757b7d6a6
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67
CHANGES
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==============================================================================
===
=== THIS FILE IS AUTOMATICALLY GENERATED DURING THE RELEASE
=== PROCESS. DO NOT MAKE CHANGES HERE. INSTEAD, REFER TO
=== doc/CHANGES-staging/README.md FOR MORE DETAILS.
===
=== This file documents the new and/or enhanced functionality added in
=== the Asterisk versions listed below. This file does NOT include
=== changes in behavior that would not be backwards compatible with
@ -8,69 +12,6 @@
===
==============================================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16 to Asterisk 17 --------------------
------------------------------------------------------------------------------
chan_sip
------------------
* The chan_sip module is now deprecated, users should migrate to the
replacement module chan_pjsip. See guides at the Asterisk Wiki:
https://wiki.asterisk.org/wiki/x/tAHOAQ
https://wiki.asterisk.org/wiki/x/hYCLAQ
Channels
------------------
* The core no longer uses the stasis cache for channels snapshots.
The following APIs are no longer available:
ast_channel_topic_cached()
ast_channel_topic_all_cached()
The ast_channel_cache_all() and ast_channel_cache_by_name() functions
now returns an ao2_container of ast_channel_snapshots rather than a
container of stasis_messages therefore you can't call stasis_cache
functions on it.
The ast_channel_topic_all() function now returns a normal topic,
not a cached one so you can't use stasis cache functions on it either.
The ast_channel_snapshot_type() stasis message now has the
ast_channel_snapshot_update structure as it's data.
ast_channel_snapshot_get_latest() still returns the latest snapshot.
Bridging
------------------
* The bridging core no longer uses the stasis cache for bridge
snapshots. The latest bridge snapshot is now stored on the
ast_bridge structure itself.
* The following APIs are no longer available since the stasis cache
is no longer used:
ast_bridge_topic_cached()
ast_bridge_topic_all_cached()
* A topic pool is now used for individual bridge topics.
* The ast_bridge_cache() function was removed since there's no
longer a separate container of snapshots.
* A new function "ast_bridges()" was created to retrieve the
container of all bridges. Users formerly calling
ast_bridge_cache() can use the new function to iterate over
bridges and retrieve the latest snapshot directly from the
bridge.
* The ast_bridge_snapshot_get_latest() function was renamed to
ast_bridge_get_snapshot_by_uniqueid().
* A new function "ast_bridge_get_snapshot()" was created to retrieve
the bridge snapshot directly from the bridge structure.
* The ast_bridge_topic_all() function now returns a normal topic
not a cached one so you can't use stasis cache functions on it
either.
* The ast_bridge_snapshot_type() stasis message now has the
ast_bridge_snapshot_update structure as it's data. It contains
the last snapshot and the new one.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.2.0 to Asterisk 16.3.0 ----------
------------------------------------------------------------------------------

218
UPGRADE-1.2.txt
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=========================================================
===
=== Information for upgrading from Asterisk 1.0 to 1.2
===
=== This file documents all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also includes advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=========================================================
Compiling:
* The Asterisk 1.2 source code now uses C language features
supported only by 'modern' C compilers. Generally, this means GCC
version 3.0 or higher, although some GCC 2.96 releases will also
work. Some non-GCC compilers that support C99 and the common GCC
extensions (including anonymous structures and unions) will also
work. All releases of GCC 2.95 do _not_ have the requisite feature
support; systems using that compiler will need to be upgraded to
a more recent compiler release.
Dialplan Expressions:
* The dialplan expression parser (which handles $[ ... ] constructs)
has gone through a major upgrade, but has one incompatible change:
spaces are no longer required around expression operators, including
string comparisons. However, you can now use quoting to keep strings
together for comparison. For more details, please read the
doc/README.variables file, and check over your dialplan for possible
problems.
Agents:
* The default for ackcall has been changed to "no" instead of "yes"
because of a bug which caused the "yes" behavior to generally act like
"no". You may need to adjust the value if your agents behave
differently than you expect with respect to acknowledgement.
* The AgentCallBackLogin application now requires a second '|' before
specifying an extension@context. This is to distinguish the options
string from the extension, so that they do not conflict. See
'show application AgentCallbackLogin' for more details.
Parking:
* Parking behavior has changed slightly; when a parked call times out,
Asterisk will attempt to deliver the call back to the extension that
parked it, rather than the 's' extension. If that extension is busy
or unavailable, the parked call will be lost.
Dialing:
* The Caller*ID of the outbound leg is now the extension that was
called, rather than the Caller*ID of the inbound leg of the call. The
"o" flag for Dial can be used to restore the original behavior if
desired. Note that if you are looking for the originating callerid
from the manager event, there is a new manager event "Dial" which
provides the source and destination channels and callerid.
IAX:
* The naming convention for IAX channels has changed in two ways:
1. The call number follows a "-" rather than a "/" character.
2. The name of the channel has been simplified to IAX2/peer-callno,
rather than IAX2/peer@peer-callno or even IAX2/peer@peer/callno.
SIP:
* The global option "port" in 1.0.X that is used to set which port to
bind to has been changed to "bindport" to be more consistent with
the other channel drivers and to avoid confusion with the "port"
option for users/peers.
* The "Registry" event now uses "Username" rather than "User" for
consistency with IAX.
Applications:
* With the addition of dialplan functions (which operate similarly
to variables), the SetVar application has been renamed to Set.
* The CallerPres application has been removed. Use SetCallerPres
instead. It accepts both numeric and symbolic names.
* The applications GetGroupCount, GetGroupMatchCount, SetGroup, and
CheckGroup have been deprecated in favor of functions. Here is a
table of their replacements:
GetGroupCount([groupname][@category] GROUP_COUNT([groupname][@category]) Set(GROUPCOUNT=${GROUP_COUNT()})
GroupMatchCount(groupmatch[@category]) GROUP_MATCH_COUNT(groupmatch[@category]) Set(GROUPCOUNT=${GROUP_MATCH_COUNT(SIP/.*)})
SetGroup(groupname[@category]) GROUP([category])=groupname Set(GROUP()=test)
CheckGroup(max[@category]) N/A GotoIf($[ ${GROUP_COUNT()} > 5 ]?103)
Note that CheckGroup does not have a direct replacement. There is
also a new function called GROUP_LIST() which will return a space
separated list of all of the groups set on a channel. The GROUP()
function can also return the name of the group set on a channel when
used in a read environment.
* The applications DBGet and DBPut have been deprecated in favor of
functions. Here is a table of their replacements:
DBGet(foo=family/key) Set(foo=${DB(family/key)})
DBPut(family/key=${foo}) Set(DB(family/key)=${foo})
* The application SetLanguage has been deprecated in favor of the
function LANGUAGE().
SetLanguage(fr) Set(LANGUAGE()=fr)
The LANGUAGE function can also return the currently set language:
Set(MYLANG=${LANGUAGE()})
* The applications AbsoluteTimeout, DigitTimeout, and ResponseTimeout
have been deprecated in favor of the function TIMEOUT(timeouttype):
AbsoluteTimeout(300) Set(TIMEOUT(absolute)=300)
DigitTimeout(15) Set(TIMEOUT(digit)=15)
ResponseTimeout(15) Set(TIMEOUT(response)=15)
The TIMEOUT() function can also return the currently set timeouts:
Set(DTIMEOUT=${TIMEOUT(digit)})
* The applications SetCIDName, SetCIDNum, and SetRDNIS have been
deprecated in favor of the CALLERID(datatype) function:
SetCIDName(Joe Cool) Set(CALLERID(name)=Joe Cool)
SetCIDNum(2025551212) Set(CALLERID(number)=2025551212)
SetRDNIS(2024561414) Set(CALLERID(RDNIS)=2024561414)
* The application Record now uses the period to separate the filename
from the format, rather than the colon.
* The application VoiceMail now supports a 'temporary' greeting for each
mailbox. This greeting can be recorded by using option 4 in the
'mailbox options' menu, and 'change your password' option has been
moved to option 5.
* The application VoiceMailMain now only matches the 'default' context if
none is specified in the arguments. (This was the previously
documented behavior, however, we didn't follow that behavior.) The old
behavior can be restored by setting searchcontexts=yes in voicemail.conf.
Queues:
* A queue is now considered empty not only if there are no members but if
none of the members are available (e.g. agents not logged on). To
restore the original behavior, use "leavewhenempty=strict" or
"joinwhenempty=strict" instead of "=yes" for those options.
* It is now possible to use multi-digit extensions in the exit context
for a queue (although you should not have overlapping extensions,
as there is no digit timeout). This means that the EXITWITHKEY event
in queue_log can now contain a key field with more than a single
character in it.
Extensions:
* By default, there is a new option called "autofallthrough" in
extensions.conf that is set to yes. Asterisk 1.0 (and earlier)
behavior was to wait for an extension to be dialed after there were no
more extensions to execute. "autofallthrough" changes this behavior
so that the call will immediately be terminated with BUSY,
CONGESTION, or HANGUP based on Asterisk's best guess. If you are
writing an extension for IVR, you must use the WaitExten application
if "autofallthrough" is set to yes.
AGI:
* AGI scripts did not always get SIGHUP at the end, previously. That
behavior has been fixed. If you do not want your script to terminate
at the end of AGI being called (e.g. on a hangup) then set SIGHUP to
be ignored within your application.
* CallerID is reported with agi_callerid and agi_calleridname instead
of a single parameter holding both.
Music On Hold:
* The preferred format for musiconhold.conf has changed; please see the
sample configuration file for the new format. The existing format
is still supported but will generate warnings when the module is loaded.
chan_modem:
* All the chan_modem channel drivers (aopen, bestdata and i4l) are deprecated
in this release, and will be removed in the next major Asterisk release.
Please migrate to chan_misdn for ISDN interfaces; there is no upgrade
path for aopen and bestdata modem users.
MeetMe:
* The conference application now allows users to increase/decrease their
speaking volume and listening volume (independently of each other and
other users); the 'admin' and 'user' menus have changed, and new sound
files are included with this release. However, if a user calling in
over a Zaptel channel that does NOT have hardware DTMF detection
increases their speaking volume, it is likely they will no longer be
able to enter/exit the menu or make any further adjustments, as the
software DTMF detector will not be able to recognize the DTMF coming
from their device.
GetVar Manager Action:
* Previously, the behavior of the GetVar manager action reported the value
of a variable in the following manner:
> name: value
This has been changed to a manner similar to the SetVar action and is now
> Variable: name
> Value: value

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UPGRADE-1.4.txt
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=========================================================
===
=== Information for upgrading from Asterisk 1.2 to 1.4
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also includes advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
===
=========================================================
Build Process (configure script):
Asterisk now uses an autoconf-generated configuration script to learn how it
should build itself for your system. As it is a standard script, running:
$ ./configure --help
will show you all the options available. This script can be used to tell the
build process what libraries you have on your system (if it cannot find them
automatically), which libraries you wish to have ignored even though they may
be present, etc.
You must run the configure script before Asterisk will build, although it will
attempt to automatically run it for you with no options specified; for most
users, that will result in a similar build to what they would have had before
the configure script was added to the build process (except for having to run
'make' again after the configure script is run). Note that the configure script
does NOT need to be re-run just to rebuild Asterisk; you only need to re-run it
when your system configuration changes or you wish to build Asterisk with
different options.
Build Process (module selection):
The Asterisk source tree now includes a basic module selection and build option
selection tool called 'menuselect'. Run 'make menuselect' to make your choices.
In this tool, you can disable building of modules that you don't care about,
turn on/off global options for the build and see which modules will not
(and cannot) be built because your system does not have the required external
dependencies installed.
The resulting file from menuselect is called 'menuselect.makeopts'. Note that
the resulting menuselect.makeopts file generally contains which modules *not*
to build. The modules listed in this file indicate which modules have unmet
dependencies, a present conflict, or have been disabled by the user in the
menuselect interface. Compiler Flags can also be set in the menuselect
interface. In this case, the resulting file contains which CFLAGS are in use,
not which ones are not in use.
If you would like to save your choices and have them applied against all
builds, the file can be copied to '~/.asterisk.makeopts' or
'/etc/asterisk.makeopts'.
Build Process (Makefile targets):
The 'valgrind' and 'dont-optimize' targets have been removed; their functionality
is available by enabling the DONT_OPTIMIZE setting in the 'Compiler Flags' menu
in the menuselect tool.
It is now possible to run most make targets against a single subdirectory; from
the top level directory, for example, 'make channels' will run 'make all' in the
'channels' subdirectory. This also is true for 'clean', 'distclean' and 'depend'.
Sound (prompt) and Music On Hold files:
Beginning with Asterisk 1.4, the sound files and music on hold files supplied for
use with Asterisk have been replaced with new versions produced from high quality
master recordings, and are available in three languages (English, French and
Spanish) and in five formats (WAV (uncompressed), mu-Law, a-Law, GSM and G.729).
In addition, the music on hold files provided by opsound.org Music are now available
in the same five formats, but no longer available in MP3 format.
The Asterisk 1.4 tarball packages will only include English prompts in GSM format,
(as were supplied with previous releases) and the opsound.org MOH files in WAV format.
All of the other variations can be installed by running 'make menuselect' and
selecting the packages you wish to install; when you run 'make install', those
packages will be downloaded and installed along with the standard files included
in the tarball.
If for some reason you expect to not have Internet access at the time you will be
running 'make install', you can make your package selections using menuselect and
then run 'make sounds' to download (only) the sound packages; this will leave the
sound packages in the 'sounds' subdirectory to be used later during installation.
WARNING: Asterisk 1.4 supports a new layout for sound files in multiple languages;
instead of the alternate-language files being stored in subdirectories underneath
the existing files (for French, that would be digits/fr, letters/fr, phonetic/fr,
etc.) the new layout creates one directory under /var/lib/asterisk/sounds for the
language itself, then places all the sound files for that language under that
directory and its subdirectories. This is the layout that will be created if you
select non-English languages to be installed via menuselect, HOWEVER Asterisk does
not default to this layout and will not find the files in the places it expects them
to be. If you wish to use this layout, make sure you put 'languageprefix=yes' in your
/etc/asterisk/asterisk.conf file, so that Asterisk will know how the files were
installed.
PBX Core:
* The (very old and undocumented) ability to use BYEXTENSION for dialing
instead of ${EXTEN} has been removed.
* Builtin (res_features) transfer functionality attempts to use the context
defined in TRANSFER_CONTEXT variable of the transferer channel first. If
not set, it uses the transferee variable. If not set in any channel, it will
attempt to use the last non macro context. If not possible, it will default
to the current context.
* The autofallthrough setting introduced in Asterisk 1.2 now defaults to 'yes';
if your dialplan relies on the ability to 'run off the end' of an extension
and wait for a new extension without using WaitExten() to accomplish that,
you will need set autofallthrough to 'no' in your extensions.conf file.
Command Line Interface:
* 'show channels concise', designed to be used by applications that will parse
its output, previously used ':' characters to separate fields. However, some
of those fields can easily contain that character, making the output not
parseable. The delimiter has been changed to '!'.
Applications:
* In previous Asterisk releases, many applications would jump to priority n+101
to indicate some kind of status or error condition. This functionality was
marked deprecated in Asterisk 1.2. An option to disable it was provided with
the default value set to 'on'. The default value for the global priority
jumping option is now 'off'.
* The applications Cut, Sort, DBGet, DBPut, SetCIDNum, SetCIDName, SetRDNIS,
AbsoluteTimeout, DigitTimeout, ResponseTimeout, SetLanguage, GetGroupCount,
and GetGroupMatchCount were all deprecated in version 1.2, and therefore have
been removed in this version. You should use the equivalent dialplan
function in places where you have previously used one of these applications.
* The application SetGlobalVar has been deprecated. You should replace uses
of this application with the following combination of Set and GLOBAL():
Set(GLOBAL(name)=value). You may also access global variables exclusively by
using the GLOBAL() dialplan function, instead of relying on variable
interpolation falling back to globals when no channel variable is set.
* The application SetVar has been renamed to Set. The syntax SetVar was marked
deprecated in version 1.2 and is no longer recognized in this version. The
use of Set with multiple argument pairs has also been deprecated. Please
separate each name/value pair into its own dialplan line.
* app_read has been updated to use the newer options codes, using "skip" or
"noanswer" will not work. Use s or n. Also there is a new feature i, for
using indication tones, so typing in skip would give you unexpected results.
* OSPAuth is added to authenticate OSP tokens in in_bound call setup messages.
* The CONNECT event in the queue_log from app_queue now has a second field
in addition to the holdtime field. It contains the unique ID of the
queue member channel that is taking the call. This is useful when trying
to link recording filenames back to a particular call from the queue.
* The old/current behavior of app_queue has a serial type behavior
in that the queue will make all waiting callers wait in the queue
even if there is more than one available member ready to take
calls until the head caller is connected with the member they
were trying to get to. The next waiting caller in line then
becomes the head caller, and they are then connected with the
next available member and all available members and waiting callers
waits while this happens. This cycle continues until there are
no more available members or waiting callers, whichever comes first.
The new behavior, enabled by setting autofill=yes in queues.conf
either at the [general] level to default for all queues or
to set on a per-queue level, makes sure that when the waiting
callers are connecting with available members in a parallel fashion
until there are no more available members or no more waiting callers,
whichever comes first. This is probably more along the lines of how
one would expect a queue should work and in most cases, you will want
to enable this new behavior. If you do not specify or comment out this
option, it will default to "no" to keep backward compatability with the old
behavior.
* Queues depend on the channel driver reporting the proper state
for each member of the queue. To get proper signalling on
queue members that use the SIP channel driver, you need to
enable a call limit (could be set to a high value so it
is not put into action) and also make sure that both inbound
and outbound calls are accounted for.
Example:
[general]
limitonpeer = yes
[peername]
type=friend
call-limit=10
* The app_queue application now has the ability to use MixMonitor to
record conversations queue members are having with queue callers. Please
see configs/queues.conf.sample for more information on this option.
* The app_queue application strategy called 'roundrobin' has been deprecated
for this release. Users are encouraged to use 'rrmemory' instead, since it
provides more 'true' round-robin call delivery. For the Asterisk 1.6 release,
'rrmemory' will be renamed 'roundrobin'.
* The app_queue application option called 'monitor-join' has been deprecated
for this release. Users are encouraged to use 'monitor-type=mixmonitor' instead,
since it provides the same functionality but is not dependent on soxmix or some
other external program in order to mix the audio.
* app_meetme: The 'm' option (monitor) is renamed to 'l' (listen only), and
the 'm' option now provides the functionality of "initially muted".
In practice, most existing dialplans using the 'm' flag should not notice
any difference, unless the keypad menu is enabled, allowing the user
to unmute themsleves.
* ast_play_and_record would attempt to cancel the recording if a DTMF
'0' was received. This behavior was not documented in most of the
applications that used ast_play_and_record and the return codes from
ast_play_and_record weren't checked for properly.
ast_play_and_record has been changed so that '0' no longer cancels a
recording. If you want to allow DTMF digits to cancel an
in-progress recording use ast_play_and_record_full which allows you
to specify which DTMF digits can be used to accept a recording and
which digits can be used to cancel a recording.
* ast_app_messagecount has been renamed to ast_app_inboxcount. There is now a
new ast_app_messagecount function which takes a single context/mailbox/folder
mailbox specification and returns the message count for that folder only.
This addresses the deficiency of not being able to count the number of
messages in folders other than INBOX and Old.
* The exit behavior of the AGI applications has changed. Previously, when
a connection to an AGI server failed, the application would cause the channel
to immediately stop dialplan execution and hangup. Now, the only time that
the AGI applications will cause the channel to stop dialplan execution is
when the channel itself requests hangup. The AGI applications now set an
AGISTATUS variable which will allow you to find out whether running the AGI
was successful or not.
Previously, there was no way to handle the case where Asterisk was unable to
locally execute an AGI script for some reason. In this case, dialplan
execution will continue as it did before, but the AGISTATUS variable will be
set to "FAILURE".
A locally executed AGI script can now exit with a non-zero exit code and this
failure will be detected by Asterisk. If an AGI script exits with a non-zero
exit code, the AGISTATUS variable will be set to "FAILURE" as opposed to
"SUCCESS".
* app_voicemail: The ODBC_STORAGE capability now requires the extended table format
previously used only by EXTENDED_ODBC_STORAGE. This means that you will need to update
your table format using the schema provided in doc/odbcstorage.txt
* app_waitforsilence: Fixes have been made to this application which changes the
default behavior with how quickly it returns. You can maintain "old-style" behavior
with the addition/use of a third "timeout" parameter.
Please consult the application documentation and make changes to your dialplan
if appropriate.
Manager:
* After executing the 'status' manager action, the "Status" manager events
included the header "CallerID:" which was actually only the CallerID number,
and not the full CallerID string. This header has been renamed to
"CallerIDNum". For compatibility purposes, the CallerID parameter will remain
until after the release of 1.4, when it will be removed. Please use the time
during the 1.4 release to make this transition.
* The AgentConnect event now has an additional field called "BridgedChannel"
which contains the unique ID of the queue member channel that is taking the
call. This is useful when trying to link recording filenames back to
a particular call from the queue.
* app_userevent has been modified to always send Event: UserEvent with the
additional header UserEvent: <userspec>. Also, the Channel and UniqueID
headers are not automatically sent, unless you specify them as separate
arguments. Please see the application help for the new syntax.
* app_meetme: Mute and Unmute events are now reported via the Manager API.
Native Manager API commands MeetMeMute and MeetMeUnmute are provided, which
are easier to use than "Action Command:". The MeetMeStopTalking event has
also been deprecated in favor of the already existing MeetmeTalking event
with a "Status" of "on" or "off" added.
* OriginateFailure and OriginateSuccess events were replaced by event
OriginateResponse with a header named "Response" to indicate success or
failure
Variables:
* The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM},
${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE},
and ${LANGUAGE} have all been deprecated in favor of their related dialplan
functions. You are encouraged to move towards the associated dialplan
function, as these variables will be removed in a future release.
* The CDR-CSV variables uniqueid, userfield, and basing time on GMT are now
adjustable from cdr.conf, instead of recompiling.
* OSP applications exports several new variables, ${OSPINHANDLE},
${OSPOUTHANDLE}, ${OSPINTOKEN}, ${OSPOUTTOKEN}, ${OSPCALLING},
${OSPINTIMELIMIT}, and ${OSPOUTTIMELIMIT}
* Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
created channel. This variables holds the channel name of the transferer.
* The dial plan variable PRI_CAUSE will be removed from future versions
of Asterisk.
It is replaced by adding a cause value to the hangup() application.
Functions:
* The function ${CHECK_MD5()} has been deprecated in favor of using an
expression: $[${MD5(<string>)} = ${saved_md5}].
* The 'builtin' functions that used to be combined in pbx_functions.so are
now built as separate modules. If you are not using 'autoload=yes' in your
modules.conf file then you will need to explicitly load the modules that
contain the functions you want to use.
* The ENUMLOOKUP() function with the 'c' option (for counting the number of
records), but the lookup fails to match any records, the returned value will
now be "0" instead of blank.
* The REALTIME() function is now available in version 1.4 and app_realtime has
been deprecated in favor of the new function. app_realtime will be removed
completely with the version 1.6 release so please take the time between
releases to make any necessary changes
* The QUEUEAGENTCOUNT() function has been deprecated in favor of
QUEUE_MEMBER_COUNT().
The IAX2 channel:
* It is possible that previous configurations depended on the order in which
peers and users were specified in iax.conf for forcing the order in which
chan_iax2 matched against them. This behavior is going away and is considered
deprecated in this version. Avoid having ambiguous peer and user entries and
to make things easy on yourself, always set the "username" option for users
so that the remote end can match on that exactly instead of trying to infer
which user you want based on host.
If you would like to go ahead and use the new behavior which doesn't use the
order in the config file to influence matching order, then change the
MAX_PEER_BUCKETS define in chan_iax2.c to a value greater than one. An
example is provided there. By changing this, you will get *much* better
performance on systems that do a lot of peer and user lookups as they will be
stored in memory in a much more efficient manner.
* The "mailboxdetail" option has been deprecated. Previously, if this option
was not enabled, the 2 byte MSGCOUNT information element would be set to all
1's to indicate there there is some number of messages waiting. With this
option enabled, the number of new messages were placed in one byte and the
number of old messages are placed in the other. This is now the default
(and the only) behavior.
The SIP channel:
* The "incominglimit" setting is replaced by the "call-limit" setting in
sip.conf.
* OSP support code is removed from SIP channel to OSP applications. ospauth
option in sip.conf is removed to osp.conf as authpolicy. allowguest option
in sip.conf cannot be set as osp anymore.
* The Asterisk RTP stack has been changed in regards to RFC2833 reception
and transmission. Packets will now be sent with proper duration instead of all
at once. If you are receiving calls from a pre-1.4 Asterisk installation you
will want to turn on the rfc2833compensate option. Without this option your
DTMF reception may act poorly.
* The $SIPUSERAGENT dialplan variable is deprecated and will be removed
in coming versions of Asterisk. Please use the dialplan function
SIPCHANINFO(useragent) instead.
* The ALERT_INFO dialplan variable is deprecated and will be removed
in coming versions of Asterisk. Please use the dialplan application
sipaddheader() to add the "Alert-Info" header to the outbound invite.
* The "canreinvite" option has changed. canreinvite=yes used to disable
re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat
to disable re-invites when NAT=yes. This is propably what you want.
The settings are now: "yes", "no", "nonat", "update". Please consult
sip.conf.sample for detailed information.
The Zap channel:
* Support for MFC/R2 has been removed, as it has not been functional for some
time and it has no maintainer.
The Agent channel:
* Callback mode (AgentCallbackLogin) is now deprecated, since the entire function
it provided can be done using dialplan logic, without requiring additional
channel and module locks (which frequently caused deadlocks). An example of
how to do this using AEL dialplan is in doc/queues-with-callback-members.txt.
The G726-32 codec:
* It has been determined that previous versions of Asterisk used the wrong codeword
packing order for G726-32 data. This version supports both available packing orders,
and can transcode between them. It also now selects the proper order when
negotiating with a SIP peer based on the codec name supplied in the SDP. However,
there are existing devices that improperly request one order and then use another;
Sipura and Grandstream ATAs are known to do this, and there may be others. To
be able to continue to use these devices with this version of Asterisk and the
G726-32 codec, a configuration parameter called 'g726nonstandard' has been added
to sip.conf, so that Asterisk can use the packing order expected by the device (even
though it requested a different order). In addition, the internal format number for
G726-32 has been changed, and the old number is now assigned to AAL2-G726-32. The
result of this is that this version of Asterisk will be able to interoperate over
IAX2 with older versions of Asterisk, as long as this version is told to allow
'g726aal2' instead of 'g726' as the codec for the call.
Installation:
* On BSD systems, the installation directories have changed to more "FreeBSDish"
directories. On startup, Asterisk will look for the main configuration in
/usr/local/etc/asterisk/asterisk.conf
If you have an old installation, you might want to remove the binaries and
move the configuration files to the new locations. The following directories
are now default:
ASTLIBDIR /usr/local/lib/asterisk
ASTVARLIBDIR /usr/local/share/asterisk
ASTETCDIR /usr/local/etc/asterisk
ASTBINDIR /usr/local/bin/asterisk
ASTSBINDIR /usr/local/sbin/asterisk
Music on Hold:
* The music on hold handling has been changed in some significant ways in hopes
to make it work in a way that is much less confusing to users. Behavior will
not change if the same configuration is used from older versions of Asterisk.
However, there are some new configuration options that will make things work
in a way that makes more sense.
Previously, many of the channel drivers had an option called "musicclass" or
something similar. This option set what music on hold class this channel
would *hear* when put on hold. Some people expected (with good reason) that
this option was to configure what music on hold class to play when putting
the bridged channel on hold. This option has now been deprecated.
Two new music on hold related configuration options for channel drivers have
been introduced. Some channel drivers support both options, some just one,
and some support neither of them. Check the sample configuration files to see
which options apply to which channel driver.
The "mohsuggest" option specifies which music on hold class to suggest to the
bridged channel when putting them on hold. The only way that this class can
be overridden is if the bridged channel has a specific music class set that
was done in the dialplan using Set(CHANNEL(musicclass)=something).
The "mohinterpret" option is similar to the old "musicclass" option. It
specifies which music on hold class this channel would like to listen to when
put on hold. This music class is only effective if this channel has no music
class set on it from the dialplan and the bridged channel putting this one on
hold had no "mohsuggest" setting.
The IAX2 and Zap channel drivers have an additional feature for the
"mohinterpret" option. If this option is set to "passthrough", then these
channel drivers will pass through the HOLD message in signalling instead of
starting music on hold on the channel. An example for how this would be
useful is in an enterprise network of Asterisk servers. When one phone on one
server puts a phone on a different server on hold, the remote server will be
responsible for playing the hold music to its local phone that was put on
hold instead of the far end server across the network playing the music.
CDR Records:
* The behavior of the "clid" field of the CDR has always been that it will
contain the callerid ANI if it is set, or the callerid number if ANI was not
set. When using the "callerid" option for various channel drivers, some
would set ANI and some would not. This has been cleared up so that all
channel drivers set ANI. If you would like to change the callerid number
on the channel from the dialplan and have that change also show up in the
CDR, then you *must* set CALLERID(ANI) as well as CALLERID(num).
API:
* There are some API functions that were not previously prefixed with the 'ast_'
prefix but now are; these include the ADSI, ODBC and AGI interfaces. If you
have a module that uses the services provided by res_adsi, res_odbc, or
res_agi, you will need to add ast_ prefixes to the functions that you call
from those modules.
Formats:
* format_wav: The GAIN preprocessor definition has been changed from 2 to 0
in Asterisk 1.4. This change was made in response to user complaints of
choppiness or the clipping of loud signal peaks. The GAIN preprocessor
definition will be retained in Asterisk 1.4, but will be removed in a
future release. The use of GAIN for the increasing of voicemail message
volume should use the 'volgain' option in voicemail.conf

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=========================================================
===
=== Information for upgrading from Asterisk 1.4 to 1.6
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also includes advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
===
=========================================================
AEL:
* Macros are now implemented underneath with the Gosub() application.
Heaven Help You if you wrote code depending on any aspect of this!
Previous to 1.6, macros were implemented with the Macro() app, which
provided a nice feature of auto-returning. The compiler will do its
best to insert a Return() app call at the end of your macro if you did
not include it, but really, you should make sure that all execution
paths within your macros end in "return;".
* The conf2ael program is 'introduced' in this release; it is in a rather
crude state, but deemed useful for making a first pass at converting
extensions.conf code into AEL. More intelligence will come with time.
Core:
* The 'languageprefix' option in asterisk.conf is now deprecated, and
the default sound file layout for non-English sounds is the 'new
style' layout introduced in Asterisk 1.4 (and used by the automatic
sound file installer in the Makefile).
* The ast_expr2 stuff has been modified to handle floating-point numbers.
Numbers of the format D.D are now acceptable input for the expr parser,
Where D is a string of base-10 digits. All math is now done in "long double",
if it is available on your compiler/architecture. This was half-way between
a bug-fix (because the MATH func returns fp by default), and an enhancement.
Also, for those counting on, or needing, integer operations, a series of
'functions' were also added to the expr language, to allow several styles
of rounding/truncation, along with a set of common floating point operations,
like sin, cos, tan, log, pow, etc. The ability to call external functions
like CDR(), etc. was also added, without having to use the ${...} notation.
* The delimiter passed to applications has been changed to the comma (','), as
that is what people are used to using within extensions.conf. If you are
using realtime extensions, you will need to translate your existing dialplan
to use this separator. To use a literal comma, you need merely to escape it
with a backslash ('\'). Another possible side effect is that you may need to
remove the obscene level of backslashing that was necessary for the dialplan
to work correctly in 1.4 and previous versions. This should make writing
dialplans less painful in the future, albeit with the pain of a one-time
conversion. If you would like to avoid this conversion immediately, set
pbx_realtime=1.4 in the [compat] section of asterisk.conf. After
transitioning, set pbx_realtime=1.6 in the same section.
* For the same purpose as above, you may set res_agi=1.4 in the [compat]
section of asterisk.conf to continue to use the '|' delimiter in the EXEC
arguments of AGI applications. After converting to use the ',' delimiter,
change this option to res_agi=1.6.
* As a side effect of the application delimiter change, many places that used
to need quotes in order to get the proper meaning are no longer required.
You now only need to quote strings in configuration files if you literally
want quotation marks within a string.
* Any applications run that contain the pipe symbol but not a comma symbol will
get a warning printed to the effect that the application delimiter has changed.
However, there are legitimate reasons why this might be useful in certain
situations, so this warning can be turned off with the dontwarn option in
asterisk.conf.
* The logger.conf option 'rotatetimestamp' has been deprecated in favor of
'rotatestrategy'. This new option supports a 'rotate' strategy that more
closely mimics the system logger in terms of file rotation.
* The concise versions of various CLI commands are now deprecated. We recommend
using the manager interface (AMI) for application integration with Asterisk.
Voicemail:
* The voicemail configuration values 'maxmessage' and 'minmessage' have
been changed to 'maxsecs' and 'minsecs' to clarify their purpose and
to make them more distinguishable from 'maxmsgs', which sets folder
size. The old variables will continue to work in this version, albeit
with a deprecation warning.
* If you use any interface for modifying voicemail aside from the built in
dialplan applications, then the option "pollmailboxes" *must* be set in
voicemail.conf for message waiting indication (MWI) to work properly. This
is because Voicemail notification is now event based instead of polling
based. The channel drivers are no longer responsible for constantly manually
checking mailboxes for changes so that they can send MWI information to users.
Examples of situations that would require this option are web interfaces to
voicemail or an email client in the case of using IMAP storage.
Applications:
* ChanIsAvail() now has a 't' option, which allows the specified device
to be queried for state without consulting the channel drivers. This
performs mostly a 'ChanExists' sort of function.
* ChannelRedirect() will not terminate the channel that fails to do a
channelredirect as it has done previously. Instead CHANNELREDIRECT_STATUS
will reflect if the attempt was successful of not.
* SetCallerPres() has been replaced with the CALLERPRES() dialplan function
and is now deprecated.
* DISA()'s fifth argument is now an options argument. If you have previously
used 'NOANSWER' in this argument, you'll need to convert that to the new
option 'n'.
* Macro() is now deprecated. If you need subroutines, you should use the
Gosub()/Return() applications. To replace MacroExclusive(), we have
introduced dialplan functions LOCK(), TRYLOCK(), and UNLOCK(). You may use
these functions in any location where you desire to ensure that only one
channel is executing that path at any one time. The Macro() applications
are deprecated for performance reasons. However, since Macro() has been
around for a long time and so many dialplans depend heavily on it, for the
sake of backwards compatibility it will not be removed . It is also worth
noting that using both Macro() and GoSub() at the same time is _heavily_
discouraged.
* Read() now sets a READSTATUS variable on exit. It does NOT automatically
return -1 (and hangup) anymore on error. If you want to hangup on error,
you need to do so explicitly in your dialplan.
* Privacy() no longer uses privacy.conf, so any options must be specified
directly in the application arguments.
* MusicOnHold application now has duration parameter which allows specifying
timeout in seconds.
* WaitMusicOnHold application is now deprecated in favor of extended MusicOnHold.
* SetMusicOnHold is now deprecated. You should use Set(CHANNEL(musicclass)=...)
instead.
* The arguments in ExecIf changed a bit, to be more like other applications.
The syntax is now ExecIf(<cond>?appiftrue(args):appiffalse(args)).
* The behavior of the Set application now depends upon a compatibility option,
set in asterisk.conf. To use the old 1.4 behavior, which allowed Set to take
multiple key/value pairs, set app_set=1.4 in [compat] in asterisk.conf. To
use the new behavior, which permits variables to be set with embedded commas,
set app_set=1.6 in [compat] in asterisk.conf. Note that you can have both
behaviors at the same time, if you switch to using MSet if you want the old
behavior.
Dialplan Functions:
* QUEUE_MEMBER_COUNT() has been deprecated in favor of the QUEUE_MEMBER() function. For
more information, issue a "show function QUEUE_MEMBER" from the CLI.
CDR:
* The cdr_sqlite module has been marked as deprecated in favor of
cdr_sqlite3_custom. It will potentially be removed from the tree
after Asterisk 1.6 is released.
* The cdr_odbc module now uses res_odbc to manage its connections. The
username and password parameters in cdr_odbc.conf, therefore, are no
longer used. The dsn parameter now points to an entry in res_odbc.conf.
* The uniqueid field in the core Asterisk structure has been changed from a
maximum 31 character field to a 149 character field, to account for all
possible values the systemname prefix could be. In the past, if the
systemname was too long, the uniqueid would have been truncated.
* The cdr_tds module now supports all versions of FreeTDS that contain
the db-lib frontend. It will also now log the userfield variable if
the target database table contains a column for it.
Formats:
* format_wav: The GAIN preprocessor definition and source code that used it
is removed. This change was made in response to user complaints of
choppiness or the clipping of loud signal peaks. To increase the volume
of voicemail messages, use the 'volgain' option in voicemail.conf
Channel Drivers:
* SIP: a small upgrade to support the "Record" button on the SNOM360,
which sends a sip INFO message with a "Record: on" or "Record: off"
header. If Asterisk is set up (via features.conf) to accept "One Touch Monitor"
requests (by default, via '*1'), then the user-configured dialpad sequence
is generated, and recording can be started and stopped via this button. The
file names and formats are all controlled via the normal mechanisms. If the
user has not configured the automon feature, the normal "415 Unsupported media type"
is returned, and nothing is done.
* SIP: The "call-limit" option is marked as deprecated. It still works in this version of
Asterisk, but will be removed in the following version. Please use the groupcount functions
in the dialplan to enforce call limits. The "limitonpeer" configuration option is
now renamed to "counteronpeer".
* SIP: The "username" option is now renamed to "defaultuser" to match "defaultip".
These are used only before registration to call a peer with the uri
sip:defaultuser@defaultip
The "username" setting still work, but is deprecated and will not work in
the next version of Asterisk.
* SIP: The old "insecure" options, deprecated in 1.4, have been removed.
"insecure=very" should be changed to "insecure=port,invite"
"insecure=yes" should be changed to "insecure=port"
Be aware that some telephony providers show the invalid syntax in their
sample configurations.
* chan_local.c: the comma delimiter inside the channel name has been changed to a
semicolon, in order to make the Local channel driver compatible with the comma
delimiter change in applications.
* H323: The "tos" setting has changed name to "tos_audio" and "cos" to "cos_audio"
to be compatible with settings in sip.conf. The "tos" and "cos" configuration
is deprecated and will stop working in the next release of Asterisk.
* Console: A new console channel driver, chan_console, has been added to Asterisk.
This new module can not be loaded at the same time as chan_alsa or chan_oss. The
default modules.conf only loads one of them (chan_oss by default). So, unless you
have modified your modules.conf to not use the autoload option, then you will need
to modify modules.conf to add another "noload" line to ensure that only one of
these three modules gets loaded.
* DAHDI: The chan_zap module that supported PSTN interfaces using
Zaptel has been renamed to chan_dahdi, and only supports the DAHDI
telephony driver package for PSTN interfaces. See the
Zaptel-to-DAHDI.txt file for more details on this transition.
* DAHDI: The "msdstrip" option has been deprecated, as it provides no value over
the method of stripping digits in the dialplan using variable substring syntax.
Configuration:
* pbx_dundi.c: tos parameter changed to use new values. Old values like lowdelay,
lowcost and other is not acceptable now. Look into qos.tex for description of
this parameter.
* queues.conf: the queue-lessthan sound file option is no longer available, and the
queue-round-seconds option no longer takes '1' as a valid parameter.
Manager:
* Manager has been upgraded to version 1.1 with a lot of changes.
Please check doc/manager_1_1.txt for information
* The IAXpeers command output has been changed to more closely resemble the
output of the SIPpeers command.
* cdr_manager now reports at the "cdr" level, not at "call" You may need to
change your manager.conf to add the level to existing AMI users, if they
want to see the CDR events generated.
* The Originate command now requires the Originate write permission. For
Originate with the Application parameter, you need the additional System
privilege if you want to do anything that calls out to a subshell.
iLBC Codec:
* Previously, the Asterisk source code distribution included the iLBC
encoder/decoder source code, from Global IP Solutions
(http://www.gipscorp.com). This code is not licensed for
distribution, and thus has been removed from the Asterisk source
code distribution. If you wish to use codec_ilbc to support iLBC
channels in Asterisk, you can run the contrib/scripts/get_ilbc_source.sh
script to download the source and put it in the proper place in
the Asterisk build tree. Once that is done you can follow your normal
steps of building Asterisk. You will need to run 'menuselect' and enable
the iLBC codec in the 'Codec Translators' category.

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===========================================================
===
=== Information for upgrading between Asterisk versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also includes advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
===
===========================================================
From 1.8.13 to 1.8.14:
* permitdirectmedia/denydirectmedia now controls whether peers can be
bridged via directmedia by comparing the ACL to the bridging peer's
address rather than its own address.
From 1.8.12 to 1.8.13:
* The complex processor detection and optimization has been removed from
the makefile in favor of using native optimization suppport when available.
BUILD_NATIVE can be disabled via menuselect under "Compiler Flags".
From 1.8.10 to 1.8.11:
* If no transport is specified in sip.conf, transport will default to UDP.
Also, if multiple transport= lines are used, only the last will be used.
From 1.6.2 to 1.8:
* chan_sip no longer sets HASH(SIP_CAUSE,<chan name>) on channels by default.
This must now be enabled by setting 'sipstorecause' to 'yes' in sip.conf.
This carries a performance penalty.
* Asterisk now requires libpri 1.4.11+ for PRI support.
* A couple of CLI commands in res_ais were changed back to their original form:
"ais show clm members" --> "ais clm show members"
"ais show evt event channels" --> "ais evt show event channels"
* The default value for 'autofill' and 'shared_lastcall' in queues.conf has
been changed to 'yes'.
* The default value for the alwaysauthreject option in sip.conf has been changed
from "no" to "yes".
* The behavior of the 'parkedcallstimeout' has changed slightly. The formulation
of the extension name that a timed out parked call is delivered to when this
option is set to 'no' was modified such that instead of converting '/' to '0',
the '/' is converted to an underscore '_'. See the updated documentation in
features.conf.sample for more information on the behavior of the
'parkedcallstimeout' option.
* Asterisk-addons no longer exists as an independent package. Those modules
now live in the addons directory of the main Asterisk source tree. They
are not enabled by default. For more information about why modules live in
addons, see README-addons.txt.
* The rarely used 'event_log' and LOG_EVENT channel have been removed; the few
users of this channel in the tree have been converted to LOG_NOTICE or removed
(in cases where the same message was already generated to another channel).
* The usage of RTP inside of Asterisk has now become modularized. This means
the Asterisk RTP stack now exists as a loadable module, res_rtp_asterisk.
If you are not using autoload=yes in modules.conf you will need to ensure
it is set to load. If not, then any module which uses RTP (such as chan_sip)
will not be able to send or receive calls.
* The app_dahdiscan.c file has been removed, but the dialplan app DAHDIScan still
remains. It now exists within app_chanspy.c and retains the exact same
functionality as before.
* The default behavior for Set, AGI, and pbx_realtime has been changed to implement
1.6 behavior by default, if there is no [compat] section in asterisk.conf. In
prior versions, the behavior defaulted to 1.4 behavior, to assist in upgrades.
Specifically, that means that pbx_realtime and res_agi expect you to use commas
to separate arguments in applications, and Set only takes a single pair of
a variable name/value. The old 1.4 behavior may still be obtained by setting
app_set, pbx_realtime, and res_agi each to 1.4 in the [compat] section of
asterisk.conf.
* The PRI channels in chan_dahdi can no longer change the channel name if a
different B channel is selected during call negotiation. To prevent using
the channel name to infer what B channel a call is using and to avoid name
collisions, the channel name format is changed.
The new channel naming for PRI channels is:
DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
* Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type)
so the dialplan can determine the B channel currently in use by the channel.
Use CHANNEL(no_media_path) to determine if the channel even has a B channel.
* Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk
channel so AMI applications can passively determine the B channel currently
in use. Calls with "no-media" as the DAHDIChannel do not have an associated
B channel. No-media calls are either on hold or call-waiting.
* The ChanIsAvail application has been changed so the AVAILSTATUS variable
no longer contains both the device state and cause code. The cause code
is now available in the AVAILCAUSECODE variable. If existing dialplan logic
is written to expect AVAILSTATUS to contain the cause code it needs to be
changed to use AVAILCAUSECODE.
* ExternalIVR will now send Z events for invalid or missing files, T events
now include the interrupted file and bugs in argument parsing have been
fixed so there may be arguments specified in incorrect ways that were
working that will no longer work. Please see
https://wiki.asterisk.org/wiki/display/AST/External+IVR+Interface for details.
* OSP lookup application changes following variable names:
OSPPEERIP to OSPINPEERIP
OSPTECH to OSPOUTTECH
OSPDEST to OSPDESTINATION
OSPCALLING to OSPOUTCALLING
OSPCALLED to OSPOUTCALLED
OSPRESULTS to OSPDESTREMAILS
* The Manager event 'iax2 show peers' output has been updated. It now has a
similar output of 'sip show peers'.
* VoiceMailMain and VMAuthenticate, if a '*' is entered in the first position
of a Mailbox or Password, will, if it exists, jump to the 'a' extension in
the current dialplan context.
* The CALLERPRES() dialplan function is deprecated in favor of
CALLERID(num-pres) and CALLERID(name-pres).
* Environment variables that start with "AST_" are reserved to the system and
may no longer be set from the dialplan.
* When a call is redirected inside of a Dial, the app and appdata fields of the
CDR will now be set to "AppDial" and "(Outgoing Line)" instead of being blank.
* The CDR handling of billsec and duration field has changed. If your table
definition specifies those fields as float,double or similar they will now
be logged with microsecond accuracy instead of a whole integer.
* chan_sip will no longer set up a local call forward when receiving a
482 Loop Detected response. The dialplan will just continue from where it
left off.
* The 'stunaddr' option has been removed from chan_sip. This feature did not
behave as expected, had no correct use case, and was not RFC compliant. The
removal of this feature will hopefully be followed by a correct RFC compliant
STUN implementation in chan_sip in the future.
* The default value for the pedantic option in sip.conf has been changed
from "no" to "yes".
* The ConnectedLineNum and ConnectedLineName headers were added to many AMI
events/responses if the CallerIDNum/CallerIDName headers were also present.
The addition of connected line support changes the behavior of the channel
caller ID somewhat. The channel caller ID value no longer time shares with
the connected line ID on outgoing call legs. The timing of some AMI
events/responses output the connected line ID as caller ID. These party ID's
are now separate.
* The Dial application d and H options do not automatically answer the call
anymore. It broke DTMF attended transfers. Since many SIP and ISDN phones
cannot send DTMF before a call is connected, you need to answer the call
leg to those phones before using Dial with these options for them to have
any effect before the dialed party answers.
* The outgoing directory (where .call files are read) now uses inotify to
detect file changes instead of polling the directory on a regular basis.
If your outgoing folder is on a NFS mount or another network file system,
changes to the files will not be detected. You can revert to polling the
directory by specifying --without-inotify to configure before compiling.
* The 'sipusers' realtime table has been removed completely. Use the 'sippeers'
table with type 'user' for user type objects.
* The sip.conf allowoverlap option now accepts 'dtmf' as a value. If you
are using the early media DTMF overlap dialing method you now need to set
allowoverlap=dtmf.
From 1.6.1 to 1.6.2:
* SIP no longer sends the 183 progress message for early media by
default. Applications requiring early media should use the
progress() dialplan app to generate the progress message.
* The firmware for the IAXy has been removed from Asterisk. It can be
downloaded from http://downloads.digium.com/pub/iaxy/. To have Asterisk
install the firmware into its proper location, place the firmware in the
contrib/firmware/iax/ directory in the Asterisk source tree before running
"make install".
* T.38 FAX error correction mode can no longer be configured in udptl.conf;
instead, it is configured on a per-peer (or global) basis in sip.conf, with
the same default as was present in udptl.conf.sample.
* T.38 FAX maximum datagram size can no longer be configured in updtl.conf;
instead, it is either supplied by the application servicing the T.38 channel
(for a FAX send or receive) or calculated from the bridged endpoint's
maximum datagram size (for a T.38 FAX passthrough call). In addition, sip.conf
allows for overriding the value supplied by a remote endpoint, which is useful
when T.38 connections are made to gateways that supply incorrectly-calculated
maximum datagram sizes.
* There have been some changes to the IAX2 protocol to address the security
concerns documented in the security advisory AST-2009-006. Please see the
IAX2 security document, doc/IAX2-security.pdf, for information regarding
backwards compatibility with versions of Asterisk that do not contain these
changes to IAX2.
* The 'canreinvite' option support by the SIP, MGCP and Skinny channel drivers
has been renamed to 'directmedia', to better reflect what it actually does.
In the case of SIP, there are still re-INVITEs issued for T.38 negotiation,
starting and stopping music-on-hold, and other reasons, and the 'canreinvite'
option never had any effect on these cases, it only affected the re-INVITEs
used for direct media path setup. For MGCP and Skinny, the option was poorly
named because those protocols don't even use INVITE messages at all. For
backwards compatibility, the old option is still supported in both normal
and Realtime configuration files, but all of the sample configuration files,
Realtime/LDAP schemas, and other documentation refer to it using the new name.
* The default console now will use colors according to the default background
color, instead of forcing the background color to black. If you are using a
light colored background for your console, you may wish to use the option
flag '-W' to present better color choices for the various messages. However,
if you'd prefer the old method of forcing colors to white text on a black
background, the compatibility option -B is provided for this purpose.
* SendImage() no longer hangs up the channel on transmission error or on
any other error; in those cases, a FAILURE status is stored in
SENDIMAGESTATUS and dialplan execution continues. The possible
return values stored in SENDIMAGESTATUS are: SUCCESS, FAILURE, and
UNSUPPORTED. ('OK' has been replaced with 'SUCCESS', and 'NOSUPPORT'
has been replaced with 'UNSUPPORTED'). This change makes the
SendImage application more consistent with other applications.
* skinny.conf now has separate sections for lines and devices.
Please have a look at configs/skinny.conf.sample and update
your skinny.conf.
* Queue names previously were treated in a case-sensitive manner,
meaning that queues with names like "sales" and "sALeS" would be
seen as unique queues. The parsing logic has changed to use
case-insensitive comparisons now when originally hashing based on
queue names, meaning that now the two queues mentioned as examples
earlier will be seen as having the same name.
* The SPRINTF() dialplan function has been moved into its own module,
func_sprintf, and is no longer included in func_strings. If you use this
function and do not use 'autoload=yes' in modules.conf, you will need
to explicitly load func_sprintf for it to be available.
* The res_indications module has been removed. Its functionality was important
enough that most of it has been moved into the Asterisk core.
Two applications previously provided by res_indications, PlayTones and
StopPlayTones, have been moved into a new module, app_playtones.
* Support for Taiwanese was incorrectly supported with the "tw" language code.
In reality, the "tw" language code is reserved for the Twi language, native
to Ghana. If you were previously using the "tw" language code, you should
switch to using either "zh" (for Mandarin Chinese) or "zh_TW" for Taiwan
specific localizations. Additionally, "mx" should be changed to "es_MX",
Georgian was incorrectly specified as "ge" but should be "ka", and Czech is
"cs", not "cz".
* DAHDISendCallreroutingFacility() parameters are now comma-separated,
instead of the old pipe.
* res_jabber: autoprune has been disabled by default, to avoid misconfiguration
that would end up being interpreted as a bug once Asterisk started removing
the contacts from a user list.
* The cdr.conf file must exist and be configured correctly in order for CDR
records to be written.
* cdr_pgsql now assumes the encoding of strings it is handed are in LATIN9,
which should cover most uses of the extended ASCII set. If your strings
use a different encoding in Asterisk, the "encoding" parameter may be set
to specify the correct character set.
From 1.6.0.1 to 1.6.1:
* The ast_agi_register_multiple() and ast_agi_unregister_multiple()
API calls were added in 1.6.0, so that modules that provide multiple
AGI commands could register/unregister them all with a single
step. However, these API calls were not implemented properly, and did
not allow the caller to know whether registration or unregistration
succeeded or failed. They have been redefined to now return success
or failure, but this means any code using these functions will need
be recompiled after upgrading to a version of Asterisk containing
these changes. In addition, the source code using these functions
should be reviewed to ensure it can properly react to failure
of registration or unregistration of its API commands.
* The ast_agi_fdprintf() API call has been renamed to ast_agi_send()
to better match what it really does, and the argument order has been
changed to be consistent with other API calls that perform similar
operations.
From 1.6.0.x to 1.6.1:
* In previous versions of Asterisk, due to the way objects were arranged in
memory by chan_sip, the order of entries in sip.conf could be adjusted to
control the behavior of matching against peers and users. The way objects
are managed has been significantly changed for reasons involving performance
and stability. A side effect of these changes is that the order of entries
in sip.conf can no longer be relied upon to control behavior.
* The following core commands dealing with dialplan have been deprecated: 'core
show globals', 'core set global' and 'core set chanvar'. Use the equivalent
'dialplan show globals', 'dialplan set global' and 'dialplan set chanvar'
instead.
* In the dialplan expression parser, the logical value of spaces
immediately preceding a standalone 0 previously evaluated to
true. It now evaluates to false. This has confused a good many
people in the past (typically because they failed to realize the
space had any significance). Since this violates the Principle of
Least Surprise, it has been changed.
* While app_directory has always relied on having a voicemail.conf or users.conf file
correctly set up, it now is dependent on app_voicemail being compiled as well.
* SIP: All of the functionality in SIPCHANINFO() has been implemented in CHANNEL(),
and you should start using that function instead for retrieving information about
the channel in a technology-agnostic way.
* If you have any third party modules which use a config file variable whose
name ends in a '+', please note that the append capability added to this
version may now conflict with that variable naming scheme. An easy
workaround is to ensure that a space occurs between the '+' and the '=',
to differentiate your variable from the append operator. This potential
conflict is unlikely, but is documented here to be thorough.
* The "Join" event from app_queue now uses the CallerIDNum header instead of
the CallerID header to indicate the CallerID number.
* If you use ODBC storage for voicemail, there is a new field called "flag"
which should be a char(8) or larger. This field specifies whether or not a
message has been designated to be "Urgent", "PRIORITY", or not.

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===========================================================
===
=== Information for upgrading between Asterisk versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also include advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
===
===========================================================
From 10.4 to 10.5:
* The complex processor detection and optimization has been removed from
the makefile in favor of using native optimization suppport when available.
BUILD_NATIVE can be disabled via menuselect under "Compiler Flags".
From 10.2 to 10.3:
* If no transport is specified in sip.conf, transport will default to UDP.
Also, if multiple transport= lines are used, only the last will be used.
From 1.8 to 10:
cel_pgsql:
- This module now expects an 'extra' column in the database for data added
using the CELGenUserEvent() application.
ConfBridge
- ConfBridge's dialplan arguments have changed and are not
backwards compatible.
File Interpreters
- The format interpreter formats/format_sln16.c for the file extension
'.sln16' has been removed. The '.sln16' file interpreter now exists
in the formats/format_sln.c module along with new support for sln12,
sln24, sln32, sln44, sln48, sln96, and sln192 file extensions.
HTTP:
- A bindaddr must be specified in order for the HTTP server
to run. Previous versions would default to 0.0.0.0 if no
bindaddr was specified.
Gtalk:
- The default value for 'context' and 'parkinglots' in gtalk.conf has
been changed to 'default', previously they were empty.
chan_dahdi:
- The mohinterpret=passthrough setting is deprecated in favor of
moh_signaling=notify.
pbx_lua:
- Execution no longer continues after applications that do dialplan jumps
(such as app.goto). Now when an application such as app.goto() is called,
control is returned back to the pbx engine and the current extension
function stops executing.
- the autoservice now defaults to being on by default
- autoservice_start() and autoservice_start() no longer return a value.
Queue:
- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
Asterisk Database:
- The internal Asterisk database has been switched from Berkeley DB 1.86 to
SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
utility in the UTILS section of menuselect. If an existing astdb is found and no
astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
convert an existing astdb to the SQLite3 version automatically at runtime.
Module Support Level
- All modules in the addons, apps, bridge, cdr, cel, channels, codecs,
formats, funcs, pbx, and res have been updated to include MODULEINFO data
that includes <support_level> tags with a value of core, extended, or deprecated.
More information is available on the Asterisk wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
Deprecated modules are now marked to not build by default and must be explicitly
enabled in menuselect.
===========================================================
===========================================================

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===========================================================
===
=== Information for upgrading between Asterisk versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also include advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
===
===========================================================
From 11.6 to 11.7:
ConfBridge
- ConfBridge now has the ability to set the language of announcements to the
conference. The language can be set on a bridge profile in confbridge.conf
or by the dialplan function CONFBRIDGE(bridge,language)=en.
chan_sip - Clarify The "sip show peers" Forcerport Column And Add Comedia
- Under the "Forcerport" column, the "N" used to mean NAT (i.e. Yes). With
the additon of auto_* NAT settings, the meaning changed and there was a
certain combination of letters added to indicate the current setting. The
combination of using "Y", "N", "A" or "a", can be confusing. Therefore, we
now display clearly what the current Forcerport setting is: "Yes", "No",
"Auto (Yes)", "Auto (No)".
- Since we are clarifying the Forcerport column, we have added a column to
display the Comedia setting since this is useful information as well. We
no longer have a simple "NAT" setting like other versions before 11.
From 11.5 to 11.6:
* res_agi will now properly indicate if there was an error in streaming an
audio file. The result code will be -1 and the result returned from the
the function will be RESULT_FAILURE instead of the prior behavior of always
returning RESULT_SUCCESS even if there was an error.
From 11.4 to 11.5:
* The default settings for chan_sip are now overriden properly by the general
settings in sip.conf. Please look over your settings upon upgrading.
From 11.3 to 11.4:
* Added the 'n' option to MeetMe to prevent application of the DENOISE function
to a channel joining a conference. Some channel drivers that vary the number
of audio samples in a voice frame will experience significant quality problems
if a denoiser is attached to the channel; this option gives them the ability
to remove the denoiser without having to unload func_speex.
* The Registry AMI event for SIP registrations will now always include the
Username field. A previous bug fix missed an instance where it was not
included; that has been corrected in this release.
From 11.2.0 to 11.2.1:
* Asterisk would previously not output certain error messages when a remote
console attempted to connect to Asterisk and no instance of Asterisk was
running. This error message is displayed on stderr; as a result, some
initialization scripts that used remote consoles to test for the presence
of a running Asterisk instance started to display erroneous error messages.
The init.d scripts and the safe_asterisk have been updated in the contrib
folder to account for this.
From 11.2 to 11.3:
* Now by default, when Asterisk is installed in a path other than /usr, the
Asterisk binary will search for shared libraries in ${libdir} in addition to
searching system libraries. This allows Asterisk to find its shared
libraries without having to specify LD_LIBRARY_PATH. This can be disabled by
passing --disable-rpath to configure.
From 10 to 11:
Voicemail:
- All voicemails now have a "msg_id" which uniquely identifies a message. For
users of filesystem and IMAP storage of voicemail, this should be transparent.
For users of ODBC, you will need to add a "msg_id" column to your voice mail
messages table. This should be a string capable of holding at least 32 characters.
All messages created in old Asterisk installations will have a msg_id added to
them when required. This operation should be transparent as well.
Parking:
- The comebacktoorigin setting must now be set per parking lot. The setting in
the general section will not be applied automatically to each parking lot.
- The BLINDTRANSFER channel variable is deleted from a channel when it is
bridged to prevent subtle bugs in the parking feature. The channel
variable is used by Asterisk internally for the Park application to work
properly. If you were using it for your own purposes, copy it to your
own channel variable before the channel is bridged.
res_ais:
- Users of res_ais in versions of Asterisk prior to Asterisk 11 must change
to use the res_corosync module, instead. OpenAIS is deprecated, but
Corosync is still actively developed and maintained. Corosync came out of
the OpenAIS project.
Dialplan Functions:
- MAILBOX_EXISTS has been deprecated. Use VM_INFO with the 'exists' parameter
instead.
- 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.
- The REDIRECTING function now supports the redirecting original party id
and reason.
- The HANGUPCAUSE and HANGUPCAUSE_KEYS functions have been introduced to
provide a replacement for the SIP_CAUSE hash. The HangupCauseClear
application has also been introduced to remove this data from the channel
when necessary.
func_enum:
- ENUM query functions now return a count of -1 on lookup error to
differentiate between a failed query and a successful query with 0 results
matching the specified type.
CDR:
- cdr_adaptive_odbc now supports specifying a schema so that Asterisk can
connect to databases that use schemas.
Configuration Files:
- Files listed below have been updated to be more consistent with how Asterisk
parses configuration files. This makes configuration files more consistent
with what is expected across modules.
- cdr.conf: [general] and [csv] sections
- dnsmgr.conf
- dsp.conf
- The 'verbose' setting in logger.conf now takes an optional argument,
specifying the verbosity level for each logging destination. The default,
if not otherwise specified, is a verbosity of 3.
AMI:
- DBDelTree now correctly returns an error when 0 rows are deleted just as
the DBDel action does.
- The IAX2 PeerStatus event now sends a 'Port' header. In Asterisk 10, this was
erroneously being sent as a 'Post' header.
CCSS:
- Macro is deprecated. Use cc_callback_sub instead of cc_callback_macro
in channel configurations.
app_meetme:
- The 'c' option (announce user count) will now work even if the 'q' (quiet)
option is enabled.
app_followme:
- Answered outgoing calls no longer get cut off when the next step is started.
You now have until the last step times out to decide if you want to accept
the call or not before being disconnected.
chan_gtalk:
- chan_gtalk has been deprecated in favor of the chan_motif channel driver. It is recommended
that users switch to using it as it is a core supported module.
chan_jingle:
- chan_jingle has been deprecated in favor of the chan_motif channel driver. It is recommended
that users switch to using it as it is a core supported module.
SIP
===
- A new option "tonezone" for setting default tonezone for the channel driver
or individual devices
- A new manager event, "SessionTimeout" has been added and is triggered when
a call is terminated due to RTP stream inactivity or SIP session timer
expiration.
- SIP_CAUSE is now deprecated. It has been modified to use the same
mechanism as the HANGUPCAUSE function. Behavior should not change, but
performance should be vastly improved. The HANGUPCAUSE function should now
be used instead of SIP_CAUSE. Because of this, the storesipcause option in
sip.conf is also deprecated.
- The sip paramater for Originating Line Information (oli, isup-oli, and
ss7-oli) is now parsed out of the From header and copied into the channel's
ANI2 information field. This is readable from the CALLERID(ani2) dialplan
function.
- ICE support has been added and is enabled by default. Some endpoints may have
problems with the ICE candidates within the SDP. If this is the case ICE support
can be disabled globally or on a per-endpoint basis using the icesupport
configuration option. Symptoms of this include one way media or no media flow.
chan_unistim
- Due to massive update in chan_unistim phone keys functions and on-screen
information changed.
users.conf:
- A defined user with hasvoicemail=yes now finally uses a Gosub to stdexten
as documented in extensions.conf.sample since v1.6.0 instead of a Macro as
documented in v1.4. Set the asterisk.conf stdexten=macro parameter to
invoke the stdexten the old way.
res_jabber
- This module has been deprecated in favor of the res_xmpp module. The res_xmpp
module is backwards compatible with the res_jabber configuration file, dialplan
functions, and AMI actions. The old CLI commands can also be made available using
the res_clialiases template for Asterisk 11.
From 1.8 to 10:
cel_pgsql:
- This module now expects an 'extra' column in the database for data added
using the CELGenUserEvent() application.
ConfBridge
- ConfBridge's dialplan arguments have changed and are not
backwards compatible.
File Interpreters
- The format interpreter formats/format_sln16.c for the file extension
'.sln16' has been removed. The '.sln16' file interpreter now exists
in the formats/format_sln.c module along with new support for sln12,
sln24, sln32, sln44, sln48, sln96, and sln192 file extensions.
HTTP:
- A bindaddr must be specified in order for the HTTP server
to run. Previous versions would default to 0.0.0.0 if no
bindaddr was specified.
Gtalk:
- The default value for 'context' and 'parkinglots' in gtalk.conf has
been changed to 'default', previously they were empty.
chan_dahdi:
- The mohinterpret=passthrough setting is deprecated in favor of
moh_signaling=notify.
pbx_lua:
- Execution no longer continues after applications that do dialplan jumps
(such as app.goto). Now when an application such as app.goto() is called,
control is returned back to the pbx engine and the current extension
function stops executing.
- the autoservice now defaults to being on by default
- autoservice_start() and autoservice_start() no longer return a value.
Queue:
- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
Asterisk Database:
- The internal Asterisk database has been switched from Berkeley DB 1.86 to
SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
utility in the UTILS section of menuselect. If an existing astdb is found and no
astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
convert an existing astdb to the SQLite3 version automatically at runtime. If
moving back from Asterisk 10 to Asterisk 1.8, the astdb2bdb utility can be used
to create a Berkeley DB copy of the SQLite3 astdb that Asterisk 10 uses.
Manager:
- The AMI protocol version was incremented to 1.2 as a result of changing two
instances of the Unlink event to Bridge events. This change was documented
as part of the AMI 1.1 update, but two Unlink events were inadvertently left
unchanged.
Module Support Level
- All modules in the addons, apps, bridge, cdr, cel, channels, codecs,
formats, funcs, pbx, and res have been updated to include MODULEINFO data
that includes <support_level> tags with a value of core, extended, or deprecated.
More information is available on the Asterisk wiki at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
Deprecated modules are now marked to not build by default and must be explicitly
enabled in menuselect.
chan_sip:
- Setting of HASH(SIP_CAUSE,<slave-channel-name>) on channels is now disabled
by default. It can be enabled using the 'storesipcause' option. This feature
has a significant performance penalty.
UDPTL:
- The default UDPTL port range in udptl.conf.sample differed from the defaults
in the source. If you didn't have a config file, you got 4500 to 4599. Now the
default is 4000 to 4999.
===========================================================
===========================================================

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===========================================================
===
=== Information for upgrading between Asterisk versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also include advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
=== UPGRADE-11.txt -- Upgrade info for 10 to 11
===
===========================================================
There are many significant architectural changes in Asterisk 12. It is
recommended that you not only read through this document for important
changes that affect an upgrade, but that you also read through the CHANGES
document in depth to better understand the new options available to you.
Additional information on the architectural changes made in Asterisk can be
found on the Asterisk wiki (https://wiki.asterisk.org)
Of particular note, the following systems in Asterisk underwent significant
changes. Documentation for the changes and a specification for their
behavior in Asterisk 12 is also available on the Asterisk wiki.
- AMI: Many events were changed, and the semantics of channels and bridges
were defined. In particular, how channels and bridges behave under
transfer scenarios and situations involving multiple parties has
changed significantly. See https://wiki.asterisk.org/wiki/x/dAFRAQ
for more information.
- CDR: CDR logic was extracted from the many locations it existed in across
Asterisk and implemented as a consumer of Stasis message bus events.
As a result, consistency of records has improved significantly and the
behavior of CDRs in transfer scenarios has been defined in the CDR
specification. However, significant behavioral changes in CDRs resulted
from the transition. The most significant change is the addition of
CDR entries when a channel who is the Party A in a CDR leaves a bridge.
See https://wiki.asterisk.org/wiki/x/pwpRAQ for more information.
- CEL: Much like CDRs, CEL was removed from the many locations it existed in
across Asterisk and implemented as a consumer of Stasis message bus
events. It now closely follows the Bridging API model of channels and
bridges, and has a much closer consistency of conveyed events as AMI.
For the changes in events, see https://wiki.asterisk.org/wiki/x/4ICLAQ.
Build System:
- Removed the CHANNEL_TRACE development mode build option. Certain aspects of
the CHANNEL_TRACE build option were incompatible with the new bridging
architecture.
- Asterisk now depends on libjansson, libuuid and optionally (but recommended)
libxslt and uriparser.
- The new SIP stack and channel driver uses a particular version of PJSIP.
Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
configuring and installing PJSIP for use with Asterisk.
AgentLogin and chan_agent:
- Along with AgentRequest, this application has been modified to be a
replacement for chan_agent. The chan_agent module and the Agent channel
driver have been removed from Asterisk, as the concept of a channel driver
proxying in front of another channel driver was incompatible with the new
architecture (and has had numerous problems through past versions of
Asterisk). The act of a channel calling the AgentLogin application places the
channel into a pool of agents that can be requested by the AgentRequest
application. Note that this application, as well as all other agent related
functionality, is now provided by the app_agent_pool module.
- This application no longer performs agent authentication. If authentication
is desired, the dialplan needs to perform this function using the
Authenticate or VMAuthenticate application or through an AGI script before
running AgentLogin.
- The agents.conf schema has changed. Rather than specifying agents on a
single line in comma delineated fashion, each agent is defined in a separate
context. This allows agents to use the power of context templates in their
definition.
- A number of parameters from agents.conf have been removed. This includes
maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
urlprefix, and savecallsin. These options were obsoleted by the move from
a channel driver model to the bridging/application model provided by
app_agent_pool.
- The AGENTUPDATECDR channel variable has also been removed, for the same
reason as the updatecdr option.
- The endcall and enddtmf configuration options are removed. Use the
dialplan function CHANNEL(dtmf_features) to set DTMF features on the agent
channel before calling AgentLogin.
AgentMonitorOutgoing
- This application has been removed. It was a holdover from when
AgentCallbackLogin was removed.
Answer
- It is no longer possible to bypass updating the CDR when answering a
channel. CDRs are based on the channel state and will be updated when
the channel is Answered.
ControlPlayback
- The channel variable CPLAYBACKSTATUS may now return the value
'REMOTESTOPPED' when playback is stopped by an external entity.
DISA
- This application now has a dependency on the app_cdr module. It uses this
module to hide the CDR created prior to execution of the DISA application.
DumpChan:
- The output of DumpChan no longer includes the DirectBridge or IndirectBridge
fields. Instead, if a channel is in a bridge, it includes a BridgeID field
containing the unique ID of the bridge that the channel happens to be in.
ForkCDR:
- Nearly every parameter in ForkCDR has been updated and changed to reflect
the changes in CDRs. Please see the documentation for the ForkCDR
application, as well as the CDR specification on the Asterisk wiki.
NoCDR:
- The NoCDR application has been deprecated. Please use the CDR_PROP function
to disable CDRs on a channel.
ParkAndAnnounce:
- The app_parkandannounce module has been removed. The application
ParkAndAnnounce is now provided by the res_parking module. See the
Parking changes for more information.
ResetCDR:
- The 'w' and 'a' options have been removed. Dispatching CDRs to registered
backends occurs on an as-needed basis in order to preserve linkedid
propagation and other needed behavior.
- The 'e' option is deprecated. Please use the CDR_PROP function to enable
CDRs on a channel that they were previously disabled on.
- The ResetCDR application is no longer a part of core Asterisk, and instead
is now delivered as part of app_cdr.
Queues:
- Queue strategy rrmemory now has a predictable order similar to strategy
rrordered. Members will be called in the order that they are added to the
queue.
- Removed the queues.conf check_state_unknown option. It is no longer
necessary.
- It is now possible to play the Queue prompts to the first user waiting in a
call queue. Note that this may impact the ability for agents to talk with
users, as a prompt may still be playing when an agent connects to the user.
This ability is disabled by default but can be enabled on an individual
queue using the 'announce-to-first-user' option.
- The configuration options eventwhencalled and eventmemberstatus have been
removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
sent. The "Variable" fields will also no longer exist on the Agent* events.
These events can be filtered out from a connected AMI client using the
eventfilter setting in manager.conf.
- The queue log now differentiates between blind and attended transfers. A
blind transfer will result in a BLINDTRANSFER message with the destination
context and extension. An attended transfer will result in an
ATTENDEDTRANSFER message. This message will indicate the method by which
the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
for running an application on a bridge or channel, or "LINK" for linking
two bridges together with local channels. The queue log will also now detect
externally initiated blind and attended transfers and record the transfer
status accordingly.
- When performing queue pause/unpause on an interface without specifying an
individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
least one member of any queue exists for that interface.
SetAMAFlags
- This application is deprecated in favor of CHANNEL(amaflags).
VoiceMail:
- Mailboxes defined by app_voicemail MUST be referenced by the rest of the
system as mailbox@context. The rest of the system cannot add @default
to mailbox identifiers for app_voicemail that do not specify a context
any longer. It is a mailbox identifier format that should only be
interpreted by app_voicemail.
- The voicemail.conf configuration file now has an 'alias' configuration
parameter for use with the Directory application. The voicemail realtime
database table schema has also been updated with an 'alias' column. Systems
using voicemail with realtime should update their schemas accordingly.
Channel Drivers:
- When a channel driver is configured to enable jiterbuffers, they are now
applied unconditionally when a channel joins a bridge. If a jitterbuffer
is already set for that channel when it enters, such as by the JITTERBUFFER
function, then the existing jitterbuffer will be used and the one set by
the channel driver will not be applied.
chan_bridge
- chan_bridge is removed and its functionality is incorporated into ConfBridge
itself.
chan_dahdi:
- Analog port dialing and deferred DTMF dialing for PRI now distinguishes
between 'w' and 'W'. The 'w' pauses dialing for half a second. The 'W'
pauses dialing for one second.
- The default for inband_on_proceeding has changed to no.
- The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
A range of channels can be specified to be destroyed. Note that this command
should only be used if you understand the risks it entails.
- The script specified by the chan_dahdi.conf mwimonitornotify option now gets
the exact configured mailbox name. For app_voicemail mailboxes this is
mailbox@context.
- Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
- ignore_failed_channels now defaults to True: the channel will continue to
be configured even if configuring it has failed. This is generally a
better setup for systems with not more than one DAHDI device or with DAHDI
>= 2.8.0 .
chan_local:
- The /b option has been removed.
- chan_local moved into the system core and is no longer a loadable module.
chan_sip:
- The 'callevents' parameter has been removed. Hold AMI events are now raised
in the core, and can be filtered out using the 'eventfilter' parameter
in manager.conf.
- Dynamic realtime tables for SIP Users can now include a 'path' field. This
will store the path information for that peer when it registers. Realtime
tables can also use the 'supportpath' field to enable Path header support.
- LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
objectIdentifier. This maps to the supportpath option in sip.conf.
Core:
- Masquerades as an operation inside Asterisk have been effectively hidden
by the migration to the Bridging API. As such, many 'quirks' of Asterisk
no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
dropping of frame/audio hooks, and other internal implementation details
that users had to deal with. This fundamental change has large implications
throughout the changes documented for this version. For more information
about the new core architecture of Asterisk, please see the Asterisk wiki.
- The following channel variables have changed behavior which is described in
the CHANGES file: TRANSFER_CONTEXT, BRIDGEPEER, BRIDGEPVTCALLID,
ATTENDED_TRANSFER_COMPLETE_SOUND, DYNAMIC_FEATURENAME, and DYNAMIC_PEERNAME.
AMI (Asterisk Manager Interface):
- Version 1.4 - The details of what happens to a channel when a masquerade
happens (transfers, parking, etc) have changed.
- The Masquerade event now includes the Uniqueid's of the clone and original
channels.
- Channels no longer swap Uniqueid's as a result of the masquerade.
- Instead of a shell game of renames, there's now a single rename, appending
<ZOMBIE> to the name of the original channel.
- *Major* changes were made to both the syntax as well as the semantics of the
AMI protocol. In particular, AMI events have been substantially modified
and improved in this version of Asterisk. The major event changes are listed
below.
- NewPeerAccount has been removed. NewAccountCode is raised instead.
- Reload events have been consolidated and standardized.
- ModuleLoadReport has been removed.
- FaxSent is now SendFAX; FaxReceived is now ReceiveFAX. This standardizes
app_fax and res_fax events.
- MusicOnHold has been replaced with MusicOnHoldStart and MusicOnHoldStop.
- JabberEvent has been removed.
- Hold is now in the core and will now raise Hold and Unhold events.
- Join is now QueueCallerJoin.
- Leave is now QueueCallerLeave.
- Agentlogin/Agentlogoff is now AgentLogin/AgentLogoff, respectively.
- ChannelUpdate has been removed.
- Local channel optimization is now conveyed via LocalOptimizationBegin and
LocalOptimizationEnd.
- BridgeAction and BridgeExec have been removed.
- BlindTransfer and AttendedTransfer events were added.
- Dial is now DialBegin and DialEnd.
- DTMF is now DTMFBegin and DTMFEnd.
- Bridge has been replaced with BridgeCreate, BridgeEnter, BridgeLeave, and
BridgeDestroy
- MusicOnHold has been replaced with MusicOnHoldStart and MusicOnHoldStop
- AGIExec is now AGIExecStart and AGIExecEnd
- AsyncAGI is now AsyncAGIStart, AsyncAGIExec, and AsyncAGIEnd
- The 'MCID' AMI event now publishes a channel snapshot when available and
its non-channel-snapshot parameters now use either the "MCallerID" or
'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
parameters in the channel snapshot.
- The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
- All AMI events now contain a 'SystemName' field, if available.
- Local channel information in events is now prefixed with 'LocalOne' and
'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
and 'LocalOptimizationEnd' events.
- The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
previous versions. They now report all SR/RR packets sent/received, and
have been restructured to better reflect the data sent in a SR/RR. In
particular, the event structure now supports multiple report blocks.
- The deprecated use of | (pipe) as a separator in the channelvars setting in
manager.conf has been removed.
- The SIP SIPqualifypeer action now sends a response indicating it will qualify
a peer once a peer has been found to qualify. Once the qualify has been
completed it will now issue a SIPqualifypeerdone event.
- The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
in a future release. Please use the common 'Exten' field instead.
- The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
'UnParkedCall' have changed significantly in the new res_parking module.
- The 'Channel' and 'From' headers are gone. For the channel that was parked
or is coming out of parking, a 'Parkee' channel snapshot is issued and it
has a number of fields associated with it. The old 'Channel' header relayed
the same data as the new 'ParkeeChannel' header.
- The 'From' field was ambiguous and changed meaning depending on the event.
for most of these, it was the name of the channel that parked the call
(the 'Parker'). There is no longer a header that provides this channel name,
however the 'ParkerDialString' will contain a dialstring to redial the
device that parked the call.
- On UnParkedCall events, the 'From' header would instead represent the
channel responsible for retrieving the parkee. It receives a channel
snapshot labeled 'Retriever'. The 'from' field is is replaced with
'RetrieverChannel'.
- Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
- The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
fashion has changed the field names 'StartExten' and 'StopExten' to
'StartSpace' and 'StopSpace' respectively.
- The AMI 'Status' response event to the AMI Status action replaces the
'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
indicate what bridge the channel is currently in.
CDR (Call Detail Records)
- Significant changes have been made to the behavior of CDRs. The CDR engine
was effectively rewritten and built on the Stasis message bus. For a full
definition of CDR behavior in Asterisk 12, please read the specification
on the Asterisk wiki (wiki.asterisk.org).
- CDRs will now be created between all participants in a bridge. For each
pair of channels in a bridge, a CDR is created to represent the path of
communication between those two endpoints. This lets an end user choose who
to bill for what during bridge operations with multiple parties.
- The duration, billsec, start, answer, and end times now reflect the times
associated with the current CDR for the channel, as opposed to a cumulative
measurement of all CDRs for that channel.
CEL:
- The Uniqueid field for a channel is now a stable identifier, and will not
change due to transfers, parking, etc.
- CEL has undergone significant rework in Asterisk 12, and is now built on the
Stasis message bus. Please see the specification for CEL on the Asterisk
wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
information. A summary of the affected events is below:
- BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT.
- BLINDTRANSFER/ATTENDEDTRANSFER events now report the peer as NULL and
additional information in the extra string field.
Dialplan Functions:
- Certain dialplan functions have been marked as 'dangerous', and may only be
executed from the dialplan. Execution from extenal sources (AMI's GetVar and
SetVar actions; etc.) may be inhibited by setting live_dangerously in the
[options] section of asterisk.conf to no. SHELL(), channel locking, and
direct file read/write functions are marked as dangerous. DB_DELETE() and
REALTIME_DESTROY() are marked as dangerous for reads, but can now safely
accept writes (which ignore the provided value).
- The default value for live_dangerously was changed from yes (in Asterisk 11
and earlier) to no (in Asterisk 12 and greater).
Dialplan:
- All channel and global variable names are evaluated in a case-sensitive
manner. In previous versions of Asterisk, variables created and evaluated in
the dialplan were evaluated case-insensitively, but built-in variables and
variable evaluation done internally within Asterisk was done
case-sensitively.
- Asterisk has always had code to ignore dash '-' characters that are not
part of a character set in the dialplan extensions. The code now
consistently ignores these characters when matching dialplan extensions.
- BRIDGE_FEATURES channel variable is now casesensitive for feature letter
codes. Uppercase variants apply them to the calling party while lowercase
variants apply them to the called party.
Features:
- The features.conf [applicationmap] <FeatureName> ActivatedBy option is
no longer honored. The feature is always activated by the channel that has
DYNAMIC_FEATURES defined on it when it enters the bridge. Use predial to set
different values of DYNAMIC_FEATURES on the channels
- Executing a dynamic feature on the bridge peer in a multi-party bridge will
execute it on all peers of the activating channel.
- There is no longer an explicit 'features reload' CLI command. Features can
still be reloaded using 'module reload features'.
- It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
features.c for atxferdropcall=no to work properly. This option now just
works.
Parking:
- Parking has been extracted from the Asterisk core as a loadable module,
res_parking.
- Configuration is found in res_parking.conf. It is no longer supported in
features.conf
- The arguments for the Park, ParkedCall, and ParkAndAnnounce applications
have been modified significantly. See the application documents for
specific details.
- Numerous changes to Parking related applications, AMI and CLI commands and
internal inter-workings have been made. Please read the CHANGES file for
the detailed list.
Security Events Framework:
- Security Event timestamps now use ISO 8601 formatted date/time instead of
the "seconds-microseconds" format that it was using previously.
AGENT:
- The password option has been disabled, as the AgentLogin application no
longer provides authentication.
AUDIOHOOK_INHERIT:
- Due to changes in the Asterisk core, this function is no longer needed to
preserve a MixMonitor on a channel during transfer operations and dialplan
execution. It is effectively obsolete.
CDR: (function)
- The 'amaflags' and 'accountcode' attributes for the CDR function are
deprecated. Use the CHANNEL function instead to access these attributes.
- The 'l' option has been removed. When reading a CDR attribute, the most
recent record is always used. When writing a CDR attribute, all non-finalized
CDRs are updated.
- The 'r' option has been removed, for the same reason as the 'l' option.
- The 's' option has been removed, as LOCKED semantics no longer exist in the
CDR engine.
VMCOUNT:
- Mailboxes defined by app_voicemail MUST be referenced by the rest of the
system as mailbox@context. The rest of the system cannot add @default
to mailbox identifiers for app_voicemail that do not specify a context
any longer. It is a mailbox identifier format that should only be
interpreted by app_voicemail.
res_rtp_asterisk:
- ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
them, an Asterisk-specific version of PJSIP needs to be installed.
Tarballs are available from https://github.com/asterisk/pjproject/tags/.
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===
=== Information for upgrading between Asterisk versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also include advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
=== UPGRADE-11.txt -- Upgrade info for 10 to 11
=== UPGRADE-12.txt -- Upgrade info for 11 to 12
===========================================================
General Asterisk Changes:
- The asterisk command line -I option and the asterisk.conf internal_timing
option are removed and always enabled if any timing module is loaded.
- The per console verbose level feature as previously implemented caused a
large performance penalty. The fix required some minor incompatibilities
if the new rasterisk is used to connect to an earlier version. If the new
rasterisk connects to an older Asterisk version then the root console verbose
level is always affected by the "core set verbose" command of the remote
console even though it may appear to only affect the current console. If
an older version of rasterisk connects to the new version then the
"core set verbose" command will have no effect.
- The asterisk compatibility options in asterisk.conf have been removed.
These options enabled certain backwards compatibility features for
pbx_realtime, res_agi, and app_set that made their behaviour similar to
Asterisk 1.4. Users who used these backwards compatibility settings should
update their dialplans to use ',' instead of '|' as a delimiter, and should
use the Set dialplan application instead of the MSet dialplan application.
Build System:
- Sample config files have been moved from configs/ to a subfolder of that
directory, 'samples'.
- The menuselect utility has been pulled into the Asterisk repository. As a
result, the libxml2 development library is now a required dependency for
Asterisk.
- Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
objects will emit additional debug information to the refs log file located
in the standard Asterisk log file directory. This log file is useful in
tracking down object leaks and other reference counting issues. Prior to
this version, this option was only available by modifying the source code
directly. This change also includes a new script, refcounter.py, in the
contrib folder that will process the refs log file.
Applications:
ConfBridge:
- The sound_place_into_conference sound used in Confbridge is now deprecated
and is no longer functional since it has been broken since its inception
and the fix involved using a different method to achieve the same goal. The
new method to achieve this functionality is by using sound_begin to play
a sound to the conference when waitmarked users are moved into the conference.
- Added 'Admin' header to ConfbridgeJoin, ConfbridgeLeave, ConfbridgeMute,
ConfbridgeUnmute, and ConfbridgeTalking AMI events.
ControlPlayback:
- The ControlPlayback and 'control stream file' AGI command will no longer
implicitly answer the channel. If you do not answer the channel prior to
using either this application or AGI command, you must send Progress
first.
Queue:
- Queue rules provided in queuerules.conf can no longer be named "general".
SetMusicOnHold:
- The SetMusicOnHold dialplan application was deprecated and has been removed.
Users of the application should use the CHANNEL function's musicclass
setting instead.
WaitMusicOnHold:
- The WaitMusicOnHold dialplan application was deprecated and has been
removed. Users of the application should use MusicOnHold with a duration
parameter instead.
CDR Backends:
- The cdr_sqlite module was deprecated and has been removed. Users of this
module should use the cdr_sqlite3_custom module instead.
Channel Drivers:
chan_dahdi:
- SS7 support now requires libss7 v2.0 or later.
- Added the inband_on_setup_ack compatibility option to chan_dahdi.conf to
deal with switches that don't send an inband progress indication in the
SETUP ACKNOWLEDGE message.
Default is now no.
chan_gtalk
- This module was deprecated and has been removed. Users of chan_gtalk
should use chan_motif.
chan_h323
- This module was deprecated and has been removed. Users of chan_h323
should use chan_ooh323.
chan_jingle
- This module was deprecated and has been removed. Users of chan_jingle
should use chan_motif.
chan_pjsip:
- Added a 'force_avp' option to chan_pjsip which will force the usage of
'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' as the media transport type
in SDP offers depending on settings, even when DTLS is used for media
encryption.
- Added a 'media_use_received_transport' option to chan_pjsip which will
cause the SDP answer to use the media transport as received in the SDP
offer.
chan_sip:
- Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip
interoperability.
- The SIPPEER dialplan function no longer supports using a colon as a
delimiter for parameters. The parameters for the function should be
delimited using a comma.
- The SIPCHANINFO dialplan function was deprecated and has been removed. Users
of the function should use the CHANNEL function instead.
- Added a 'force_avp' option for chan_sip. When enabled this option will
cause the media transport in the offer or answer SDP to be 'RTP/AVP',
'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' even if a DTLS stream has been
configured. This option can be set to improve interoperability with WebRTC
clients that don't use the RFC defined transport for DTLS.
- The 'dtlsverify' option in chan_sip now has additional values besides
'yes' and 'no'. If 'yes' is specified both the certificate and fingerprint
will be verified. If 'no' is specified then neither the certificate or
fingerprint is verified. If 'certificate' is specified then only the
certificate is verified. If 'fingerprint' is specified then only the
fingerprint is verified.
- A 'dtlsfingerprint' option has been added to chan_sip which allows the
hash to be specified for the DTLS fingerprint placed in SDP. Supported
values are 'sha-1' and 'sha-256' with 'sha-256' being the default.
- The 'progressinband=never' option is now more zealous in the persecution of
progress messages coming from Asterisk. Channels bridged with a SIP channel
that has 'progressinband=never' set will not be able to forward their
progress indications through to the SIP device. chan_sip will now turn such
progress indications into a 180 Ringing (if a 180 has not yet been
transmitted) if 'progressinband=never'.
- The codec preference order in an SDP during an offer is slightly different
than previous releases. Prior to Asterisk 13, the preference order of
codecs used to be:
(1) Our preferred codec
(2) Our configured codecs
(3) Any non-audio joint codecs
One of the ways the new media format architecture in Asterisk 13 improves
performance is by reference counting formats, such that they can be reused
in many places without additional allocation. To not require a large
amount of locking, an instance of a format is immutable by convention.
This works well except for formats with attributes. Since a media format
with an attribute is a different object than the same format without an
attribute, we have to carry over the formats with attributes from an
inbound offer so that the correct attributes are offered in an outgoing
INVITE request. This requires some subtle tweaks to the preference order
to ensure that the media format with attributes is offered to a remote
peer, as opposed to the same media format (but without attributes) that
may be stored in the peer object.
All of this means that our offer offer list will now be:
(1) Our preferred codec
(2) Any joint codecs offered by the inbound offer
(3) All other codecs that are not the preferred codec and not a joint
codec offered by the inbound offer
chan_unistim:
- The unistim.conf 'dateformat' has changed meaning of options values to conform
values used inside Unistim protocol
- Added 'dtmf_duration' option with changing default operation to disable
receivied dtmf playback on unistim phone
Core:
Account Codes:
- accountcode behavior changed somewhat to add functional peeraccount
support. The main change is that local channels now cross accountcode
and peeraccount across the special bridge between the ;1 and ;2 channels
just like channels between normal bridges. See the CHANGES file for
more information.
ARI:
- The ARI version has been changed to 1.5.0. This is to reflect backwards
compatible changes made since 12.0.0 was released.
- Added a new ARI resource 'mailboxes' which allows the creation and
modification of mailboxes managed by external MWI. Modules res_mwi_external
and res_stasis_mailbox must be enabled to use this resource.
- Added new events for externally initiated transfers. The event
BridgeBlindTransfer is now raised when a channel initiates a blind transfer
of a bridge in the ARI controlled application to the dialplan; the
BridgeAttendedTransfer event is raised when a channel initiates an
attended transfer of a bridge in the ARI controlled application to the
dialplan.
- Channel variables may now be specified as a body parameter to the
POST /channels operation. The 'variables' key in the JSON is interpreted
as a sequence of key/value pairs that will be added to the created channel
as channel variables. Other parameters in the JSON body are treated as
query parameters of the same name.
- A bug fix in bridge creation has caused a behavioural change in how
subscriptions are created for bridges. A bridge created through ARI, does
not, by itself, have a subscription created for any particular Stasis
application. When a channel in a Stasis application joins a bridge, an
implicit event subscription is created for that bridge as well. Previously,
when a channel left such a bridge, the subscription was leaked; this allowed
for later bridge events to continue to be pushed to the subscribed
applications. That leak has been fixed; as a result, bridge events that were
delivered after a channel left the bridge are no longer delivered. An
application must subscribe to a bridge through the applications resource if
it wishes to receive all events related to a bridge.
AMI:
- The AMI version has been changed to 2.5.0. This is to reflect backwards
compatible changes made since 12.0.0 was released.
- The DialStatus field in the DialEnd event can now have additional values.
This includes ABORT, CONTINUE, and GOTO.
- The res_mwi_external_ami module can, if loaded, provide additional AMI
actions and events that convey MWI state within Asterisk. This includes
the MWIGet, MWIUpdate, and MWIDelete actions, as well as the MWIGet and
MWIGetComplete events that occur in response to an MWIGet action.
- AMI now contains a new class authorization, 'security'. This is used with
the following new events: FailedACL, InvalidAccountID, SessionLimit,
MemoryLimit, LoadAverageLimit, RequestNotAllowed, AuthMethodNotAllowed,
RequestBadFormat, SuccessfulAuth, UnexpectedAddress, ChallengeResponseFailed,
InvalidPassword, ChallengeSent, and InvalidTransport.
- Bridge related events now have two additional fields: BridgeName and
BridgeCreator. BridgeName is a descriptive name for the bridge;
BridgeCreator is the name of the entity that created the bridge. This
affects the following events: ConfbridgeStart, ConfbridgeEnd,
ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
- MixMonitor AMI actions now require users to have authorization classes.
* MixMonitor - system
* MixMonitorMute - call or system
* StopMixMonitor - call or system
- Removed the undocumented manager.conf block-sockets option. It interferes with
TCP/TLS inactivity timeouts.
- The response to the PresenceState AMI action has historically contained two
Message keys. The first of these is used as an informative message regarding
the success/failure of the action; the second contains a Presence state
specific message. Having two keys with the same unique name in an AMI
message is cumbersome for some client; hence, the Presence specific Message
has been deprecated. The message will now contain a PresenceMessage key
for the presence specific information; the Message key containing presence
information will be removed in the next major version of AMI.
- The manager.conf 'eventfilter' now takes an "extended" regular expression
instead of a "basic" one.
CDRs:
- The "endbeforehexten" setting now defaults to "yes", instead of "no".
When set to "no", yhis setting will cause a new CDR to be generated when a
channel enters into hangup logic (either the 'h' extension or a hangup
handler subroutine). In general, this is not the preferred default: this
causes extra CDRs to be generated for a channel in many common dialplans.
CLI commands:
- "core show settings" now lists the current console verbosity in addition
to the root console verbosity.
- "core set verbose" has not been able to support the by module verbose
logging levels since verbose logging levels were made per console. That
syntax is now removed and a silence option added in its place.
Logging:
- The 'verbose' setting in logger.conf still takes an optional argument,
specifying the verbosity level for each logging destination. However,
the default is now to once again follow the current root console level.
As a result, using the AMI Command action with "core set verbose" could
again set the root console verbose level and affect the verbose level
logged.
HTTP:
- Added http.conf session_inactivity timer option to close HTTP connections
that aren't doing anything.
- Added support for persistent HTTP connections. To enable persistent
HTTP connections configure the keep alive time between HTTP requests. The
keep alive time between HTTP requests is configured in http.conf with the
session_keep_alive parameter.
Realtime Configuration:
- WARNING: The database migration script that adds the 'extensions' table for
realtime had to be modified due to an error when installing for MySQL. The
'extensions' table's 'id' column was changed to be a primary key. This could
potentially cause a migration problem. If so, it may be necessary to
manually alter the affected table/column to bring it back in line with the
migration scripts.
- New columns have been added to realtime tables for 'support_path' on
ps_registrations and ps_aors and for 'path' on ps_contacts for the new
SIP Path support in chan_pjsip.
- The following new tables have been added for pjsip realtime: 'ps_systems',
'ps_globals', 'ps_tranports', 'ps_registrations'.
- The following columns were added to the 'ps_aors' realtime table:
'maximum_expiration', 'outbound_proxy', and 'support_path'.
- The following columns were added to the 'ps_contacts' realtime table:
'outbound_proxy', 'user_agent', and 'path'.
- New columns have been added to the ps_endpoints realtime table for the
'media_address', 'redirect_method' and 'set_var' options. Also the
'mwi_fromuser' column was renamed to 'mwi_from_user'. A new column
'message_context' was added to let users configure how MESSAGE requests are
routed to the dialplan.
- A new column was added to the 'ps_globals' realtime table for the 'debug'
option.
- PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from
yes/no enumerators to string values. 'cos_audio' and 'cos_video' have been
changed from yes/no enumerators to integer values. PJSIP transport column
'tos' has been changed from a yes/no enumerator to a string value. 'cos' has
been changed from a yes/no enumerator to an integer value.
- The 'queues' and 'queue_members' realtime tables have been added to the
config Alembic scripts.
- A new set of Alembic scripts has been added for CDR tables. This will create
a 'cdr' table with the default schema that Asterisk expects.
- A new upgrade script has been added that adds a 'queue_rules' table for
app_queue. Users of app_queue can store queue rules in a database. It is
important to note that app_queue only looks for this table on module load or
module reload; for more information, see the CHANGES file.
Resources:
res_odbc:
- The compatibility setting, allow_empty_string_in_nontext, has been removed.
Empty column values will be stored as empty strings during realtime updates.
res_jabber:
- This module was deprecated and has been removed. Users of this module should
use res_xmpp instead.
res_http_websocket:
- Added a compatibility option to ari.conf, sip.conf, and pjsip.conf
'websocket_write_timeout'. When a websocket connection exists where Asterisk
writes a substantial amount of data to the connected client, and the connected
client is slow to process the received data, the socket may be disconnected.
In such cases, it may be necessary to adjust this value.
Default is 100 ms.
Scripts:
safe_asterisk:
- The safe_asterisk script was previously not installed on top of an existing
version. This caused bug-fixes in that script not to be deployed. If your
safe_asterisk script is customized, be sure to keep your changes. Custom
values for variables should be created in *.sh file(s) inside
ASTETCDIR/startup.d/. See ASTERISK-21965.
- Changed a log message in safe_asterisk and the $NOTIFY mail subject. If
you use tools to parse either of them, update your parse functions
accordingly. The changed strings are:
- "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
- "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"
Utilities:
- The refcounter program has been removed in favor of the refcounter.py script
in contrib/scripts.
===========================================================
===========================================================

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===========================================================
===
=== Information for upgrading between Asterisk versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also include advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
=== UPGRADE-11.txt -- Upgrade info for 10 to 11
=== UPGRADE-12.txt -- Upgrade info for 11 to 12
=== UPGRADE-13.txt -- Upgrade info for 12 to 13
===========================================================
From 14.6.0 to 14.7.0:
Core:
- ast_app_parse_timelen now returns an error if it encounters extra characters
at the end of the string to be parsed.
From 14.4.0 to 14.5.0:
Core:
- Support for embedded modules has been removed. This has not worked in
many years. LOADABLE_MODULES menuselect option is also removed as
loadable module support is now always enabled.
From 14.3.0 to 14.4.0:
res_rtp_asterisk:
- The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
Data and Control Packets on a Single Port." For the PJSIP channel driver,
chan_pjsip, you can set "rtcp_mux = yes" on a PJSIP endpoint in pjsip.conf
to enable the feature. For chan_sip you can set "rtcp_mux = yes" either
globally or on a per-peer basis in sip.conf.
New in 14.0.0
ARI:
- The policy for when to send "Dial" events has changed. Previously, "Dial"
events were sent on the calling channel's topic. However, starting in Asterisk
14, if there is no calling channel on which to send the event, the event is
instead sent on the called channel's topic. Note that for the ARI channels
resource's dial operation, this means that the "Dial" events will always be
sent on the called channel's topic.
Channel Drivers:
chan_dahdi:
- For users using the FXO port (FXS signaling) distinctive ring detection
feature, you will need to adjust the dringX count values. The count
values now only record ring end events instead of any DAHDI event. A
ring-ring-ring pattern would exceed the pattern limits and stop
Caller-ID detection.
chan_sip:
- The SIP dial string has been extended past the [!dnid] option by another
exclamation mark: [!dnid[!fromuri]. An exclamation mark in the To-URI
will now mean changes to the From-URI.
Core:
- The REF_DEBUG compiler flag is now used to enable refdebug by default.
The setting can be overridden in asterisk.conf by setting refdebug in
the options category. No recompile is required to enable/disable it.
- Modified processing of command-line options to first parse only what
is necessary to read asterisk.conf. Once asterisk.conf is fully loaded,
the remaining options are processed. The -X option now applies to
asterisk.conf only. To enable #exec for other config files you must
set execincludes=yes in asterisk.conf. Any other option set on the
command-line will now override the equivalent setting from asterisk.conf.
AMI:
- The 'ModuleCheck' Action's Version key will no longer show the module
version. The value will always be blank.
CLI:
- The 'core show file version' command has been removed. When Asterisk
moved to Git, the source control version support was removed. As a
result, the CLi command was no longer useful and was removed as well.
Logging:
- The first callid created is now 1 instead of 0. The value 0
is now reserved to represent a lack of callid.
AMI:
- The Command action now sends the output from the CLI command as a series
of Output headers for each line instead of as a block of text with the
--END COMMAND-- delimiter to match the output from other actions.
Commands that fail to execute (no such command, invalid syntax etc.) now
return an Error response instead of Success.
app_amd:
- The 'maximum_number_of_words' configuration option and parameter to the AMD
application previously did not match the documented functionality + variable
name. In Asterisk 13, a value of '3' would mean that if '3' words were detected,
the result would be detection as a 'MACHINE'. As of this version, the value
reflects the maximum words that if EXCEEDED (rather than reached), would
result in detection as a machine. This means that you should update this
value to be one higher than your previos value, if your previous value
was working well for you.
===========================================================
===========================================================

63
UPGRADE-15.txt
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===========================================================
===
=== Information for upgrading between Asterisk versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also include advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
=== UPGRADE-11.txt -- Upgrade info for 10 to 11
=== UPGRADE-12.txt -- Upgrade info for 11 to 12
=== UPGRADE-13.txt -- Upgrade info for 12 to 13
=== UPGRADE-14.txt -- Upgrade info for 13 to 14
===========================================================
From 15.2.0 to 15.3.0:
res_pjsip
------------------
* Users who are matching endpoints by SIP header need to reevaluate their
global "endpoint_identifier_order" option in light of the "ip" endpoint
identifier method split into the "ip" and "header" endpoint identifier
methods.
res_pjsip_endpoint_identifier_ip
------------------
* The endpoint identifier "ip" method previously recognized endpoints either
by IP address or a matching SIP header. The "ip" endpoint identifier method
is now split into the "ip" and "header" endpoint identifier methods. The
"ip" endpoint identifier method only matches by IP address and the "header"
endpoint identifier method only matches by SIP header. The split allows the
user to control the relative priority of the IP address and the SIP header
identification methods in the global "endpoint_identifier_order" option.
e.g., If you have two type=identify sections where one matches by IP address
for endpoint alice and the other matches by SIP header for endpoint bob then
you can now predict which endpoint is matched when a request comes in that
matches both.
New in 15.0.0:
Build System:
- '--with-pjproject-bundled' is now the default when running ./configure
It can be disabled with '--without-pjproject-bundled'.
Core:
- Multi-stream support has been added so a channel can have multiple
streams of the same type such as audio and video.
- The 'Data Retrieval API' has been removed. This API was not actively
maintained, was not added to new modules (such as res_pjsip), and there
exist better alternatives to acquire the same information, such as the
ARI. As a result, the 'DataGet' AMI action as well as the 'data get'
CLI command have been removed.

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UPGRADE-16.txt
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===========================================================
===
=== Information for upgrading between Asterisk versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also include advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
=== UPGRADE-11.txt -- Upgrade info for 10 to 11
=== UPGRADE-12.txt -- Upgrade info for 11 to 12
=== UPGRADE-13.txt -- Upgrade info for 12 to 13
=== UPGRADE-14.txt -- Upgrade info for 13 to 14
=== UPGRADE-15.txt -- Upgrade info for 14 to 15
===========================================================
New in 16.0.0:
app_fax:
- The app_fax module is now deprecated, users should migrate to the
replacement module res_fax.
app_macro:
- The app_macro module is now deprecated and by default it is no longer
built. Users should migrate to app_stack (Gosub). A warning is logged
the first time any Macro is used.
AMI:
- The ContactStatus and Status fields for the manager events ContactStatus
and ContactStatusDetail are now set to "NonQualified" when a contact exists
but has not been qualified.
- The ContactStatus event will no longer be sent by PJSIP when a device
refreshes its registration.
- The "Newexten" event is now part of the "dialplan" class. The documentation
for Asterisk 15 already specified this, but the implementation was actually
using the "call" class instead.
ARI:
- The ContactInfo event's contact_status field is now set to "NonQualified"
when a contact exists but has not been qualified.
Build System:
- MALLOC_DEBUG no longer has an effect on Asterisk's ABI. Asterisk built
with MALLOC_DEBUG can now successfully load binary modules built without
MALLOC_DEBUG and vice versa. Third-party pre-compiled modules no longer
need to have a special build with it enabled.
- Asterisk now depends on libjansson >= 2.11. If this version is not
available on your distro you can use `./configure --with-jansson-bundled`.
chan_dahdi:
- Timeouts for reading digits from analog phones are now configurable in
chan_dahdi.conf: firstdigit_timeout, interdigit_timeout, matchdigit_timeout.
cdr_syslog:
- The cdr_syslog module is now deprecated and by default it is no longer
built.
res_config_sqlite:
- The res_config_sqlite module is now deprecated, users should migrate to the
replacement module res_config_sqlite3.
res_monitor:
- The res_monitor module is now deprecated, users should migrate to the
replacement module app_mixmonitor.
Core:
- libedit is no longer available as an embedded library and must be provided
by the system.
- The module loader now enforces inter-module dependencies. This ensures that
a module is not started before another it depends on, even if preload is used.
If a dependency is not available or fails to startup this will block any
dependants from startup.
- Parts of the Asterisk core which can load configuration from realtime are now
built-in modules. It is no longer necessary to preload realtime drivers as
they are always initialized before the built-in modules.

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28
doc/CHANGES-staging/README.md
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@ -5,24 +5,28 @@ used by the release script to update the CHANGES file automatically. The only
time that it is necessary to add something to the CHANGES-staging directory is
if you are either adding a new feature to Asterisk or adding new functionality
to an existing feature. The file does not need to have a meaningful name, but
it probably should. If there are multiple items that need documenting, each can
be separated with a subject line, which should always start with "Subject:",
followed by the subject of the change. This is case sensitive! For example, if
you are making a change to PJSIP, then you might add the file
"res_pjsip_my_cool_feature" to this directory, with a short description of what
it does. If you are adding multiple entries, they should be done in the same
commit to avoid merge conflicts. Here's an example:
it probably should. If there are multiple items that need documenting, you can
add multiple files, each with their own description. If the message is going to
be the same for each subject, then you can add multiple subject headers to one
file. The "Subject: xxx" line is case sensitive! For example, if you are making
a change to PJSIP, then you might add the file "res_pjsip_my_cool_feature" to
this directory, with a short description of what it does. If you are adding
multiple entries, they should be done in the same commit to avoid merge
conflicts. Here's an example:
> Subject: res_pjsip
> Subject: Core
>
> Here's a pretty good description of my new feature that explains exactly what
> it does and how to use it.
Here's a master-only example:
> Subject: res_ari
> Master-Only: True
>
> Subject: core
> Master-Only: true
>
> Here's another description of something else I added that is a big enough
> change to warrant another entry in the CHANGES file.
> This change will only go into the master branch. The "Master-Only" header
> will never be in a change not in master.
Note that the second subject has another header: "Master-Only". Changes that go
into the master branch and ONLY the master branch are the only ones that should

36
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Subject: Bridging
Master-Only: true
The bridging core no longer uses the stasis cache for bridge
snapshots. The latest bridge snapshot is now stored on the
ast_bridge structure itself.
The following APIs are no longer available since the stasis cache
is no longer used:
ast_bridge_topic_cached()
ast_bridge_topic_all_cached()
A topic pool is now used for individual bridge topics.
The ast_bridge_cache() function was removed since there's no
longer a separate container of snapshots.
A new function "ast_bridges()" was created to retrieve the
container of all bridges. Users formerly calling
ast_bridge_cache() can use the new function to iterate over
bridges and retrieve the latest snapshot directly from the
bridge.
The ast_bridge_snapshot_get_latest() function was renamed to
ast_bridge_get_snapshot_by_uniqueid().
A new function "ast_bridge_get_snapshot()" was created to retrieve
the bridge snapshot directly from the bridge structure.
The ast_bridge_topic_all() function now returns a normal topic
not a cached one so you can't use stasis cache functions on it
either.
The ast_bridge_snapshot_type() stasis message now has the
ast_bridge_snapshot_update structure as it's data. It contains
the last snapshot and the new one.

7
doc/CHANGES-staging/chan_sip_deprecated.txt
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@ -0,0 +1,7 @@
Subject: chan_sip
Master-Only: true
The chan_sip module is now deprecated, users should migrate to the
replacement module chan_pjsip. See guides at the Asterisk Wiki:
https://wiki.asterisk.org/wiki/x/tAHOAQ
https://wiki.asterisk.org/wiki/x/hYCLAQ

16
doc/CHANGES-staging/channels_stasis_cache.txt
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Subject: Channels
Master-Only: true
The core no longer uses the stasis cache for channels snapshots.
The following APIs are no longer available:
ast_channel_topic_cached()
ast_channel_topic_all_cached()
The ast_channel_cache_all() and ast_channel_cache_by_name() functions
now returns an ao2_container of ast_channel_snapshots rather than a
container of stasis_messages therefore you can't call stasis_cache
functions on it.
The ast_channel_topic_all() function now returns a normal topic,
not a cached one so you can't use stasis cache functions on it either.
The ast_channel_snapshot_type() stasis message now has the
ast_channel_snapshot_update structure as it's data.
ast_channel_snapshot_get_latest() still returns the latest snapshot.

30
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@ -5,23 +5,27 @@ used by the release script to update the UPGRADE.txt file automatically. The
only time that it is necessary to add something to the UPGRADE-staging directory
is if you are making a breaking change to an existing feature in Asterisk. The
file does not need to have a meaningful name, but it probably should. If there
are multiple items that need documenting, each can be separated with a subject
line, which should always start with "Subject:", followed by the subject of the
change. This is case sensitive! For example, if you are making a change to PJSIP,
then you might add the file "res_pjsip_breaking_change" to this directory, with
a short description of what it does. If you are adding multiple entries, they
should be done in the same commit to avoid merge conflicts. Here's an example:
are multiple items that need documenting, you can add multiple files, each with
their own description. If the message is going to be the same for each subject,
then you can add multiple subject headers to one file. The "Subject: xxx" line
is case sensitive! For example, if you are making a change to PJSIP, then you
might add the file "res_pjsip_my_cool_feature" to this directory, with a short
description of what it does. If you are adding multiple entries, they should be
done in the same commit to avoid merge conflicts. Here's an example:
> Subject: res_pjsip
> Subject: Core
>
> Here's a pretty good description of what I changed that explains exactly what
> it does and why it breaks things (and why they needed to be broken).
>
> Subject: core
> Master-Only: true
> Here's a pretty good description of my new feature that explains exactly what
> it does and how to use it.
Here's a master-only example:
> Subject: res_ari
> Master-Only: True
>
> Here's another description of something else I added that is a big enough
> change to warrant another entry in the UPDATE.txt file.
> This change will only go into the master branch. The "Master-Only" header
> will never be in a change not in master.
Note that the second subject has another header: "Master-Only". Changes that go
into the master branch and ONLY the master branch are the only ones that should

4
doc/UPGRADE-staging/applications_jabberstatus.txt
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@ -0,0 +1,4 @@
Subject: Applications
Master-Only: true
The JabberStatus application, deprecated in Asterisk 12, has been removed.

36
doc/UPGRADE-staging/bridging_stasis_cache.txt
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@ -0,0 +1,36 @@
Subject: Bridging
Master-Only: true
The bridging core no longer uses the stasis cache for bridge
snapshots. The latest bridge snapshot is now stored on the
ast_bridge structure itself.
The following APIs are no longer available since the stasis cache
is no longer used:
ast_bridge_topic_cached()
ast_bridge_topic_all_cached()
A topic pool is now used for individual bridge topics.
The ast_bridge_cache() function was removed since there's no
longer a separate container of snapshots.
A new function "ast_bridges()" was created to retrieve the
container of all bridges. Users formerly calling
ast_bridge_cache() can use the new function to iterate over
bridges and retrieve the latest snapshot directly from the
bridge.
The ast_bridge_snapshot_get_latest() function was renamed to
ast_bridge_get_snapshot_by_uniqueid().
A new function "ast_bridge_get_snapshot()" was created to retrieve
the bridge snapshot directly from the bridge structure.
The ast_bridge_topic_all() function now returns a normal topic
not a cached one so you can't use stasis cache functions on it
either.
The ast_bridge_snapshot_type() stasis message now has the
ast_bridge_snapshot_update structure as it's data. It contains
the last snapshot and the new one.

7
doc/UPGRADE-staging/chan_sip_deprecated.txt
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@ -0,0 +1,7 @@
Subject: chan_sip
Master-Only: true
The chan_sip module is now deprecated, users should migrate to the
replacement module chan_pjsip. See guides at the Asterisk Wiki:
https://wiki.asterisk.org/wiki/x/tAHOAQ
https://wiki.asterisk.org/wiki/x/hYCLAQ

16
doc/UPGRADE-staging/channels_stasis_cache.txt
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@ -0,0 +1,16 @@
Subject: Channels
Master-Only: true
The core no longer uses the stasis cache for channels snapshots.
The following APIs are no longer available:
ast_channel_topic_cached()
ast_channel_topic_all_cached()
The ast_channel_cache_all() and ast_channel_cache_by_name() functions
now returns an ao2_container of ast_channel_snapshots rather than a
container of stasis_messages therefore you can't call stasis_cache
functions on it.
The ast_channel_topic_all() function now returns a normal topic,
not a cached one so you can't use stasis cache functions on it either.
The ast_channel_snapshot_type() stasis message now has the
ast_channel_snapshot_update structure as it's data.
ast_channel_snapshot_get_latest() still returns the latest snapshot.

5
doc/UPGRADE-staging/func_callerid_callerpres.txt
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@ -0,0 +1,5 @@
Subject: func_callerid
Master-Only: true
The CALLERPRES() dialplan function, deprecated in Asterisk 1.8, has been
removed.

5
doc/UPGRADE-staging/res_parking_parkingslot.txt
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@ -0,0 +1,5 @@
Subject: res_parking
Master-Only: true
The PARKINGSLOT channel variable, deprecated in Asterisk 12 in favor of the
PARKING_SPACE channel variable, will no longer be set.

4
doc/UPGRADE-staging/res_xmpp_jabberstatus.txt
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@ -0,0 +1,4 @@
Subject: res_xmpp
Master-Only: true
The JabberStatus application, deprecated in Asterisk 12, has been removed.
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