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 ===========================================================
24 General Asterisk Changes:
25 - The asterisk command line -I option and the asterisk.conf internal_timing
26 option are removed and always enabled if any timing module is loaded.
28 - The per console verbose level feature as previously implemented caused a
29 large performance penalty. The fix required some minor incompatibilities
30 if the new rasterisk is used to connect to an earlier version. If the new
31 rasterisk connects to an older Asterisk version then the root console verbose
32 level is always affected by the "core set verbose" command of the remote
33 console even though it may appear to only affect the current console. If
34 an older version of rasterisk connects to the new version then the
35 "core set verbose" command will have no effect.
37 - The asterisk compatibility options in asterisk.conf have been removed.
38 These options enabled certain backwards compatibility features for
39 pbx_realtime, res_agi, and app_set that made their behaviour similar to
40 Asterisk 1.4. Users who used these backwards compatibility settings should
41 update their dialplans to use ',' instead of '|' as a delimiter, and should
42 use the Set dialplan application instead of the MSet dialplan application.
45 - Sample config files have been moved from configs/ to a subfolder of that
48 - The menuselect utility has been pulled into the Asterisk repository. As a
49 result, the libxml2 development library is now a required dependency for
52 - Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
53 objects will emit additional debug information to the refs log file located
54 in the standard Asterisk log file directory. This log file is useful in
55 tracking down object leaks and other reference counting issues. Prior to
56 this version, this option was only available by modifying the source code
57 directly. This change also includes a new script, refcounter.py, in the
58 contrib folder that will process the refs log file.
63 - The sound_place_into_conference sound used in Confbridge is now deprecated
64 and is no longer functional since it has been broken since its inception
65 and the fix involved using a different method to achieve the same goal. The
66 new method to achieve this functionality is by using sound_begin to play
67 a sound to the conference when waitmarked users are moved into the conference.
70 - Queue rules provided in queuerules.conf can no longer be named "general".
73 - The SetMusicOnHold dialplan application was deprecated and has been removed.
74 Users of the application should use the CHANNEL function's musicclass
78 - The WaitMusicOnHold dialplan application was deprecated and has been
79 removed. Users of the application should use MusicOnHold with a duration
83 - The cdr_sqlite module was deprecated and has been removed. Users of this
84 module should use the cdr_sqlite3_custom module instead.
89 - SS7 support now requires libss7 v2.0 or later.
91 - Added the inband_on_setup_ack compatibility option to chan_dahdi.conf to
92 deal with switches that don't send an inband progress indication in the
93 SETUP ACKNOWLEDGE message.
97 - This module was deprecated and has been removed. Users of chan_gtalk
98 should use chan_motif.
101 - This module was deprecated and has been removed. Users of chan_h323
102 should use chan_ooh323.
105 - This module was deprecated and has been removed. Users of chan_jingle
106 should use chan_motif.
109 - Added a 'force_avp' option to chan_pjsip which will force the usage of
110 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' as the media transport type
111 in SDP offers depending on settings, even when DTLS is used for media
114 - Added a 'media_use_received_transport' option to chan_pjsip which will
115 cause the SDP answer to use the media transport as received in the SDP
119 - Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip
122 - The SIPPEER dialplan function no longer supports using a colon as a
123 delimiter for parameters. The parameters for the function should be
124 delimited using a comma.
126 - The SIPCHANINFO dialplan function was deprecated and has been removed. Users
127 of the function should use the CHANNEL function instead.
129 - Added a 'force_avp' option for chan_sip. When enabled this option will
130 cause the media transport in the offer or answer SDP to be 'RTP/AVP',
131 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' even if a DTLS stream has been
132 configured. This option can be set to improve interoperability with WebRTC
133 clients that don't use the RFC defined transport for DTLS.
135 - The 'dtlsverify' option in chan_sip now has additional values besides
136 'yes' and 'no'. If 'yes' is specified both the certificate and fingerprint
137 will be verified. If 'no' is specified then neither the certificate or
138 fingerprint is verified. If 'certificate' is specified then only the
139 certificate is verified. If 'fingerprint' is specified then only the
140 fingerprint is verified.
142 - A 'dtlsfingerprint' option has been added to chan_sip which allows the
143 hash to be specified for the DTLS fingerprint placed in SDP. Supported
144 values are 'sha-1' and 'sha-256' with 'sha-256' being the default.
146 - The 'progressinband=never' option is now more zealous in the persecution of
147 progress messages coming from Asterisk. Channels bridged with a SIP channel
148 that has 'progressinband=never' set will not be able to forward their
149 progress indications through to the SIP device. chan_sip will now turn such
150 progress indications into a 180 Ringing (if a 180 has not yet been
151 transmitted) if 'progressinband=never'.
153 - The codec preference order in an SDP during an offer is slightly different
154 than previous releases. Prior to Asterisk 13, the preference order of
156 (1) Our preferred codec
157 (2) Our configured codecs
158 (3) Any non-audio joint codecs
160 One of the ways the new media format architecture in Asterisk 13 improves
161 performance is by reference counting formats, such that they can be reused
162 in many places without additional allocation. To not require a large
163 amount of locking, an instance of a format is immutable by convention.
164 This works well except for formats with attributes. Since a media format
165 with an attribute is a different object than the same format without an
166 attribute, we have to carry over the formats with attributes from an
167 inbound offer so that the correct attributes are offered in an outgoing
168 INVITE request. This requires some subtle tweaks to the preference order
169 to ensure that the media format with attributes is offered to a remote
170 peer, as opposed to the same media format (but without attributes) that
171 may be stored in the peer object.
173 All of this means that our offer offer list will now be:
174 (1) Our preferred codec
175 (2) Any joint codecs offered by the inbound offer
176 (3) All other codecs that are not the preferred codec and not a joint
177 codec offered by the inbound offer
180 - The unistim.conf 'dateformat' has changed meaning of options values to conform
181 values used inside Unistim protocol
183 - Added 'dtmf_duration' option with changing default operation to disable
184 receivied dtmf playback on unistim phone
189 - accountcode behavior changed somewhat to add functional peeraccount
190 support. The main change is that local channels now cross accountcode
191 and peeraccount across the special bridge between the ;1 and ;2 channels
192 just like channels between normal bridges. See the CHANGES file for
196 - The ARI version has been changed to 1.5.0. This is to reflect backwards
197 compatible changes made since 12.0.0 was released.
199 - Added a new ARI resource 'mailboxes' which allows the creation and
200 modification of mailboxes managed by external MWI. Modules res_mwi_external
201 and res_stasis_mailbox must be enabled to use this resource.
203 - Added new events for externally initiated transfers. The event
204 BridgeBlindTransfer is now raised when a channel initiates a blind transfer
205 of a bridge in the ARI controlled application to the dialplan; the
206 BridgeAttendedTransfer event is raised when a channel initiates an
207 attended transfer of a bridge in the ARI controlled application to the
210 - Channel variables may now be specified as a body parameter to the
211 POST /channels operation. The 'variables' key in the JSON is interpreted
212 as a sequence of key/value pairs that will be added to the created channel
213 as channel variables. Other parameters in the JSON body are treated as
214 query parameters of the same name.
216 - A bug fix in bridge creation has caused a behavioural change in how
217 subscriptions are created for bridges. A bridge created through ARI, does
218 not, by itself, have a subscription created for any particular Stasis
219 application. When a channel in a Stasis application joins a bridge, an
220 implicit event subscription is created for that bridge as well. Previously,
221 when a channel left such a bridge, the subscription was leaked; this allowed
222 for later bridge events to continue to be pushed to the subscribed
223 applications. That leak has been fixed; as a result, bridge events that were
224 delivered after a channel left the bridge are no longer delivered. An
225 application must subscribe to a bridge through the applications resource if
226 it wishes to receive all events related to a bridge.
229 - The AMI version has been changed to 2.5.0. This is to reflect backwards
230 compatible changes made since 12.0.0 was released.
232 - The DialStatus field in the DialEnd event can now have additional values.
233 This includes ABORT, CONTINUE, and GOTO.
235 - The res_mwi_external_ami module can, if loaded, provide additional AMI
236 actions and events that convey MWI state within Asterisk. This includes
237 the MWIGet, MWIUpdate, and MWIDelete actions, as well as the MWIGet and
238 MWIGetComplete events that occur in response to an MWIGet action.
240 - AMI now contains a new class authorization, 'security'. This is used with
241 the following new events: FailedACL, InvalidAccountID, SessionLimit,
242 MemoryLimit, LoadAverageLimit, RequestNotAllowed, AuthMethodNotAllowed,
243 RequestBadFormat, SuccessfulAuth, UnexpectedAddress, ChallengeResponseFailed,
244 InvalidPassword, ChallengeSent, and InvalidTransport.
246 - Bridge related events now have two additional fields: BridgeName and
247 BridgeCreator. BridgeName is a descriptive name for the bridge;
248 BridgeCreator is the name of the entity that created the bridge. This
249 affects the following events: ConfbridgeStart, ConfbridgeEnd,
250 ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
251 ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
252 AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
254 - MixMonitor AMI actions now require users to have authorization classes.
255 * MixMonitor - system
256 * MixMonitorMute - call or system
257 * StopMixMonitor - call or system
259 - Removed the undocumented manager.conf block-sockets option. It interferes with
260 TCP/TLS inactivity timeouts.
262 - The response to the PresenceState AMI action has historically contained two
263 Message keys. The first of these is used as an informative message regarding
264 the success/failure of the action; the second contains a Presence state
265 specific message. Having two keys with the same unique name in an AMI
266 message is cumbersome for some client; hence, the Presence specific Message
267 has been deprecated. The message will now contain a PresenceMessage key
268 for the presence specific information; the Message key containing presence
269 information will be removed in the next major version of AMI.
271 - The manager.conf 'eventfilter' now takes an "extended" regular expression
272 instead of a "basic" one.
275 - The "endbeforehexten" setting now defaults to "yes", instead of "no".
276 When set to "no", yhis setting will cause a new CDR to be generated when a
277 channel enters into hangup logic (either the 'h' extension or a hangup
278 handler subroutine). In general, this is not the preferred default: this
279 causes extra CDRs to be generated for a channel in many common dialplans.
282 - "core show settings" now lists the current console verbosity in addition
283 to the root console verbosity.
285 - "core set verbose" has not been able to support the by module verbose
286 logging levels since verbose logging levels were made per console. That
287 syntax is now removed and a silence option added in its place.
290 - The 'verbose' setting in logger.conf still takes an optional argument,
291 specifying the verbosity level for each logging destination. However,
292 the default is now to once again follow the current root console level.
293 As a result, using the AMI Command action with "core set verbose" could
294 again set the root console verbose level and affect the verbose level
298 - Added http.conf session_inactivity timer option to close HTTP connections
299 that aren't doing anything.
301 - Added support for persistent HTTP connections. To enable persistent
302 HTTP connections configure the keep alive time between HTTP requests. The
303 keep alive time between HTTP requests is configured in http.conf with the
304 session_keep_alive parameter.
306 Realtime Configuration:
307 - WARNING: The database migration script that adds the 'extensions' table for
308 realtime had to be modified due to an error when installing for MySQL. The
309 'extensions' table's 'id' column was changed to be a primary key. This could
310 potentially cause a migration problem. If so, it may be necessary to
311 manually alter the affected table/column to bring it back in line with the
314 - New columns have been added to realtime tables for 'support_path' on
315 ps_registrations and ps_aors and for 'path' on ps_contacts for the new
316 SIP Path support in chan_pjsip.
318 - The following new tables have been added for pjsip realtime: 'ps_systems',
319 'ps_globals', 'ps_tranports', 'ps_registrations'.
321 - The following columns were added to the 'ps_aors' realtime table:
322 'maximum_expiration', 'outbound_proxy', and 'support_path'.
324 - The following columns were added to the 'ps_contacts' realtime table:
325 'outbound_proxy', 'user_agent', and 'path'.
327 - New columns have been added to the ps_endpoints realtime table for the
328 'media_address', 'redirect_method' and 'set_var' options. Also the
329 'mwi_fromuser' column was renamed to 'mwi_from_user'. A new column
330 'message_context' was added to let users configure how MESSAGE requests are
331 routed to the dialplan.
333 - A new column was added to the 'ps_globals' realtime table for the 'debug'
336 - PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from
337 yes/no enumerators to string values. 'cos_audio' and 'cos_video' have been
338 changed from yes/no enumerators to integer values. PJSIP transport column
339 'tos' has been changed from a yes/no enumerator to a string value. 'cos' has
340 been changed from a yes/no enumerator to an integer value.
342 - The 'queues' and 'queue_members' realtime tables have been added to the
343 config Alembic scripts.
345 - A new set of Alembic scripts has been added for CDR tables. This will create
346 a 'cdr' table with the default schema that Asterisk expects.
348 - A new upgrade script has been added that adds a 'queue_rules' table for
349 app_queue. Users of app_queue can store queue rules in a database. It is
350 important to note that app_queue only looks for this table on module load or
351 module reload; for more information, see the CHANGES file.
356 - The compatibility setting, allow_empty_string_in_nontext, has been removed.
357 Empty column values will be stored as empty strings during realtime updates.
360 - This module was deprecated and has been removed. Users of this module should
361 use res_xmpp instead.
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.
373 - The safe_asterisk script was previously not installed on top of an existing
374 version. This caused bug-fixes in that script not to be deployed. If your
375 safe_asterisk script is customized, be sure to keep your changes. Custom
376 values for variables should be created in *.sh file(s) inside
377 ASTETCDIR/startup.d/. See ASTERISK-21965.
379 - Changed a log message in safe_asterisk and the $NOTIFY mail subject. If
380 you use tools to parse either of them, update your parse functions
381 accordingly. The changed strings are:
382 - "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
383 - "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"
386 - The refcounter program has been removed in favor of the refcounter.py script
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