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<td id="header-avatar-image-container" valign="top" style="padding: 0px; border-collapse: collapse; vertical-align: top; width: 32px; padding-right: 9px"><a href="https://wiki.asterisk.org/wiki/display/~mjordan?src=email" style="color: #3b73af; text-decoration: none"><img id="header-avatar-image" class="image_fix" src="cid:avatar_ce51dcf276530e4a4b00548e2a6d0905" height="32" width="32" border="0" style="border-radius: 3px; vertical-align: top" /></a></td>
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<td id="page-title-pattern-icon-image-container-cell" style="padding: 0px; border-collapse: collapse; width: 16px; padding: 9px 8px 0px 0px; mso-text-raise: 5px; mso-line-height-rule: exactly"><a href="https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13?src=email" title="page icon" style="vertical-align: top;; color: #3b73af; text-decoration: none"><img style="vertical-align: top; display: block;" src="cid:page-icon" alt="page icon" title="page icon" height="16" width="16" border="0" /></a></td>
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<td class="content-excerpt-pattern-container mobile-resize-text " style="padding: 0px; border-collapse: collapse; padding: 0 0 0 24px"> <p style="margin: 10px 0 0 0; margin-top: 0">From 12 to 13:</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - Sample config files have been moved from configs/ to a subfolder of that</p> <p style="margin: 10px 0 0 0"> directory, 'samples'.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The menuselect utility has been pulled into the Asterisk repository. As a</p> <p style="margin: 10px 0 0 0"> result, the libxml2 development library is now a required dependency for</p> <p style="margin: 10px 0 0 0"> Asterisk.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The asterisk command line -I option and the asterisk.conf internal_timing</p> <p style="margin: 10px 0 0 0"> option are removed and always enabled if any timing module is loaded.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The per console verbose level feature as previously implemented caused a</p> <p style="margin: 10px 0 0 0"> large performance penalty. The fix required some minor incompatibilities</p> <p style="margin: 10px 0 0 0"> if the new rasterisk is used to connect to an earlier version. If the new</p> <p style="margin: 10px 0 0 0"> rasterisk connects to an older Asterisk version then the root console verbose</p> <p style="margin: 10px 0 0 0"> level is always affected by the "core set verbose" command of the remote</p> <p style="margin: 10px 0 0 0"> console even though it may appear to only affect the current console. If</p> <p style="margin: 10px 0 0 0"> an older version of rasterisk connects to the new version then the</p> <p style="margin: 10px 0 0 0"> "core set verbose" command will have no effect.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted</p> <p style="margin: 10px 0 0 0"> objects will emit additional debug information to the refs log file located</p> <p style="margin: 10px 0 0 0"> in the standard Asterisk log file directory. This log file is useful in</p> <p style="margin: 10px 0 0 0"> tracking down object leaks and other reference counting issues. Prior to</p> <p style="margin: 10px 0 0 0"> this version, this option was only available by modifying the source code</p> <p style="margin: 10px 0 0 0"> directly. This change also includes a new script, refcounter.py, in the</p> <p style="margin: 10px 0 0 0"> contrib folder that will process the refs log file.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The asterisk compatibility options in asterisk.conf have been removed.</p> <p style="margin: 10px 0 0 0"> These options enabled certain backwards compatibility features for</p> <p style="margin: 10px 0 0 0"> pbx_realtime, res_agi, and app_set that made their behaviour similar to</p> <p style="margin: 10px 0 0 0"> Asterisk 1.4. Users who used these backwards compatibility settings should</p> <p style="margin: 10px 0 0 0"> update their dialplans to use ',' instead of '|' as a delimiter, and should</p> <p style="margin: 10px 0 0 0"> use the Set dialplan application instead of the MSet dialplan application.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">accountcode:</p> <p style="margin: 10px 0 0 0"> - Accountcode behavior changed somewhat to add functional peeraccount</p> <p style="margin: 10px 0 0 0"> support. The main change is that local channels now cross accountcode</p> <p style="margin: 10px 0 0 0"> and peeraccount across the special bridge between the ;1 and ;2 channels</p> <p style="margin: 10px 0 0 0"> just like channels between normal bridges. See the CHANGES file for</p> <p style="margin: 10px 0 0 0"> more information.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">ARI:</p> <p style="margin: 10px 0 0 0"> - The ARI version has been changed from 1.0.0 to 1.1.0. This is to reflect</p> <p style="margin: 10px 0 0 0"> the backwards compatible changes listed below.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - Added a new ARI resource 'mailboxes' which allows the creation and</p> <p style="margin: 10px 0 0 0"> modification of mailboxes managed by external MWI. Modules res_mwi_external</p> <p style="margin: 10px 0 0 0"> and res_stasis_mailbox must be enabled to use this resource.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - Added new events for externally initiated transfers. The event</p> <p style="margin: 10px 0 0 0"> BridgeBlindTransfer is now raised when a channel initiates a blind transfer</p> <p style="margin: 10px 0 0 0"> of a bridge in the ARI controlled application to the dialplan; the</p> <p style="margin: 10px 0 0 0"> BridgeAttendedTransfer event is raised when a channel initiates an</p> <p style="margin: 10px 0 0 0"> attended transfer of a bridge in the ARI controlled application to the</p> <p style="margin: 10px 0 0 0"> dialplan.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - Channel variables may now be specified as a body parameter to the</p> <p style="margin: 10px 0 0 0"> POST /channels operation. The 'variables' key in the JSON is interpreted</p> <p style="margin: 10px 0 0 0"> as a sequence of key/value pairs that will be added to the created channel</p> <p style="margin: 10px 0 0 0"> as channel variables. Other parameters in the JSON body are treated as</p> <p style="margin: 10px 0 0 0"> query parameters of the same name.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - A bug fix in bridge creation has caused a behavioural change in how</p> <p style="margin: 10px 0 0 0"> subscriptions are created for bridges. A bridge created through ARI, does</p> <p style="margin: 10px 0 0 0"> not, by itself, have a subscription created for any particular Stasis</p> <p style="margin: 10px 0 0 0"> application. When a channel in a Stasis application joins a bridge, an</p> <p style="margin: 10px 0 0 0"> implicit event subscription is created for that bridge as well. Previously,</p> <p style="margin: 10px 0 0 0"> when a channel left such a bridge, the subscription was leaked; this allowed</p> <p style="margin: 10px 0 0 0"> for later bridge events to continue to be pushed to the subscribed</p> <p style="margin: 10px 0 0 0"> applications. That leak has been fixed; as a result, bridge events that were</p> <p style="margin: 10px 0 0 0"> delivered after a channel left the bridge are no longer delivered. An</p> <p style="margin: 10px 0 0 0"> application must subscribe to a bridge through the applications resource if</p> <p style="margin: 10px 0 0 0"> it wishes to receive all events related to a bridge.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">AMI:</p> <p style="margin: 10px 0 0 0"> - The AMI version has been changed from 2.0.0 to 2.1.0. This is to reflect</p> <p style="margin: 10px 0 0 0"> the backwards compatible changes listed below.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The DialStatus field in the DialEnd event can now have additional values.</p> <p style="margin: 10px 0 0 0"> This includes ABORT, CONTINUE, and GOTO.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The res_mwi_external_ami module can, if loaded, provide additional AMI</p> <p style="margin: 10px 0 0 0"> actions and events that convey MWI state within Asterisk. This includes</p> <p style="margin: 10px 0 0 0"> the MWIGet, MWIUpdate, and MWIDelete actions, as well as the MWIGet and</p> <p style="margin: 10px 0 0 0"> MWIGetComplete events that occur in response to an MWIGet action.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - AMI now contains a new class authorization, 'security'. This is used with</p> <p style="margin: 10px 0 0 0"> the following new events: FailedACL, InvalidAccountID, SessionLimit,</p> <p style="margin: 10px 0 0 0"> MemoryLimit, LoadAverageLimit, RequestNotAllowed, AuthMethodNotAllowed,</p> <p style="margin: 10px 0 0 0"> RequestBadFormat, SuccessfulAuth, UnexpectedAddress, ChallengeResponseFailed,</p> <p style="margin: 10px 0 0 0"> InvalidPassword, ChallengeSent, and InvalidTransport.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - Bridge related events now have two additional fields: BridgeName and</p> <p style="margin: 10px 0 0 0"> BridgeCreator. BridgeName is a descriptive name for the bridge;</p> <p style="margin: 10px 0 0 0"> BridgeCreator is the name of the entity that created the bridge. This</p> <p style="margin: 10px 0 0 0"> affects the following events: ConfbridgeStart, ConfbridgeEnd,</p> <p style="margin: 10px 0 0 0"> ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,</p> <p style="margin: 10px 0 0 0"> ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,</p> <p style="margin: 10px 0 0 0"> AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - MixMonitor AMI actions now require users to have authorization classes.</p> <p style="margin: 10px 0 0 0"> * MixMonitor - system</p> <p style="margin: 10px 0 0 0"> * MixMonitorMute - call or system</p> <p style="margin: 10px 0 0 0"> * StopMixMonitor - call or system</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - Removed the undocumented manager.conf block-sockets option. It interferes with</p> <p style="margin: 10px 0 0 0"> TCP/TLS inactivity timeouts.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The response to the PresenceState AMI action has historically contained two</p> <p style="margin: 10px 0 0 0"> Message keys. The first of these is used as an informative message regarding</p> <p style="margin: 10px 0 0 0"> the success/failure of the action; the second contains a Presence state</p> <p style="margin: 10px 0 0 0"> specific message. Having two keys with the same unique name in an AMI</p> <p style="margin: 10px 0 0 0"> message is cumbersome for some client; hence, the Presence specific Message</p> <p style="margin: 10px 0 0 0"> has been deprecated. The message will now contain a PresenceMessage key</p> <p style="margin: 10px 0 0 0"> for the presence specific information; the Message key containing presence</p> <p style="margin: 10px 0 0 0"> information will be removed in the next major version of AMI.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">CDRs:</p> <p style="margin: 10px 0 0 0"> - The "endbeforehexten" setting now defaults to "yes", instead of "no".</p> <p style="margin: 10px 0 0 0"> When set to "no", yhis setting will cause a new CDR to be generated when a</p> <p style="margin: 10px 0 0 0"> channel enters into hangup logic (either the 'h' extension or a hangup</p> <p style="margin: 10px 0 0 0"> handler subroutine). In general, this is not the preferred default: this</p> <p style="margin: 10px 0 0 0"> causes extra CDRs to be generated for a channel in many common dialplans.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The cdr_sqlite module was deprecated and has been removed. Users of this</p> <p style="margin: 10px 0 0 0"> module should use the cdr_sqlite3_custom module instead.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">chan_dahdi:</p> <p style="margin: 10px 0 0 0"> - SS7 support now requires libss7 v2.0 or later.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - Added the inband_on_setup_ack compatibility option to chan_dahdi.conf to</p> <p style="margin: 10px 0 0 0"> deal with switches that don't send an inband progress indication in the</p> <p style="margin: 10px 0 0 0"> SETUP ACKNOWLEDGE message.</p> <p style="margin: 10px 0 0 0"> Default is now no.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">chan_gtalk</p> <p style="margin: 10px 0 0 0"> - This module was deprecated and has been removed. Users of chan_gtalk</p> <p style="margin: 10px 0 0 0"> should use chan_motif.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">chan_h323</p> <p style="margin: 10px 0 0 0"> - This module was deprecated and has been removed. Users of chan_h323</p> <p style="margin: 10px 0 0 0"> should use chan_ooh323.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">chan_jingle</p> <p style="margin: 10px 0 0 0"> - This module was deprecated and has been removed. Users of chan_jingle</p> <p style="margin: 10px 0 0 0"> should use chan_motif.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">chan_pjsip:</p> <p style="margin: 10px 0 0 0"> - Added a 'force_avp' option to chan_pjsip which will force the usage of</p> <p style="margin: 10px 0 0 0"> 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' as the media transport type</p> <p style="margin: 10px 0 0 0"> in SDP offers depending on settings, even when DTLS is used for media</p> <p style="margin: 10px 0 0 0"> encryption.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - Added a 'media_use_received_transport' option to chan_pjsip which will</p> <p style="margin: 10px 0 0 0"> cause the SDP answer to use the media transport as received in the SDP</p> <p style="margin: 10px 0 0 0"> offer.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">chan_sip:</p> <p style="margin: 10px 0 0 0"> - Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip</p> <p style="margin: 10px 0 0 0"> interoperability.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The SIPPEER dialplan function no longer supports using a colon as a</p> <p style="margin: 10px 0 0 0"> delimiter for parameters. The parameters for the function should be</p> <p style="margin: 10px 0 0 0"> delimited using a comma.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The SIPCHANINFO dialplan function was deprecated and has been removed. Users</p> <p style="margin: 10px 0 0 0"> of the function should use the CHANNEL function instead.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - Added a 'force_avp' option for chan_sip. When enabled this option will</p> <p style="margin: 10px 0 0 0"> cause the media transport in the offer or answer SDP to be 'RTP/AVP',</p> <p style="margin: 10px 0 0 0"> 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' even if a DTLS stream has been</p> <p style="margin: 10px 0 0 0"> configured. This option can be set to improve interoperability with WebRTC</p> <p style="margin: 10px 0 0 0"> clients that don't use the RFC defined transport for DTLS.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The 'dtlsverify' option in chan_sip now has additional values besides</p> <p style="margin: 10px 0 0 0"> 'yes' and 'no'. If 'yes' is specified both the certificate and fingerprint</p> <p style="margin: 10px 0 0 0"> will be verified. If 'no' is specified then neither the certificate or</p> <p style="margin: 10px 0 0 0"> fingerprint is verified. If 'certificate' is specified then only the</p> <p style="margin: 10px 0 0 0"> certificate is verified. If 'fingerprint' is specified then only the</p> <p style="margin: 10px 0 0 0"> fingerprint is verified.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - A 'dtlsfingerprint' option has been added to chan_sip which allows the</p> <p style="margin: 10px 0 0 0"> hash to be specified for the DTLS fingerprint placed in SDP. Supported</p> <p style="margin: 10px 0 0 0"> values are 'sha-1' and 'sha-256' with 'sha-256' being the default.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The 'progressinband=never' option is now more zealous in the persecution of</p> <p style="margin: 10px 0 0 0"> progress messages coming from Asterisk. Channels bridged with a SIP channel</p> <p style="margin: 10px 0 0 0"> that has 'progressinband=never' set will not be able to forward their</p> <p style="margin: 10px 0 0 0"> progress indications through to the SIP device. chan_sip will now turn such</p> <p style="margin: 10px 0 0 0"> progress indications into a 180 Ringing (if a 180 has not yet been</p> <p style="margin: 10px 0 0 0"> transmitted) if 'progressinband=never'.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The codec preference order in an SDP during an offer is slightly different</p> <p style="margin: 10px 0 0 0"> than previous releases. Prior to Asterisk 13, the preference order of</p> <p style="margin: 10px 0 0 0"> codecs used to be:</p> <p style="margin: 10px 0 0 0"> (1) Our preferred codec</p> <p style="margin: 10px 0 0 0"> (2) Our configured codecs</p> <p style="margin: 10px 0 0 0"> (3) Any non-audio joint codecs</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> One of the ways the new media format architecture in Asterisk 13 improves</p> <p style="margin: 10px 0 0 0"> performance is by reference counting formats, such that they can be reused</p> <p style="margin: 10px 0 0 0"> in many places without additional allocation. To not require a large</p> <p style="margin: 10px 0 0 0"> amount of locking, an instance of a format is immutable by convention.</p> <p style="margin: 10px 0 0 0"> This works well except for formats with attributes. Since a media format</p> <p style="margin: 10px 0 0 0"> with an attribute is a different object than the same format without an</p> <p style="margin: 10px 0 0 0"> attribute, we have to carry over the formats with attributes from an</p> <p style="margin: 10px 0 0 0"> inbound offer so that the correct attributes are offered in an outgoing</p> <p style="margin: 10px 0 0 0"> INVITE request. This requires some subtle tweaks to the preference order</p> <p style="margin: 10px 0 0 0"> to ensure that the media format with attributes is offered to a remote</p> <p style="margin: 10px 0 0 0"> peer, as opposed to the same media format (but without attributes) that</p> <p style="margin: 10px 0 0 0"> may be stored in the peer object.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> All of this means that our offer offer list will now be:</p> <p style="margin: 10px 0 0 0"> (1) Our preferred codec</p> <p style="margin: 10px 0 0 0"> (2) Any joint codecs offered by the inbound offer</p> <p style="margin: 10px 0 0 0"> (3) All other codecs that are not the preferred codec and not a joint</p> <p style="margin: 10px 0 0 0"> codec offered by the inbound offer</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">CLI commands:</p> <p style="margin: 10px 0 0 0"> - "core show settings" now lists the current console verbosity in addition</p> <p style="margin: 10px 0 0 0"> to the root console verbosity.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - "core set verbose" has not been able to support the by module verbose</p> <p style="margin: 10px 0 0 0"> logging levels since verbose logging levels were made per console. That</p> <p style="margin: 10px 0 0 0"> syntax is now removed and a silence option added in its place.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">ConfBridge:</p> <p style="margin: 10px 0 0 0">- The sound_place_into_conference sound used in Confbridge is now deprecated</p> <p style="margin: 10px 0 0 0"> and is no longer functional since it has been broken since its inception</p> <p style="margin: 10px 0 0 0"> and the fix involved using a different method to achieve the same goal. The</p> <p style="margin: 10px 0 0 0"> new method to achieve this functionality is by using sound_begin to play</p> <p style="margin: 10px 0 0 0"> a sound to the conference when waitmarked users are moved into the conference.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">Configuration Files:</p> <p style="margin: 10px 0 0 0"> - The 'verbose' setting in logger.conf still takes an optional argument,</p> <p style="margin: 10px 0 0 0"> specifying the verbosity level for each logging destination. However,</p> <p style="margin: 10px 0 0 0"> the default is now to once again follow the current root console level.</p> <p style="margin: 10px 0 0 0"> As a result, using the AMI Command action with "core set verbose" could</p> <p style="margin: 10px 0 0 0"> again set the root console verbose level and affect the verbose level</p> <p style="margin: 10px 0 0 0"> logged.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The manager.conf 'eventfilter' now takes an "extended" regular expression</p> <p style="margin: 10px 0 0 0"> instead of a "basic" one.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The unistim.conf 'dateformat' has changed meaning of options values to conform</p> <p style="margin: 10px 0 0 0"> values used inside Unistim protocol</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">HTTP:</p> <p style="margin: 10px 0 0 0"> - Added http.conf session_inactivity timer option to close HTTP connections</p> <p style="margin: 10px 0 0 0"> that aren't doing anything.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - Added support for persistent HTTP connections. To enable persistent</p> <p style="margin: 10px 0 0 0"> HTTP connections configure the keep alive time between HTTP requests. The</p> <p style="margin: 10px 0 0 0"> keep alive time between HTTP requests is configured in http.conf with the</p> <p style="margin: 10px 0 0 0"> session_keep_alive parameter.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">MusicOnHold</p> <p style="margin: 10px 0 0 0"> - The SetMusicOnHold dialplan application was deprecated and has been removed.</p> <p style="margin: 10px 0 0 0"> Users of the application should use the CHANNEL function's musicclass</p> <p style="margin: 10px 0 0 0"> setting instead.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The WaitMusicOnHold dialplan application was deprecated and has been</p> <p style="margin: 10px 0 0 0"> removed. Users of the application should use MusicOnHold with a duration</p> <p style="margin: 10px 0 0 0"> parameter instead.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">ODBC:</p> <p style="margin: 10px 0 0 0">- The compatibility setting, allow_empty_string_in_nontext, has been removed.</p> <p style="margin: 10px 0 0 0"> Empty column values will be stored as empty strings during realtime updates.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">Realtime Configuration:</p> <p style="margin: 10px 0 0 0"> - WARNING: The database migration script that adds the 'extensions' table for</p> <p style="margin: 10px 0 0 0"> realtime had to be modified due to an error when installing for MySQL. The</p> <p style="margin: 10px 0 0 0"> 'extensions' table's 'id' column was changed to be a primary key. This could</p> <p style="margin: 10px 0 0 0"> potentially cause a migration problem. If so, it may be necessary to</p> <p style="margin: 10px 0 0 0"> manually alter the affected table/column to bring it back in line with the</p> <p style="margin: 10px 0 0 0"> migration scripts.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - New columns have been added to realtime tables for 'support_path' on</p> <p style="margin: 10px 0 0 0"> ps_registrations and ps_aors and for 'path' on ps_contacts for the new</p> <p style="margin: 10px 0 0 0"> SIP Path support in chan_pjsip.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The following new tables have been added for pjsip realtime: 'ps_systems',</p> <p style="margin: 10px 0 0 0"> 'ps_globals', 'ps_tranports', 'ps_registrations'.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The following columns were added to the 'ps_aors' realtime table:</p> <p style="margin: 10px 0 0 0"> 'maximum_expiration', 'outbound_proxy', and 'support_path'.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The following columns were added to the 'ps_contacts' realtime table:</p> <p style="margin: 10px 0 0 0"> 'outbound_proxy', 'user_agent', and 'path'.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - New columns have been added to the ps_endpoints realtime table for the</p> <p style="margin: 10px 0 0 0"> 'media_address', 'redirect_method' and 'set_var' options. Also the</p> <p style="margin: 10px 0 0 0"> 'mwi_fromuser' column was renamed to 'mwi_from_user'. A new column</p> <p style="margin: 10px 0 0 0"> 'message_context' was added to let users configure how MESSAGE requests are</p> <p style="margin: 10px 0 0 0"> routed to the dialplan.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - A new column was added to the 'ps_globals' realtime table for the 'debug'</p> <p style="margin: 10px 0 0 0"> option.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from</p> <p style="margin: 10px 0 0 0"> yes/no enumerators to string values. 'cos_audio' and 'cos_video' have been</p> <p style="margin: 10px 0 0 0"> changed from yes/no enumerators to integer values. PJSIP transport column</p> <p style="margin: 10px 0 0 0"> 'tos' has been changed from a yes/no enumerator to a string value. 'cos' has</p> <p style="margin: 10px 0 0 0"> been changed from a yes/no enumerator to an integer value.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - The 'queues' and 'queue_members' realtime tables have been added to the</p> <p style="margin: 10px 0 0 0"> config Alembic scripts.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - A new set of Alembic scripts has been added for CDR tables. This will create</p> <p style="margin: 10px 0 0 0"> a 'cdr' table with the default schema that Asterisk expects.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">res_jabber:</p> <p style="margin: 10px 0 0 0"> - This module was deprecated and has been removed. Users of this module should</p> <p style="margin: 10px 0 0 0"> use res_xmpp instead.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">safe_asterisk:</p> <p style="margin: 10px 0 0 0"> - The safe_asterisk script was previously not installed on top of an existing</p> <p style="margin: 10px 0 0 0"> version. This caused bug-fixes in that script not to be deployed. If your</p> <p style="margin: 10px 0 0 0"> safe_asterisk script is customized, be sure to keep your changes. Custom</p> <p style="margin: 10px 0 0 0"> values for variables should be created in *.sh file(s) inside</p> <p style="margin: 10px 0 0 0"> ASTETCDIR/startup.d/. See ASTERISK-21965.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> - Changed a log message in safe_asterisk and the $NOTIFY mail subject. If</p> <p style="margin: 10px 0 0 0"> you use tools to parse either of them, update your parse functions</p> <p style="margin: 10px 0 0 0"> accordingly. The changed strings are:</p> <p style="margin: 10px 0 0 0"> - "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."</p> <p style="margin: 10px 0 0 0"> - "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">Unistim:</p> <p style="margin: 10px 0 0 0"> - Added 'dtmf_duration' option with changing default operation to disable</p> <p style="margin: 10px 0 0 0"> receivied dtmf playback on unistim phone</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">Utilities:</p> <p style="margin: 10px 0 0 0"> - The refcounter program has been removed in favor of the refcounter.py script</p> <p style="margin: 10px 0 0 0"> in contrib/scripts.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0">WebSockets:</p> <p style="margin: 10px 0 0 0"> - Added a compatibility option to ari.conf, sip.conf, and pjsip.conf</p> <p style="margin: 10px 0 0 0"> 'websocket_write_timeout'. When a websocket connection exists where Asterisk</p> <p style="margin: 10px 0 0 0"> writes a substantial amount of data to the connected client, and the connected</p> <p style="margin: 10px 0 0 0"> client is slow to process the received data, the socket may be disconnected.</p> <p style="margin: 10px 0 0 0"> In such cases, it may be necessary to adjust this value.</p> <p style="margin: 10px 0 0 0"> Default is 100 ms.</p> <p style="margin: 10px 0 0 0"> </p> <p style="margin: 10px 0 0 0"> </p> </td>
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