The previous behavior was to simply set the accountcode of an outgoing channel to the accountcode of the channel initiating the call. It was done this way a long time ago to allow the accountcode set on the SIP/100 channel to be propagated to a local channel so the dialplan execution on the Local;2 channel would have the SIP/100 accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200 Propagating the SIP/100 accountcode to the local channels is very useful. Without any dialplan manipulation, all channels in this call would have the same accountcode. Using dialplan, you can set a different accountcode on the SIP/200 channel either by setting the accountcode on the Local;2 channel or by the Dial application's b(pre-dial), M(macro) or U(gosub) options, or by the FollowMe application's b(pre-dial) option, or by the Queue application's macro or gosub options. Before Asterisk v12, the altered accountcode on SIP/200 will remain until the local channels optimize out and the accountcode would change to the SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount support but ultimately had to punt on the support. The peeraccount support was rendered useless because of how the CDR code needed to unconditionally force the caller's accountcode onto the peer channel's accountcode. The CEL events were thus intentionally made to always use the channel's accountcode as the peeraccount value. With the arrival of Asterisk v12, the situation has improved somewhat so peeraccount support can be made to work. Using the indicated example, the the accountcode values become as follows when the peeraccount is set on SIP/100 before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already has an accountcode it can only change by the following explicit user actions: 1) A channel originate method that can specify an accountcode to use. 2) The calling channel propagating its non-empty peeraccount or its non-empty accountcode if the peeraccount was empty to the outgoing channel's accountcode before initiating the dial. e.g., Dial and FollowMe. The exception to this propagation method is Queue. Queue will only propagate peeraccounts this way only if the outgoing channel does not have an accountcode. 3) Dialplan using CHANNEL(accountcode). 4) Dialplan using CHANNEL(peeraccount) on the other end of a local channel pair. If a channel does not have an accountcode it can get one from the following places: 1) The channel driver's configuration at channel creation. 2) Explicit user action as already indicated. 3) Entering a basic or stasis-mixing bridge from a peer channel's peeraccount value. You can specify the accountcode for an outgoing channel by setting the CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue applications. Queue adds the wrinkle that it will not overwrite an existing accountcode on the outgoing channel with the calling channels values. Accountcode and peeraccount values propagate to an outgoing channel before dialing. Accountcodes also propagate when channels enter or leave a basic or stasis-mixing bridge. The peeraccount value only makes sense for mixing bridges with two channels; it is meaningless otherwise. * Made peeraccount functional by changing accountcode propagation as described above. * Fixed CEL extracting the wrong ie value for the peeraccount. This was done intentionally in Asterisk v1.8 when that version had to punt on peeraccount. * Fixed a few places dealing with accountcodes that were reading from channels without the lock held. AFS-65 #close Review: https://reviewboard.asterisk.org/r/3601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
373 lines
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373 lines
18 KiB
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===========================================================
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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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accountcode:
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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 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
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parameter instead.
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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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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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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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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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Unistim:
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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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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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WebSockets:
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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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===========================================================
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===========================================================
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