400 lines
19 KiB
Plaintext
400 lines
19 KiB
Plaintext
===========================================================
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===
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=== Information for upgrading between Asterisk versions
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===
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=== These files document all the changes that MUST be taken
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=== into account when upgrading between the Asterisk
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=== versions listed below. These changes may require that
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=== you modify your configuration files, dialplan or (in
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=== some cases) source code if you have your own Asterisk
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=== modules or patches. These files also include advance
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=== notice of any functionality that has been marked as
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=== 'deprecated' and may be removed in a future release,
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=== along with the suggested replacement functionality.
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===
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=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
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=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
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=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
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=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
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=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
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=== UPGRADE-11.txt -- Upgrade info for 10 to 11
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=== UPGRADE-12.txt -- Upgrade info for 11 to 12
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===========================================================
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General Asterisk Changes:
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- The asterisk command line -I option and the asterisk.conf internal_timing
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option are removed and always enabled if any timing module is loaded.
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- The per console verbose level feature as previously implemented caused a
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large performance penalty. The fix required some minor incompatibilities
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if the new rasterisk is used to connect to an earlier version. If the new
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rasterisk connects to an older Asterisk version then the root console verbose
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level is always affected by the "core set verbose" command of the remote
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console even though it may appear to only affect the current console. If
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an older version of rasterisk connects to the new version then the
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"core set verbose" command will have no effect.
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- The asterisk compatibility options in asterisk.conf have been removed.
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These options enabled certain backwards compatibility features for
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pbx_realtime, res_agi, and app_set that made their behaviour similar to
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Asterisk 1.4. Users who used these backwards compatibility settings should
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update their dialplans to use ',' instead of '|' as a delimiter, and should
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use the Set dialplan application instead of the MSet dialplan application.
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Build System:
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- Sample config files have been moved from configs/ to a subfolder of that
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directory, 'samples'.
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- The menuselect utility has been pulled into the Asterisk repository. As a
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result, the libxml2 development library is now a required dependency for
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Asterisk.
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- Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
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objects will emit additional debug information to the refs log file located
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in the standard Asterisk log file directory. This log file is useful in
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tracking down object leaks and other reference counting issues. Prior to
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this version, this option was only available by modifying the source code
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directly. This change also includes a new script, refcounter.py, in the
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contrib folder that will process the refs log file.
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Applications:
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ConfBridge:
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- The sound_place_into_conference sound used in Confbridge is now deprecated
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and is no longer functional since it has been broken since its inception
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and the fix involved using a different method to achieve the same goal. The
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new method to achieve this functionality is by using sound_begin to play
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a sound to the conference when waitmarked users are moved into the conference.
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- Added 'Admin' header to ConfbridgeJoin, ConfbridgeLeave, ConfbridgeMute,
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ConfbridgeUnmute, and ConfbridgeTalking AMI events.
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ControlPlayback:
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- The ControlPlayback and 'control stream file' AGI command will no longer
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implicitly answer the channel. If you do not answer the channel prior to
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using either this application or AGI command, you must send Progress
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first.
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Queue:
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- Queue rules provided in queuerules.conf can no longer be named "general".
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SetMusicOnHold:
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- The SetMusicOnHold dialplan application was deprecated and has been removed.
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Users of the application should use the CHANNEL function's musicclass
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setting instead.
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WaitMusicOnHold:
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- The WaitMusicOnHold dialplan application was deprecated and has been
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removed. Users of the application should use MusicOnHold with a duration
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parameter instead.
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CDR Backends:
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- The cdr_sqlite module was deprecated and has been removed. Users of this
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module should use the cdr_sqlite3_custom module instead.
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Channel Drivers:
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chan_dahdi:
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- SS7 support now requires libss7 v2.0 or later.
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- Added the inband_on_setup_ack compatibility option to chan_dahdi.conf to
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deal with switches that don't send an inband progress indication in the
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SETUP ACKNOWLEDGE message.
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Default is now no.
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chan_gtalk
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- This module was deprecated and has been removed. Users of chan_gtalk
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should use chan_motif.
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chan_h323
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- This module was deprecated and has been removed. Users of chan_h323
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should use chan_ooh323.
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chan_jingle
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- This module was deprecated and has been removed. Users of chan_jingle
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should use chan_motif.
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chan_pjsip:
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- Added a 'force_avp' option to chan_pjsip which will force the usage of
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'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' as the media transport type
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in SDP offers depending on settings, even when DTLS is used for media
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encryption.
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- Added a 'media_use_received_transport' option to chan_pjsip which will
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cause the SDP answer to use the media transport as received in the SDP
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offer.
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chan_sip:
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- Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip
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interoperability.
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- The SIPPEER dialplan function no longer supports using a colon as a
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delimiter for parameters. The parameters for the function should be
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delimited using a comma.
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- The SIPCHANINFO dialplan function was deprecated and has been removed. Users
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of the function should use the CHANNEL function instead.
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- Added a 'force_avp' option for chan_sip. When enabled this option will
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cause the media transport in the offer or answer SDP to be 'RTP/AVP',
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'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' even if a DTLS stream has been
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configured. This option can be set to improve interoperability with WebRTC
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clients that don't use the RFC defined transport for DTLS.
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- The 'dtlsverify' option in chan_sip now has additional values besides
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'yes' and 'no'. If 'yes' is specified both the certificate and fingerprint
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will be verified. If 'no' is specified then neither the certificate or
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fingerprint is verified. If 'certificate' is specified then only the
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certificate is verified. If 'fingerprint' is specified then only the
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fingerprint is verified.
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- A 'dtlsfingerprint' option has been added to chan_sip which allows the
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hash to be specified for the DTLS fingerprint placed in SDP. Supported
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values are 'sha-1' and 'sha-256' with 'sha-256' being the default.
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- The 'progressinband=never' option is now more zealous in the persecution of
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progress messages coming from Asterisk. Channels bridged with a SIP channel
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that has 'progressinband=never' set will not be able to forward their
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progress indications through to the SIP device. chan_sip will now turn such
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progress indications into a 180 Ringing (if a 180 has not yet been
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transmitted) if 'progressinband=never'.
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- The codec preference order in an SDP during an offer is slightly different
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than previous releases. Prior to Asterisk 13, the preference order of
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codecs used to be:
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(1) Our preferred codec
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(2) Our configured codecs
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(3) Any non-audio joint codecs
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One of the ways the new media format architecture in Asterisk 13 improves
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performance is by reference counting formats, such that they can be reused
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in many places without additional allocation. To not require a large
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amount of locking, an instance of a format is immutable by convention.
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This works well except for formats with attributes. Since a media format
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with an attribute is a different object than the same format without an
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attribute, we have to carry over the formats with attributes from an
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inbound offer so that the correct attributes are offered in an outgoing
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INVITE request. This requires some subtle tweaks to the preference order
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to ensure that the media format with attributes is offered to a remote
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peer, as opposed to the same media format (but without attributes) that
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may be stored in the peer object.
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All of this means that our offer offer list will now be:
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(1) Our preferred codec
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(2) Any joint codecs offered by the inbound offer
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(3) All other codecs that are not the preferred codec and not a joint
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codec offered by the inbound offer
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chan_unistim:
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- The unistim.conf 'dateformat' has changed meaning of options values to conform
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values used inside Unistim protocol
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- Added 'dtmf_duration' option with changing default operation to disable
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receivied dtmf playback on unistim phone
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Core:
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Account Codes:
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- accountcode behavior changed somewhat to add functional peeraccount
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support. The main change is that local channels now cross accountcode
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and peeraccount across the special bridge between the ;1 and ;2 channels
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just like channels between normal bridges. See the CHANGES file for
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more information.
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ARI:
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- The ARI version has been changed to 1.5.0. This is to reflect backwards
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compatible changes made since 12.0.0 was released.
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- Added a new ARI resource 'mailboxes' which allows the creation and
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modification of mailboxes managed by external MWI. Modules res_mwi_external
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and res_stasis_mailbox must be enabled to use this resource.
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- Added new events for externally initiated transfers. The event
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BridgeBlindTransfer is now raised when a channel initiates a blind transfer
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of a bridge in the ARI controlled application to the dialplan; the
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BridgeAttendedTransfer event is raised when a channel initiates an
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attended transfer of a bridge in the ARI controlled application to the
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dialplan.
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- Channel variables may now be specified as a body parameter to the
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POST /channels operation. The 'variables' key in the JSON is interpreted
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as a sequence of key/value pairs that will be added to the created channel
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as channel variables. Other parameters in the JSON body are treated as
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query parameters of the same name.
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- A bug fix in bridge creation has caused a behavioural change in how
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subscriptions are created for bridges. A bridge created through ARI, does
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not, by itself, have a subscription created for any particular Stasis
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application. When a channel in a Stasis application joins a bridge, an
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implicit event subscription is created for that bridge as well. Previously,
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when a channel left such a bridge, the subscription was leaked; this allowed
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for later bridge events to continue to be pushed to the subscribed
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applications. That leak has been fixed; as a result, bridge events that were
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delivered after a channel left the bridge are no longer delivered. An
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application must subscribe to a bridge through the applications resource if
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it wishes to receive all events related to a bridge.
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AMI:
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- The AMI version has been changed to 2.5.0. This is to reflect backwards
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compatible changes made since 12.0.0 was released.
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- The DialStatus field in the DialEnd event can now have additional values.
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This includes ABORT, CONTINUE, and GOTO.
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- The res_mwi_external_ami module can, if loaded, provide additional AMI
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actions and events that convey MWI state within Asterisk. This includes
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the MWIGet, MWIUpdate, and MWIDelete actions, as well as the MWIGet and
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MWIGetComplete events that occur in response to an MWIGet action.
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- AMI now contains a new class authorization, 'security'. This is used with
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the following new events: FailedACL, InvalidAccountID, SessionLimit,
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MemoryLimit, LoadAverageLimit, RequestNotAllowed, AuthMethodNotAllowed,
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RequestBadFormat, SuccessfulAuth, UnexpectedAddress, ChallengeResponseFailed,
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InvalidPassword, ChallengeSent, and InvalidTransport.
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- Bridge related events now have two additional fields: BridgeName and
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BridgeCreator. BridgeName is a descriptive name for the bridge;
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BridgeCreator is the name of the entity that created the bridge. This
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affects the following events: ConfbridgeStart, ConfbridgeEnd,
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ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
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ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
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AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave
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- MixMonitor AMI actions now require users to have authorization classes.
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* MixMonitor - system
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* MixMonitorMute - call or system
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* StopMixMonitor - call or system
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- Removed the undocumented manager.conf block-sockets option. It interferes with
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TCP/TLS inactivity timeouts.
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- The response to the PresenceState AMI action has historically contained two
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Message keys. The first of these is used as an informative message regarding
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the success/failure of the action; the second contains a Presence state
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specific message. Having two keys with the same unique name in an AMI
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message is cumbersome for some client; hence, the Presence specific Message
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has been deprecated. The message will now contain a PresenceMessage key
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for the presence specific information; the Message key containing presence
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information will be removed in the next major version of AMI.
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- The manager.conf 'eventfilter' now takes an "extended" regular expression
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instead of a "basic" one.
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CDRs:
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- The "endbeforehexten" setting now defaults to "yes", instead of "no".
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When set to "no", yhis setting will cause a new CDR to be generated when a
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channel enters into hangup logic (either the 'h' extension or a hangup
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handler subroutine). In general, this is not the preferred default: this
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causes extra CDRs to be generated for a channel in many common dialplans.
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CLI commands:
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- "core show settings" now lists the current console verbosity in addition
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to the root console verbosity.
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- "core set verbose" has not been able to support the by module verbose
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logging levels since verbose logging levels were made per console. That
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syntax is now removed and a silence option added in its place.
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Logging:
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- The 'verbose' setting in logger.conf still takes an optional argument,
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specifying the verbosity level for each logging destination. However,
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the default is now to once again follow the current root console level.
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As a result, using the AMI Command action with "core set verbose" could
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again set the root console verbose level and affect the verbose level
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logged.
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HTTP:
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- Added http.conf session_inactivity timer option to close HTTP connections
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that aren't doing anything.
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- Added support for persistent HTTP connections. To enable persistent
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HTTP connections configure the keep alive time between HTTP requests. The
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keep alive time between HTTP requests is configured in http.conf with the
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session_keep_alive parameter.
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Realtime Configuration:
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- WARNING: The database migration script that adds the 'extensions' table for
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realtime had to be modified due to an error when installing for MySQL. The
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'extensions' table's 'id' column was changed to be a primary key. This could
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potentially cause a migration problem. If so, it may be necessary to
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manually alter the affected table/column to bring it back in line with the
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migration scripts.
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- New columns have been added to realtime tables for 'support_path' on
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ps_registrations and ps_aors and for 'path' on ps_contacts for the new
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SIP Path support in chan_pjsip.
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- The following new tables have been added for pjsip realtime: 'ps_systems',
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'ps_globals', 'ps_tranports', 'ps_registrations'.
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- The following columns were added to the 'ps_aors' realtime table:
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'maximum_expiration', 'outbound_proxy', and 'support_path'.
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- The following columns were added to the 'ps_contacts' realtime table:
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'outbound_proxy', 'user_agent', and 'path'.
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- New columns have been added to the ps_endpoints realtime table for the
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'media_address', 'redirect_method' and 'set_var' options. Also the
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'mwi_fromuser' column was renamed to 'mwi_from_user'. A new column
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'message_context' was added to let users configure how MESSAGE requests are
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routed to the dialplan.
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- A new column was added to the 'ps_globals' realtime table for the 'debug'
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option.
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- PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from
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yes/no enumerators to string values. 'cos_audio' and 'cos_video' have been
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changed from yes/no enumerators to integer values. PJSIP transport column
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'tos' has been changed from a yes/no enumerator to a string value. 'cos' has
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been changed from a yes/no enumerator to an integer value.
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- The 'queues' and 'queue_members' realtime tables have been added to the
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config Alembic scripts.
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- A new set of Alembic scripts has been added for CDR tables. This will create
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a 'cdr' table with the default schema that Asterisk expects.
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- A new upgrade script has been added that adds a 'queue_rules' table for
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app_queue. Users of app_queue can store queue rules in a database. It is
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important to note that app_queue only looks for this table on module load or
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module reload; for more information, see the CHANGES file.
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Resources:
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res_odbc:
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- The compatibility setting, allow_empty_string_in_nontext, has been removed.
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Empty column values will be stored as empty strings during realtime updates.
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res_jabber:
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- This module was deprecated and has been removed. Users of this module should
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use res_xmpp instead.
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res_http_websocket:
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- Added a compatibility option to ari.conf, sip.conf, and pjsip.conf
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'websocket_write_timeout'. When a websocket connection exists where Asterisk
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writes a substantial amount of data to the connected client, and the connected
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client is slow to process the received data, the socket may be disconnected.
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In such cases, it may be necessary to adjust this value.
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Default is 100 ms.
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Scripts:
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safe_asterisk:
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- The safe_asterisk script was previously not installed on top of an existing
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version. This caused bug-fixes in that script not to be deployed. If your
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safe_asterisk script is customized, be sure to keep your changes. Custom
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values for variables should be created in *.sh file(s) inside
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ASTETCDIR/startup.d/. See ASTERISK-21965.
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- Changed a log message in safe_asterisk and the $NOTIFY mail subject. If
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you use tools to parse either of them, update your parse functions
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accordingly. The changed strings are:
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- "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
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- "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"
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Utilities:
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- The refcounter program has been removed in favor of the refcounter.py script
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in contrib/scripts.
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===========================================================
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===========================================================
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