a2c912e997
In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
366 lines
17 KiB
Text
366 lines
17 KiB
Text
===========================================================
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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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From 12 to 13:
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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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- 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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- 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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- 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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ARI:
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- The ARI version has been changed from 1.0.0 to 1.1.0. This is to reflect
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the backwards compatible changes listed below.
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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 from 2.0.0 to 2.1.0. This is to reflect
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the backwards compatible changes listed below.
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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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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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- 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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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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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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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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Configuration Files:
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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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- The manager.conf 'eventfilter' now takes an "extended" regular expression
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instead of a "basic" one.
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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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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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MusicOnHold
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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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- The WaitMusicOnHold dialplan application was deprecated and has been
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removed. Users of the application should use MusicOnHold with a duration
|
|
parameter instead.
|
|
|
|
ODBC:
|
|
- The compatibility setting, allow_empty_string_in_nontext, has been removed.
|
|
Empty column values will be stored as empty strings during realtime updates.
|
|
|
|
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.
|
|
|
|
res_jabber:
|
|
- This module was deprecated and has been removed. Users of this module should
|
|
use res_xmpp instead.
|
|
|
|
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)"
|
|
|
|
Unistim:
|
|
- Added 'dtmf_duration' option with changing default operation to disable
|
|
receivied dtmf playback on unistim phone
|
|
|
|
Utilities:
|
|
- The refcounter program has been removed in favor of the refcounter.py script
|
|
in contrib/scripts.
|
|
|
|
WebSockets:
|
|
- 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.
|
|
|
|
===========================================================
|
|
===========================================================
|