497 lines
23 KiB
Text
497 lines
23 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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===
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===========================================================
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from 11.17 to 11.18
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Source Control:
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- Asterisk has moved from Subversion to Git. As a result, several changes
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were required in functionality. These are listed individually in the
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notes below.
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Build System:
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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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AMI:
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- The 'ModuleCheck' Action's Version key will now always report the
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current version of Asterisk.
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CLI:
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- The 'core show file version' command has been altered. In the past,
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this command would show the SVN revision of the source files compiled
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in Asterisk. However, when Asterisk moved to Git, the source control
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version support was removed. As a result, the version information shown
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by the CLI command is always the Asterisk version. This CLI command
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will be removed in Asterisk 14.
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chan_dahdi:
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- Added the force_restart_unavailable_chans compatibility option. When
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enabled it causes Asterisk to restart the ISDN B channel if an outgoing
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call receives cause 44 (Requested channel not available). The new option
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is enabled by default in current release branches for backward
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compatibility.
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cdr_odbc:
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- Added support for post-1.8 CDR columns 'peeraccount', 'linkedid', and
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'sequence'. Support for the new columns can be enabled via the newcdrcolumns
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option in cdr_odbc.conf.
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cdr_csv:
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- Added a new configuration option, "newcdrcolumns", which enables use of the
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post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
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from 11.16 to 11.17
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chan_dahdi:
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- For users using the FXO port (FXS signaling) distinctive ring detection
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feature, you will need to adjust the dringX count values. The count
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values now only record ring end events instead of any DAHDI event. A
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ring-ring-ring pattern would exceed the pattern limits and stop
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Caller-ID detection.
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from 11.15 to 11.16
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chan_dahdi:
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- The CALLERID(ani2) value for incoming calls is now populated in featdmf
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signaling mode. The information was previously discarded.
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from 11.13.0 to 11.13.1:
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* Due to the POODLE vulnerability (see
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https://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2014-3566), the
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default TLS method for TLS clients will no longer allow SSLv3. As
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SSLv2 was already deprecated, it is no longer allowed by default as
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well. TLS servers no longer allow SSLv2 or SSLv3 connections. This
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affects the chan_sip channel driver, AMI, and the Asterisk HTTP server.
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* The res_jabber resource module no longer uses SSLv3 to connect to an
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XMPP server. It will now only use TLSv1 or later methods.
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from 11.10.2 to 11.11.0
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- Added a compatibility option for chan_sip, 'websocket_write_timeout'.
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When a websocket connection exists where Asterisk writes a substantial
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amount of data to the connected client, and the connected client is slow
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to process the received data, the socket may be disconnected. In such
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cases, it may be necessary to adjust this value. Default is 100 ms.
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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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- 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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from 11.10.0 to 11.10.1
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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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- Added http.conf session_inactivity timer option to close HTTP connections
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that aren't doing anything.
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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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from 11.9 to 11.10
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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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- 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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from 11.8 to 11.9
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- res_fax now returns the correct rates for V.27ter (4800 or 9600 bit/s).
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Because of this the default settings would not load, so the minrate (minimum
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transmission rate) option was changed to default to 4800 since that is the
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minimum rate for v.27 which is included in the default modem options.
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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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- When communicating with a peer on an Asterisk 1.4 or earlier system, the
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chan_iax2 parameter 'connectedline' must be set to "no" in iax.conf. This
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prevents an incompatible connected line frame from an Astersik 1.8 or later
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system from causing a hangup in an Asterisk 1.4 or earlier system. Note that
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this particular incompatibility has always existed between 1.4 and 1.8 and
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later versions; this upgrade note is simply informing users of its existance.
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- A compatibility setting, allow_empty_string_in_nontext, has been added to
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res_odbc.conf. When enabled (default behavior), empty column values are
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stored as empty strings during realtime updates. Disabling this option
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causes empty column values to be stored as NULLs for non-text columns.
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Disable it for PostgreSQL backends in order to avoid errors caused by
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updating integer columns with an empty string instead of NULL
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(sippeers, sipregs, ..).
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From 11.7 to 11.8:
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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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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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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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From 11.6 to 11.7:
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ConfBridge
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- ConfBridge now has the ability to set the language of announcements to the
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conference. The language can be set on a bridge profile in confbridge.conf
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or by the dialplan function CONFBRIDGE(bridge,language)=en.
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chan_sip - Clarify The "sip show peers" Forcerport Column And Add Comedia
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- Under the "Forcerport" column, the "N" used to mean NAT (i.e. Yes). With
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the additon of auto_* NAT settings, the meaning changed and there was a
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certain combination of letters added to indicate the current setting. The
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combination of using "Y", "N", "A" or "a", can be confusing. Therefore, we
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now display clearly what the current Forcerport setting is: "Yes", "No",
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"Auto (Yes)", "Auto (No)".
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- Since we are clarifying the Forcerport column, we have added a column to
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display the Comedia setting since this is useful information as well. We
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no longer have a simple "NAT" setting like other versions before 11.
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* Certain dialplan functions have been marked as 'dangerous', and may only be
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executed from the dialplan. Execution from extenal sources (AMI's GetVar and
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SetVar actions; etc.) may be inhibited by setting live_dangerously in the
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[options] section of asterisk.conf to no. SHELL(), channel locking, and direct
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file read/write functions are marked as dangerous. DB_DELETE() and
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REALTIME_DESTROY() are marked as dangerous for reads, but can now safely
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accept writes (which ignore the provided value).
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From 11.5 to 11.6:
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* res_agi will now properly indicate if there was an error in streaming an
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audio file. The result code will be -1 and the result returned from the
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the function will be RESULT_FAILURE instead of the prior behavior of always
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returning RESULT_SUCCESS even if there was an error.
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* The libuuid development library is now optional for res_rtp_asterisk. If the
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library is not present when building ICE and TURN support will not be present.
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* The option "register_retry_403" has been added to chan_sip to work around
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servers that are known to erroneously send 403 in response to valid
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REGISTER requests and allows Asterisk to continue attepmting to connect.
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Due to a failed merge, this option is present, but non-functional until 11.8.0.
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From 11.4 to 11.5:
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* The default settings for chan_sip are now overriden properly by the general
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settings in sip.conf. Please look over your settings upon upgrading.
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* It is now possible to play the Queue prompts to the first user waiting in a call queue.
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Note that this may impact the ability for agents to talk with users, as a prompt may
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still be playing when an agent connects to the user. This ability is disabled by
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default but can be enabled on an individual queue using the 'announce-to-first-user'
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option.
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* The libuuid development library is now required for res_rtp_asterisk. Consult
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your distribution for the appropriate development library name.
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From 11.3 to 11.4:
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* Added the 'n' option to MeetMe to prevent application of the DENOISE function
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to a channel joining a conference. Some channel drivers that vary the number
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of audio samples in a voice frame will experience significant quality problems
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if a denoiser is attached to the channel; this option gives them the ability
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to remove the denoiser without having to unload func_speex.
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* The Registry AMI event for SIP registrations will now always include the
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Username field. A previous bug fix missed an instance where it was not
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included; that has been corrected in this release.
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From 11.2.0 to 11.2.1:
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* Asterisk would previously not output certain error messages when a remote
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console attempted to connect to Asterisk and no instance of Asterisk was
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running. This error message is displayed on stderr; as a result, some
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initialization scripts that used remote consoles to test for the presence
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of a running Asterisk instance started to display erroneous error messages.
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The init.d scripts and the safe_asterisk have been updated in the contrib
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folder to account for this.
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From 11.2 to 11.3:
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* Now by default, when Asterisk is installed in a path other than /usr, the
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Asterisk binary will search for shared libraries in ${libdir} in addition to
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searching system libraries. This allows Asterisk to find its shared
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libraries without having to specify LD_LIBRARY_PATH. This can be disabled by
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passing --disable-rpath to configure.
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From 11.1 to 11.2:
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* Asterisk has always had code to ignore dash '-' characters that are not
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part of a character set in the dialplan extensions. The code now
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consistently ignores these characters when matching dialplan extensions.
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* Removed the queues.conf check_state_unknown option. It is no longer
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necessary.
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From 11.0 to 11.1:
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Queues:
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- Queue strategy rrmemory now has a predictable order similar to strategy
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rrordered. Members will be called in the order that they are added to the
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queue.
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From 10 to 11:
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Voicemail:
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- All voicemails now have a "msg_id" which uniquely identifies a message. For
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users of filesystem and IMAP storage of voicemail, this should be transparent.
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For users of ODBC, you will need to add a "msg_id" column to your voice mail
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messages table. This should be a string capable of holding at least 32 characters.
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All messages created in old Asterisk installations will have a msg_id added to
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them when required. This operation should be transparent as well.
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Parking:
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- The comebacktoorigin setting must now be set per parking lot. The setting in
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the general section will not be applied automatically to each parking lot.
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- The BLINDTRANSFER channel variable is deleted from a channel when it is
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bridged to prevent subtle bugs in the parking feature. The channel
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variable is used by Asterisk internally for the Park application to work
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properly. If you were using it for your own purposes, copy it to your
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own channel variable before the channel is bridged.
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res_ais:
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- Users of res_ais in versions of Asterisk prior to Asterisk 11 must change
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to use the res_corosync module, instead. OpenAIS is deprecated, but
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Corosync is still actively developed and maintained. Corosync came out of
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the OpenAIS project.
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Dialplan Functions:
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- MAILBOX_EXISTS has been deprecated. Use VM_INFO with the 'exists' parameter
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instead.
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- Macro has been deprecated in favor of GoSub. For redirecting and connected
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line purposes use the following variables instead of their macro equivalents:
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REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS,
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CONNECTED_LINE_SEND_SUB, CONNECTED_LINE_SEND_SUB_ARGS.
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- The REDIRECTING function now supports the redirecting original party id
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and reason.
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- The HANGUPCAUSE and HANGUPCAUSE_KEYS functions have been introduced to
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provide a replacement for the SIP_CAUSE hash. The HangupCauseClear
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application has also been introduced to remove this data from the channel
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when necessary.
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func_enum:
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- ENUM query functions now return a count of -1 on lookup error to
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differentiate between a failed query and a successful query with 0 results
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matching the specified type.
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CDR:
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- cdr_adaptive_odbc now supports specifying a schema so that Asterisk can
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connect to databases that use schemas.
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Configuration Files:
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- Files listed below have been updated to be more consistent with how Asterisk
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parses configuration files. This makes configuration files more consistent
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with what is expected across modules.
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- cdr.conf: [general] and [csv] sections
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- dnsmgr.conf
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- dsp.conf
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- The 'verbose' setting in logger.conf now takes an optional argument,
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specifying the verbosity level for each logging destination. The default,
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if not otherwise specified, is a verbosity of 3.
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AMI:
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- DBDelTree now correctly returns an error when 0 rows are deleted just as
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the DBDel action does.
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- The IAX2 PeerStatus event now sends a 'Port' header. In Asterisk 10, this was
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erroneously being sent as a 'Post' header.
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CCSS:
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- Macro is deprecated. Use cc_callback_sub instead of cc_callback_macro
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in channel configurations.
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app_meetme:
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- The 'c' option (announce user count) will now work even if the 'q' (quiet)
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option is enabled.
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app_followme:
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- Answered outgoing calls no longer get cut off when the next step is started.
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You now have until the last step times out to decide if you want to accept
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the call or not before being disconnected.
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chan_gtalk:
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- chan_gtalk has been deprecated in favor of the chan_motif channel driver. It is recommended
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that users switch to using it as it is a core supported module.
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chan_jingle:
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- chan_jingle has been deprecated in favor of the chan_motif channel driver. It is recommended
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that users switch to using it as it is a core supported module.
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SIP
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===
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- A new option "tonezone" for setting default tonezone for the channel driver
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or individual devices
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- A new manager event, "SessionTimeout" has been added and is triggered when
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a call is terminated due to RTP stream inactivity or SIP session timer
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expiration.
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- SIP_CAUSE is now deprecated. It has been modified to use the same
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mechanism as the HANGUPCAUSE function. Behavior should not change, but
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performance should be vastly improved. The HANGUPCAUSE function should now
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be used instead of SIP_CAUSE. Because of this, the storesipcause option in
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sip.conf is also deprecated.
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- The sip paramater for Originating Line Information (oli, isup-oli, and
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ss7-oli) is now parsed out of the From header and copied into the channel's
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ANI2 information field. This is readable from the CALLERID(ani2) dialplan
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function.
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- ICE support has been added and is enabled by default. Some endpoints may have
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problems with the ICE candidates within the SDP. If this is the case ICE support
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can be disabled globally or on a per-endpoint basis using the icesupport
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configuration option. Symptoms of this include one way media or no media flow.
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chan_unistim
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- Due to massive update in chan_unistim phone keys functions and on-screen
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information changed.
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users.conf:
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- A defined user with hasvoicemail=yes now finally uses a Gosub to stdexten
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as documented in extensions.conf.sample since v1.6.0 instead of a Macro as
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documented in v1.4. Set the asterisk.conf stdexten=macro parameter to
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invoke the stdexten the old way.
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res_jabber
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- This module has been deprecated in favor of the res_xmpp module. The res_xmpp
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module is backwards compatible with the res_jabber configuration file, dialplan
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functions, and AMI actions. The old CLI commands can also be made available using
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the res_clialiases template for Asterisk 11.
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From 1.8 to 10:
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cel_pgsql:
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- This module now expects an 'extra' column in the database for data added
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using the CELGenUserEvent() application.
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ConfBridge
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- ConfBridge's dialplan arguments have changed and are not
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backwards compatible.
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File Interpreters
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- The format interpreter formats/format_sln16.c for the file extension
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'.sln16' has been removed. The '.sln16' file interpreter now exists
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in the formats/format_sln.c module along with new support for sln12,
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sln24, sln32, sln44, sln48, sln96, and sln192 file extensions.
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HTTP:
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- A bindaddr must be specified in order for the HTTP server
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to run. Previous versions would default to 0.0.0.0 if no
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bindaddr was specified.
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Gtalk:
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- The default value for 'context' and 'parkinglots' in gtalk.conf has
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been changed to 'default', previously they were empty.
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chan_dahdi:
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- The mohinterpret=passthrough setting is deprecated in favor of
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moh_signaling=notify.
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pbx_lua:
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- Execution no longer continues after applications that do dialplan jumps
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|
(such as app.goto). Now when an application such as app.goto() is called,
|
|
control is returned back to the pbx engine and the current extension
|
|
function stops executing.
|
|
- the autoservice now defaults to being on by default
|
|
- autoservice_start() and autoservice_start() no longer return a value.
|
|
|
|
Queue:
|
|
- Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
|
|
- QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.
|
|
|
|
Asterisk Database:
|
|
- The internal Asterisk database has been switched from Berkeley DB 1.86 to
|
|
SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
|
|
utility in the UTILS section of menuselect. If an existing astdb is found and no
|
|
astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
|
|
convert an existing astdb to the SQLite3 version automatically at runtime. If
|
|
moving back from Asterisk 10 to Asterisk 1.8, the astdb2bdb utility can be used
|
|
to create a Berkeley DB copy of the SQLite3 astdb that Asterisk 10 uses.
|
|
|
|
Manager:
|
|
- The AMI protocol version was incremented to 1.2 as a result of changing two
|
|
instances of the Unlink event to Bridge events. This change was documented
|
|
as part of the AMI 1.1 update, but two Unlink events were inadvertently left
|
|
unchanged.
|
|
|
|
Module Support Level
|
|
- All modules in the addons, apps, bridge, cdr, cel, channels, codecs,
|
|
formats, funcs, pbx, and res have been updated to include MODULEINFO data
|
|
that includes <support_level> tags with a value of core, extended, or deprecated.
|
|
More information is available on the Asterisk wiki at
|
|
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
|
|
|
|
Deprecated modules are now marked to not build by default and must be explicitly
|
|
enabled in menuselect.
|
|
|
|
chan_sip:
|
|
- Setting of HASH(SIP_CAUSE,<slave-channel-name>) on channels is now disabled
|
|
by default. It can be enabled using the 'storesipcause' option. This feature
|
|
has a significant performance penalty.
|
|
- In order to improve compliance with RFC 3261, SIP usernames are now properly
|
|
escaped when encoding reserved characters. Prior to this change, the use of
|
|
these characters in certain SIP settings affecting usernames could cause
|
|
injections of these characters in their raw form into SIP headers which could
|
|
in turn cause all sorts of nasty behaviors. All characters that are not
|
|
alphanumeric or are not contained in the the following lists specified by
|
|
RFC 3261 section 25.1 will be escaped as %XX when encoding a SIP username:
|
|
* mark: "-" / "_" / "." / "!" / "~" / "*" / "'" / "(" / ")"
|
|
* user-unreserved: "&" / "=" / "+" / "$" / "," / ";" / "?" / "/"
|
|
|
|
UDPTL:
|
|
- The default UDPTL port range in udptl.conf.sample differed from the defaults
|
|
in the source. If you didn't have a config file, you got 4500 to 4599. Now the
|
|
default is 4000 to 4999.
|
|
|
|
===========================================================
|
|
===========================================================
|