1 ===========================================================
3 === Information for upgrading between Asterisk versions
5 === These files document all the changes that MUST be taken
6 === into account when upgrading between the Asterisk
7 === versions listed below. These changes may require that
8 === you modify your configuration files, dialplan or (in
9 === some cases) source code if you have your own Asterisk
10 === modules or patches. These files also include advance
11 === notice of any functionality that has been marked as
12 === 'deprecated' and may be removed in a future release,
13 === along with the suggested replacement functionality.
15 === UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
16 === UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
17 === UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
18 === UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
19 === UPGRADE-10.txt -- Upgrade info for 1.8 to 10
20 === UPGRADE-11.txt -- Upgrade info for 10 to 11
21 === UPGRADE-12.txt -- Upgrade info for 11 to 12
22 ===========================================================
26 - Sample config files have been moved from configs/ to a subfolder of that
29 - The menuselect utility has been pulled into the Asterisk repository. As a
30 result, the libxml2 development library is now a required dependency for
33 - The asterisk command line -I option and the asterisk.conf internal_timing
34 option are removed and always enabled if any timing module is loaded.
36 - The per console verbose level feature as previously implemented caused a
37 large performance penalty. The fix required some minor incompatibilities
38 if the new rasterisk is used to connect to an earlier version. If the new
39 rasterisk connects to an older Asterisk version then the root console verbose
40 level is always affected by the "core set verbose" command of the remote
41 console even though it may appear to only affect the current console. If
42 an older version of rasterisk connects to the new version then the
43 "core set verbose" command will have no effect.
45 - Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
46 objects will emit additional debug information to the refs log file located
47 in the standard Asterisk log file directory. This log file is useful in
48 tracking down object leaks and other reference counting issues. Prior to
49 this version, this option was only available by modifying the source code
50 directly. This change also includes a new script, refcounter.py, in the
51 contrib folder that will process the refs log file.
53 - The asterisk compatibility options in asterisk.conf have been removed.
54 These options enabled certain backwards compatibility features for
55 pbx_realtime, res_agi, and app_set that made their behaviour similar to
56 Asterisk 1.4. Users who used these backwards compatibility settings should
57 update their dialplans to use ',' instead of '|' as a delimiter, and should
58 use the Set dialplan application instead of the MSet dialplan application.
61 - Accountcode behavior changed somewhat to add functional peeraccount
62 support. The main change is that local channels now cross accountcode
63 and peeraccount across the special bridge between the ;1 and ;2 channels
64 just like channels between normal bridges. See the CHANGES file for
68 - The ARI version has been changed from 1.0.0 to 1.1.0. This is to reflect
69 the backwards compatible changes listed below.
71 - Added a new ARI resource 'mailboxes' which allows the creation and
72 modification of mailboxes managed by external MWI. Modules res_mwi_external
73 and res_stasis_mailbox must be enabled to use this resource.
75 - Added new events for externally initiated transfers. The event
76 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
77 of a bridge in the ARI controlled application to the dialplan; the
78 BridgeAttendedTransfer event is raised when a channel initiates an
79 attended transfer of a bridge in the ARI controlled application to the
82 - Channel variables may now be specified as a body parameter to the
83 POST /channels operation. The 'variables' key in the JSON is interpreted
84 as a sequence of key/value pairs that will be added to the created channel
85 as channel variables. Other parameters in the JSON body are treated as
86 query parameters of the same name.
88 - A bug fix in bridge creation has caused a behavioural change in how
89 subscriptions are created for bridges. A bridge created through ARI, does
90 not, by itself, have a subscription created for any particular Stasis
91 application. When a channel in a Stasis application joins a bridge, an
92 implicit event subscription is created for that bridge as well. Previously,
93 when a channel left such a bridge, the subscription was leaked; this allowed
94 for later bridge events to continue to be pushed to the subscribed
95 applications. That leak has been fixed; as a result, bridge events that were
96 delivered after a channel left the bridge are no longer delivered. An
97 application must subscribe to a bridge through the applications resource if
98 it wishes to receive all events related to a bridge.
101 - The AMI version has been changed from 2.0.0 to 2.1.0. This is to reflect
102 the backwards compatible changes listed below.
104 - The DialStatus field in the DialEnd event can now have additional values.
105 This includes ABORT, CONTINUE, and GOTO.
107 - The res_mwi_external_ami module can, if loaded, provide additional AMI
108 actions and events that convey MWI state within Asterisk. This includes
109 the MWIGet, MWIUpdate, and MWIDelete actions, as well as the MWIGet and
110 MWIGetComplete events that occur in response to an MWIGet action.
112 - AMI now contains a new class authorization, 'security'. This is used with
113 the following new events: FailedACL, InvalidAccountID, SessionLimit,
114 MemoryLimit, LoadAverageLimit, RequestNotAllowed, AuthMethodNotAllowed,
115 RequestBadFormat, SuccessfulAuth, UnexpectedAddress, ChallengeResponseFailed,
116 InvalidPassword, ChallengeSent, and InvalidTransport.
118 - Bridge related events now have two additional fields: BridgeName and
119 BridgeCreator. BridgeName is a descriptive name for the bridge;
120 BridgeCreator is the name of the entity that created the bridge. This
121 affects the following events: ConfbridgeStart, ConfbridgeEnd,
122 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
123 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
124 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
126 - MixMonitor AMI actions now require users to have authorization classes.
127 * MixMonitor - system
128 * MixMonitorMute - call or system
129 * StopMixMonitor - call or system
131 - Removed the undocumented manager.conf block-sockets option. It interferes with
132 TCP/TLS inactivity timeouts.
134 - The response to the PresenceState AMI action has historically contained two
135 Message keys. The first of these is used as an informative message regarding
136 the success/failure of the action; the second contains a Presence state
137 specific message. Having two keys with the same unique name in an AMI
138 message is cumbersome for some client; hence, the Presence specific Message
139 has been deprecated. The message will now contain a PresenceMessage key
140 for the presence specific information; the Message key containing presence
141 information will be removed in the next major version of AMI.
144 - The "endbeforehexten" setting now defaults to "yes", instead of "no".
145 When set to "no", yhis setting will cause a new CDR to be generated when a
146 channel enters into hangup logic (either the 'h' extension or a hangup
147 handler subroutine). In general, this is not the preferred default: this
148 causes extra CDRs to be generated for a channel in many common dialplans.
150 - The cdr_sqlite module was deprecated and has been removed. Users of this
151 module should use the cdr_sqlite3_custom module instead.
154 - SS7 support now requires libss7 v2.0 or later.
156 - Added the inband_on_setup_ack compatibility option to chan_dahdi.conf to
157 deal with switches that don't send an inband progress indication in the
158 SETUP ACKNOWLEDGE message.
162 - This module was deprecated and has been removed. Users of chan_gtalk
163 should use chan_motif.
166 - This module was deprecated and has been removed. Users of chan_h323
167 should use chan_ooh323.
170 - This module was deprecated and has been removed. Users of chan_jingle
171 should use chan_motif.
174 - Added a 'force_avp' option to chan_pjsip which will force the usage of
175 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' as the media transport type
176 in SDP offers depending on settings, even when DTLS is used for media
179 - Added a 'media_use_received_transport' option to chan_pjsip which will
180 cause the SDP answer to use the media transport as received in the SDP
184 - Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip
187 - The SIPPEER dialplan function no longer supports using a colon as a
188 delimiter for parameters. The parameters for the function should be
189 delimited using a comma.
191 - The SIPCHANINFO dialplan function was deprecated and has been removed. Users
192 of the function should use the CHANNEL function instead.
194 - Added a 'force_avp' option for chan_sip. When enabled this option will
195 cause the media transport in the offer or answer SDP to be 'RTP/AVP',
196 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' even if a DTLS stream has been
197 configured. This option can be set to improve interoperability with WebRTC
198 clients that don't use the RFC defined transport for DTLS.
200 - The 'dtlsverify' option in chan_sip now has additional values besides
201 'yes' and 'no'. If 'yes' is specified both the certificate and fingerprint
202 will be verified. If 'no' is specified then neither the certificate or
203 fingerprint is verified. If 'certificate' is specified then only the
204 certificate is verified. If 'fingerprint' is specified then only the
205 fingerprint is verified.
207 - A 'dtlsfingerprint' option has been added to chan_sip which allows the
208 hash to be specified for the DTLS fingerprint placed in SDP. Supported
209 values are 'sha-1' and 'sha-256' with 'sha-256' being the default.
211 - The 'progressinband=never' option is now more zealous in the persecution of
212 progress messages coming from Asterisk. Channels bridged with a SIP channel
213 that has 'progressinband=never' set will not be able to forward their
214 progress indications through to the SIP device. chan_sip will now turn such
215 progress indications into a 180 Ringing (if a 180 has not yet been
216 transmitted) if 'progressinband=never'.
218 - The codec preference order in an SDP during an offer is slightly different
219 than previous releases. Prior to Asterisk 13, the preference order of
221 (1) Our preferred codec
222 (2) Our configured codecs
223 (3) Any non-audio joint codecs
225 One of the ways the new media format architecture in Asterisk 13 improves
226 performance is by reference counting formats, such that they can be reused
227 in many places without additional allocation. To not require a large
228 amount of locking, an instance of a format is immutable by convention.
229 This works well except for formats with attributes. Since a media format
230 with an attribute is a different object than the same format without an
231 attribute, we have to carry over the formats with attributes from an
232 inbound offer so that the correct attributes are offered in an outgoing
233 INVITE request. This requires some subtle tweaks to the preference order
234 to ensure that the media format with attributes is offered to a remote
235 peer, as opposed to the same media format (but without attributes) that
236 may be stored in the peer object.
238 All of this means that our offer offer list will now be:
239 (1) Our preferred codec
240 (2) Any joint codecs offered by the inbound offer
241 (3) All other codecs that are not the preferred codec and not a joint
242 codec offered by the inbound offer
245 - "core show settings" now lists the current console verbosity in addition
246 to the root console verbosity.
248 - "core set verbose" has not been able to support the by module verbose
249 logging levels since verbose logging levels were made per console. That
250 syntax is now removed and a silence option added in its place.
253 - The sound_place_into_conference sound used in Confbridge is now deprecated
254 and is no longer functional since it has been broken since its inception
255 and the fix involved using a different method to achieve the same goal. The
256 new method to achieve this functionality is by using sound_begin to play
257 a sound to the conference when waitmarked users are moved into the conference.
261 - The 'verbose' setting in logger.conf still takes an optional argument,
262 specifying the verbosity level for each logging destination. However,
263 the default is now to once again follow the current root console level.
264 As a result, using the AMI Command action with "core set verbose" could
265 again set the root console verbose level and affect the verbose level
268 - The manager.conf 'eventfilter' now takes an "extended" regular expression
269 instead of a "basic" one.
271 - The unistim.conf 'dateformat' has changed meaning of options values to conform
272 values used inside Unistim protocol
275 - Added http.conf session_inactivity timer option to close HTTP connections
276 that aren't doing anything.
278 - Added support for persistent HTTP connections. To enable persistent
279 HTTP connections configure the keep alive time between HTTP requests. The
280 keep alive time between HTTP requests is configured in http.conf with the
281 session_keep_alive parameter.
284 - The SetMusicOnHold dialplan application was deprecated and has been removed.
285 Users of the application should use the CHANNEL function's musicclass
288 - The WaitMusicOnHold dialplan application was deprecated and has been
289 removed. Users of the application should use MusicOnHold with a duration
293 - The compatibility setting, allow_empty_string_in_nontext, has been removed.
294 Empty column values will be stored as empty strings during realtime updates.
296 Realtime Configuration:
297 - WARNING: The database migration script that adds the 'extensions' table for
298 realtime had to be modified due to an error when installing for MySQL. The
299 'extensions' table's 'id' column was changed to be a primary key. This could
300 potentially cause a migration problem. If so, it may be necessary to
301 manually alter the affected table/column to bring it back in line with the
304 - New columns have been added to realtime tables for 'support_path' on
305 ps_registrations and ps_aors and for 'path' on ps_contacts for the new
306 SIP Path support in chan_pjsip.
308 - The following new tables have been added for pjsip realtime: 'ps_systems',
309 'ps_globals', 'ps_tranports', 'ps_registrations'.
311 - The following columns were added to the 'ps_aors' realtime table:
312 'maximum_expiration', 'outbound_proxy', and 'support_path'.
314 - The following columns were added to the 'ps_contacts' realtime table:
315 'outbound_proxy', 'user_agent', and 'path'.
317 - New columns have been added to the ps_endpoints realtime table for the
318 'media_address', 'redirect_method' and 'set_var' options. Also the
319 'mwi_fromuser' column was renamed to 'mwi_from_user'. A new column
320 'message_context' was added to let users configure how MESSAGE requests are
321 routed to the dialplan.
323 - A new column was added to the 'ps_globals' realtime table for the 'debug'
326 - PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from
327 yes/no enumerators to string values. 'cos_audio' and 'cos_video' have been
328 changed from yes/no enumerators to integer values. PJSIP transport column
329 'tos' has been changed from a yes/no enumerator to a string value. 'cos' has
330 been changed from a yes/no enumerator to an integer value.
332 - The 'queues' and 'queue_members' realtime tables have been added to the
333 config Alembic scripts.
335 - A new set of Alembic scripts has been added for CDR tables. This will create
336 a 'cdr' table with the default schema that Asterisk expects.
339 - This module was deprecated and has been removed. Users of this module should
340 use res_xmpp instead.
343 - The safe_asterisk script was previously not installed on top of an existing
344 version. This caused bug-fixes in that script not to be deployed. If your
345 safe_asterisk script is customized, be sure to keep your changes. Custom
346 values for variables should be created in *.sh file(s) inside
347 ASTETCDIR/startup.d/. See ASTERISK-21965.
349 - Changed a log message in safe_asterisk and the $NOTIFY mail subject. If
350 you use tools to parse either of them, update your parse functions
351 accordingly. The changed strings are:
352 - "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
353 - "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"
356 - Added 'dtmf_duration' option with changing default operation to disable
357 receivied dtmf playback on unistim phone
360 - The refcounter program has been removed in favor of the refcounter.py script
364 - Added a compatibility option to ari.conf, sip.conf, and pjsip.conf
365 'websocket_write_timeout'. When a websocket connection exists where Asterisk
366 writes a substantial amount of data to the connected client, and the connected
367 client is slow to process the received data, the socket may be disconnected.
368 In such cases, it may be necessary to adjust this value.
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