=== 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.
- Sample config files have been moved from configs/ to a subfolder of that
- The menuselect utility has been pulled into the Asterisk repository. As a
result, the libxml2 development library is now a required dependency for
- 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.
- 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.
- 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
- Queue rules provided in queuerules.conf can no longer be named "general".
- The SetMusicOnHold dialplan application was deprecated and has been removed.
Users of the application should use the CHANNEL function's musicclass
- The WaitMusicOnHold dialplan application was deprecated and has been
removed. Users of the application should use MusicOnHold with a duration
- The cdr_sqlite module was deprecated and has been removed. Users of this
module should use the cdr_sqlite3_custom module instead.
- 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.
- This module was deprecated and has been removed. Users of chan_gtalk
should use chan_motif.
- This module was deprecated and has been removed. Users of chan_h323
should use chan_ooh323.
- This module was deprecated and has been removed. Users of chan_jingle
should use chan_motif.
- 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
- 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
- Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip
- 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
- 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
- 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
- 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
- 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.
- 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.
- 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.
- "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.
- 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
- 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
- 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
- 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'
- 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.
- The compatibility setting, allow_empty_string_in_nontext, has been removed.
Empty column values will be stored as empty strings during realtime updates.
- This module was deprecated and has been removed. Users of this module should
use res_xmpp instead.
- 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.
- 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)"
- The refcounter program has been removed in favor of the refcounter.py script