/
UPGRADE.txt
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UPGRADE.txt
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===========================================================
===
=== THIS FILE IS AUTOMATICALLY GENERATED DURING THE RELEASE
=== PROCESS. DO NOT MAKE CHANGES HERE. INSTEAD, REFER TO
=== doc/UPGRADE-staging/README.md FOR MORE DETAILS.
===
=== Information for upgrading between Asterisk versions
===
=== This file documents all the changes that MUST be taken
=== into account when upgrading between certain Asterisk
=== versions. 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. This file 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.
===
===========================================================
------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 20.0.0 --------------------------
------------------------------------------------------------------------------
res_monitor
------------------
* This module is no longer built by default in
accordance with the Module Deprecation Policy.
If you require this functionality you will need
to enable it for building in menuselect. Note
that in the future res_monitor will be removed.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
------------------------------------------------------------------------------
AMI
------------------
* The XML Manager Event Interface (amxml) now generates attribute names
that are compliant with the XML 1.1 specification. Previously, an
attribute name that started with a digit would be rendered as-is, even
though attribute names must not begin with a digit. We now prefix
attribute names that start with a digit with an underscore ('_') to
prevent XML validation failures.
STIR/SHAKEN
------------------
* The STIR/SHAKEN configuration option has been split into
4 different choices: off, attest, verify, and on. Off and
on behave the same way as before. Attest will only perform
attestation on the endpoint, and verify will only perform
verification on the endpoint.
chan_iax2
------------------
* Encryption is now supported for RSA authentication.
Currently, these auth configurations will cause a crash:
auth = md5,rsa
auth = plaintext,md5,rsa
With a patched peer, the following will cause a crash:
auth = rsa
auth = md5,rsa
auth = plaintext,md5,rsa
If both the peer and user are patches, no crash occurs.
Existing good configurations should continue to work.
res_http_media_cache
------------------
* When fetching a file for playback from a URL, Asterisk will now first
use the value of the Content-Type header in the HTTP response to
determine the format of the audio data, and only if it is unable to do
that will it attempt to parse the URL and extract the extension from
the path portion. Previously Asterisk would first look at the end of
the URL, which may have included query string parameters or a URL
fragment, which was error prone.
res_pjsip
------------------
* The 'async_operations' setting on transports is no longer
obeyed and instead is always set to 1. This is due to the
functionality not being applicable to Asterisk and causing
excess unnecessary memory usage. This setting will now be
ignored but can also be removed from the configuration file.
------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 19.0.0 --------------------------
------------------------------------------------------------------------------
Log Rotate
------------------
* The sample logger files have been changed to have .log as their file
extension. This was done so that when attached to issues on the issue
tracker, they are able to be opened in the browser for convenience.
Because of this, the asterisk.logrotate script has been updated to look
for .log extensions instead of no extension for files such as full
and messages.
chan_sip
------------------
* chan_sip is no longer built by default. To build it, make sure to
enable it when running 'make menuselect'
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.0.0 to Asterisk 19.0.0 ------------
------------------------------------------------------------------------------
STIR/SHAKEN
------------------
* The configuration option public_key_url in stir_shaken.conf
has been renamed to public_cert_url to better fit what it
contains. Only the name has changed - functionality is the
same.
* STIR/SHAKEN originally needed an origid to be specified in
stir_shaken.conf under the certificate config object in
order to work. Now, one is automatically created by
generating a UUID, as recommended by RFC8588. Any origid
you have in your stir_shaken.conf will need to be removed
for the module to read in certificates.
menuselect
------------------
* menuselect --enable, --disable, --enable-category and --disable-category will
now fail with a non-zero exit code instead of silently failing if an invalid
option or category is specified.
res_srtp
------------------
* SRTP replay protection has been added to res_srtp and
a new configuration option "srtpreplayprotection" has
been added to the rtp.conf config file. For security
reasons, the default setting is "yes". Buggy clients
may not handle this correctly which could result in
no, or one way, audio and Asterisk error messages like
"replay check failed".
------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 18.0.0 --------------------------
------------------------------------------------------------------------------
Core
------------------
* The ast_format_cap_from_stream_topology() function has been renamed
to ast_stream_topology_get_formats().
app_bridgeaddchan
------------------
* The BridgeAdd application now behaves more like the Bridge application.
The application now sets the BRIDGERESULT channel variable to indicate
what happened when the channel resumes in dialplan. This is instead of
hanging up the channel on failure conditions.
app_mixmonitor
------------------
* In Asterisk 13.29, a new option flag was added to MixMonitor (the 'S'
option) that when combined with the r() or t() options would inject
silence into these files if audio was going to be written to one and
not that other. This allowed the files specified by r() and t() to
subsequently be mixed outside of Asterisk and be appropriately
synchronized. This behavior is now the default, and a new option has
been added to disable this behavior if desired (the 'n' option).
app_queue
------------------
* The 'Reason' header in the QueueMemberPause AMI Event has been
removed. The 'PausedReason' header should be used instead.
* If they are not specified in [general], "shared_lastcall" and "autofill"
now always default to OFF. Before this version, they would be off ('no') if
queues.conf did not have a [general] section, but on ('yes') if it did.
app_voicemail
------------------
* The MessageExists dialplan application and the MESSAGE_EXISTS dialplan
function were removed. The were deprecated in Asterisk 1.6.0 and
Asterisk 11.0.0 respectively. The VM_INFO() dialplan function is the
supported mechanism to query the status of a given mailbox.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
------------------------------------------------------------------------------
AMI
------------------
* The AMI Originate action, which optionally takes a dialplan application as
an argument, no longer accepts "Originate" as the application due to
security concerns.
ARI
------------------
* The "TextMessageReceived" event used to include a list of "TextMessageVariable"
objects as part of its output. Due to a couple of bugs in Asterisk a list of
received variables was never included even if ones were available. However,
variables set to send would be (which they should have not been), but would
fail validation due to the bad formatting.
So basically there was no way to get a "TextMessageReceived" event with
variables. Due to this the API has changed. The "TextMessageVariable" object
no longer exists. "TextMessageReceived" now returns a JSON object of key/value
pairs. So for instance instead of a list of "TextMessageVariable" objects:
[ TextMessageVariable, TextMessageVariable, TextMessageVariable]
where a TextMessageVariable was supposed to be:
{ "key": "<var name>", "value":, "<var value>" }
The output is now just:
{ "<var name>": "<var value>" }
This aligns more with how variables are specified when sending a message, as
well as other variable lists in ARI.
Core
------------------
* The streams API function ast_stream_get_formats is
now defined as returning the format capabilities const.
This has always been the case but was never enforced
through the API itself. Any consumer of this API that
is not treating the formats as immutable should update
their code to create a new format capabilities and set
it on the stream instead.
res_stasis
------------------
* The "TextMessageReceived" event used to include a list of "TextMessageVariable"
objects as part of its output. Due to a couple of bugs in Asterisk a list of
received variables was never included even if ones were available. However,
variables set to send would be (which they should have not been), but would
fail validation due to the bad formatting.
So basically there was no way to get a "TextMessageReceived" event with
variables. Due to this the API has changed. The "TextMessageVariable" object
no longer exists. "TextMessageReceived" now returns a JSON object of key/value
pairs. So for instance instead of a list of "TextMessageVariable" objects:
[ TextMessageVariable, TextMessageVariable, TextMessageVariable]
where a TextMessageVariable was supposed to be:
{ "key": "<var name>", "value":, "<var value>" }
The output is now just:
{ "<var name>": "<var value>" }
This aligns more with how variables are specified when sending a message, as
well as other variable lists in ARI.
res_stir_shaken
------------------
* A new directory has been added under the default (e.g., /var/lib/asterisk) -
inside the 'keys' directory - named 'stir_shaken'. This directory will
hold public keys that have been downloaded for STIR/SHAKEN verification.
------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 17.0.0 --------------------------
------------------------------------------------------------------------------
Applications
------------------
* The JabberStatus application, deprecated in Asterisk 12, has been removed.
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.
Build
------------------
* Asterisk headers are no longer installed and uninstalled automatically when
performing a "make install" or a "make uninstall". To install/uninstall the
headers, use "make install-headers" and "make uninstall-headers". The headers
also continue to be uninstalled when performing a "make uninstall-all".
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.
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
func_callerid
------------------
* The CALLERPRES() dialplan function, deprecated in Asterisk 1.8, has been
removed.
res_parking
------------------
* The PARKINGSLOT channel variable, deprecated in Asterisk 12 in favor of the
PARKING_SPACE channel variable, will no longer be set.
res_xmpp
------------------
* The JabberStatus application, deprecated in Asterisk 12, has been removed.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.0.0 to Asterisk 17.0.0 ------------
------------------------------------------------------------------------------
Core
------------------
* res_pjsip_pubsub is now required so call transfer progress can be monitored
and reported in the channel variable TRANSFERSTATUS.
app_voicemail.c
------------------
* The "Voicemail Build Options" section of menuselect has been removed along with
the FILE_STORAGE, ODBC_STORAGE and IMAP_STORAGE menuselect options. All 3 variants
of the voicemail app can now be built at the same by enabling app_voicemail,
app_voicemail_imap, and app_voicemail_odbc under the "Applications" section.
By default, only app_voicemail is enabled. Also, the modules.conf sample has
been updated to "noload" app_voicemail_imap and app_voicemail_odbc should they
all be built. Packagers must update their build scripts appropriately.
chan_pjsip
------------------
* res_pjsip_pubsub is now required so call transfer progress can be monitored
and reported in the channel variable TRANSFERSTATUS.
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.
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.
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.
From 12 to 13:
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
received 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.
From 11 to 12:
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.