add changelog for enable chan_mobile
reduce gratuitous documentation

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Update to release 18.12.1

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Update to release 18.11.3

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Update to release 18.11.2

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- Update to release 18.11.1
  * [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't available
  * [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or xmlstarlet ro be installed when it shouldn't

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- Updated to jansson-2.14
- Update to release 18.11.0:

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- Update to release 18.10.1 also with many bug fixes and small improvements
  * Security fixes:
    - AST-2022-004: pjproject: integer underflow on STUN message
    - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
    - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
  * New Features 18.10.0:
    - [ASTERISK-29808] cdr: allow disabling CDR by default
    - [ASTERISK-29830] ami: Add AMI event for Wink
    - [ASTERISK-29802] app_sf: Add full tech-agnostic SF support
    - [ASTERISK-29759] app_sendtext: Add ReceiveText application
    - [ASTERISK-29706] func_json: Add JSON parsing function

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- Reenable build with support for DAHDI on supported platforms

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- Update to release 18.9.0
  * New Features
    - [ASTERISK-29720] - res_tonedetect: Add call progress tone detection
    - [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to Queue Members
  * Bugs fixed
    - [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
    - [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen
    - [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are configured and we have a schema warning
    - [ASTERISK-29776] - stir/shaken: Requires GNU designator
    - [ASTERISK-29764] - chan_misdn: Fix for Doxygen
    - [ASTERISK-29773] - progdocs: doxyref.h outdated
    - [ASTERISK-29765] - xmldoc: Fix for Doxygen
    - [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes
    - [ASTERISK-29762] - channels: Fix for Doxygen
    - [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in same bridge
    - [ASTERISK-29754] - odbc: Fix for Doxygen
    - [ASTERISK-29753] - parking: Fix for Doxygen
    - [ASTERISK-29755] - frame: Fix for Doxygen
    - [ASTERISK-29756] - res_ari: Fix for Doxygen
    - [ASTERISK-29751] - channel: Fix for Doxygen
    - [ASTERISK-29750] - stasis: Fix for Doxygen
    - [ASTERISK-29752] - app: Fix for Doxygen
    - [ASTERISK-29749] - res_xmpp: Fix for Doxygen
    - [ASTERISK-29742] - addons: Fix for Doxygen.
    - [ASTERISK-29747] - res_pjsip: Fix for Doxygen
    - [ASTERISK-29737] - chan_iax2: Fix for Doxygen
    - [ASTERISK-29743] - bridges: Fix for Doxygen
    - [ASTERISK-29741] - tests: Fix for Doxygen
    - [ASTERISK-29740] - apps: Fix for Doxygen
    - [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file
    - [ASTERISK-29736] - bridge_channel: Fix for Doxygen
    - [ASTERISK-29735] - progdocs: Avoid multiple use of section labels
    - [ASTERISK-29734] - progdocs: Use Doxygen \example correctly
    - [ASTERISK-29744] - app_morsecode: Fix deadlock
    - [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing
    - [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
    - [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
    - [ASTERISK-29702] - sig_analog: Fix truncated buffer copy
    - [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from dynamic hints
    - [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
    - [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
    - [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable dispatched.
    - [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized
    - [ASTERISK-29713] - GCC 11.2: two stringop-overread
    - [ASTERISK-29682] - Squash compiler issues generated by gcc 11
    - [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
    - [ASTERISK-27816] - func_talkdetect's logic is completely broken
    - [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
    - [ASTERISK-26497] - make install downloads x86_32 variants of external modules on non Intel architectures
  * Improvements
    - [ASTERISK-29777] - documentation: Standardize example syntax
    - [ASTERISK-29715] - app_voicemail: Refactor email generation functions
    - [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
    - [ASTERISK-29714] - Spelling errors
    - [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at dial time
    - [ASTERISK-29662] - Add mix option to Playback application for say and filename

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- Update to release 18.6.0

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Update to asterisk -18.4.0

Changes:
- Bug
  * Category: Applications/app_queue
   ASTERISK-28356: app_queue: CLI set ringinuse for realtime member not
   working
   Reported by: Michael
     * [35302efe73] Sean Bright -- app_queue: Add alembic migration to add
       ringinuse to queue_members.
   ASTERISK-24631: Incorrect description of option "context" in
   queues.conf.sample
   Reported by: Etienne Lessard
     * [31364fa4c8] Sean Bright -- queues.conf.sample: Correct 'context'
       documentation.
   ASTERISK-26614: app_queue: updatecdr option in queues.conf does
   effectively nothing
   Reported by: Alexander Gonchiy
     * [e27fa9eceb] Sean Bright -- app_queue.c: Remove dead 'updatecdr' code.
   ASTERISK-27542: app_queue: When "queue show" CLI command is executed a
   crash occurs
   Reported by: Miguel Sanz
     * [4393207751] Sean Bright -- app_queue.c: Don't crash when realtime
       queue name is empty.
   ASTERISK-29355: app_queue: Queue member status message sent even if status
   doesn't change
   Reported by: Roman Pertsev
     * [55c467eab1] Joshua C. Colp -- app_queue: Only send QueueMemberStatus
       if status changes.
  * Category: Bridges/bridge_simple
   ASTERISK-29379: Segfault - ast_channel_is_multistream (chan=0x0) at
   channel_internal_api.c:1590
   Reported by: Ross Beer
     * [88aec107df] George Joseph -- bridge_channel_write_frame: Check for
       NULL channel
  * Category: Channels/chan_local
   ASTERISK-29035: chan_local: Multistream support breaks T.38 faxing
   Reported by: Matthias Hensler
     * [ed2f637b47] Joshua C. Colp -- core_unreal: Fix deadlock with T.38
       control frames.
  * Category: Core/BuildSystem
   ASTERISK-29348: menuselect doesn't return errors in many cases
   Reported by: George Joseph
     * [f47c5cbdf9] Jaco Kroon -- menuselect: exit non-zero in case of
       failure on --enable|disable options.
  * Category: Core/CodecInterface
   ASTERISK-29328: translate.c: possible buffer overflow when upsampling
   Reported by: Jean Aunis - Prescom
     * [dec44306cf] Jean Aunis -- translate.c: Take sampling rate into
       account when checking codec's buffer size
  * Category: Core/Stasis
   ASTERISK-29355: app_queue: Queue member status message sent even if status
   doesn't change
   Reported by: Roman Pertsev
     * [55c467eab1] Joshua C. Colp -- app_queue: Only send QueueMemberStatus
       if status changes.
  * Category: Documentation
   ASTERISK-24434: Fix differing usage of assignment operators in
   modules.conf
   Reported by: Rusty Newton
     * [be3153346b] Sean Bright -- modules.conf: Fix more differing usages of
       assignment operators.
   ASTERISK-24631: Incorrect description of option "context" in
   queues.conf.sample
   Reported by: Etienne Lessard
     * [31364fa4c8] Sean Bright -- queues.conf.sample: Correct 'context'
       documentation.
   ASTERISK-25358: dateformat not read from logger.conf by remote console
   Reported by: Igor Liferenko
     * [a0009c807e] Mark Murawski -- logger: Console sessions will now
       respect logger.conf dateformat= option
  * Category: Resources/General
   ASTERISK-29130: prometheus: Crash when scraping bridge
   Reported by: Francisco Correia
     * [19eef2a6dc] George Joseph -- res_prometheus: Clone containers before
       iterating
  * Category: Resources/res_pjsip
   ASTERISK-29354: res_pjsip: Allow partial reloading of transports
   Reported by: Joshua C. Colp
     * [f213833514] Joshua C. Colp -- res_pjsip: Add support for partial
       transport reload.
  * Category: Resources/res_pjsip_session
   ASTERISK-29215: res_pjsip_session: NULL active_media_state topology caused
   asterisk crash
   Reported by: sungtae kim
     * [c78d0ce429] George Joseph -- res_pjsip_session: Make
       reschedule_reinvite check for NULL topologies
  * Category: Resources/res_rtp_asterisk
   ASTERISK-29364: res_rtp_asterisk: standard deviation miscalculation
   Reported by: Kevin Harwell
     * [17c86dcfaa] Kevin Harwell -- res_rtp_asterisk: Fix standard deviation
       calculation
   ASTERISK-29373: res_rtp_asterisk: Flash events are duplicated
   Reported by: N A
     * [b0d828f14a] Joshua C. Colp -- res_rtp_asterisk: Only raise flash
       control frame on end.
   ASTERISK-29352: res_rtp_asterisk: Fix frame delivery time when SSRC
   changes
   Reported by: Joshua C. Colp
     * [2e7fc84398] Joshua C. Colp -- res_rtp_asterisk: Force resync on SSRC
       change.
- Improvement
  * Category: Core/General
   ASTERISK-29339: loader: Let's output warnings for deprecated modules!
   Reported by: Joshua C. Colp
     * [a9a9864478] Joshua C. Colp -- loader: Output warnings for deprecated
       modules.
   ASTERISK-29337: menuselect: Add ability to set deprecated in and removed
   in versions for modules
   Reported by: Joshua C. Colp
     * [6aac148d59] Joshua C. Colp -- menuselect: Add ability to set
       deprecated and removed versions.
     * [60fb559ccc] Joshua C. Colp -- xml: Allow deprecated_in and removed_in
       for MODULEINFO.
   ASTERISK-29335: xml: Embed module information into core XML documentation.
   Reported by: Joshua C. Colp
     * [60800b038a] Joshua C. Colp -- xml: Embed module information into core
       XML documentation.
  * Category: Documentation
   ASTERISK-29336: documentation: Fix inconsistent support levels
   Reported by: Joshua C. Colp
     * [be3e469f98] Joshua C. Colp -- documentation: Fix non-matching module
       support levels.
   ASTERISK-29335: xml: Embed module information into core XML documentation.
   Reported by: Joshua C. Colp
     * [60800b038a] Joshua C. Colp -- xml: Embed module information into core
       XML documentation.

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update to 18.3.0

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- Update to release 18.2.2

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- Update to 18.2.1 with security fixes:
  * AST-2021-001: Remote crash in res_pjsip_diversion
  * AST-2021-002: Remote crash possible when negotiating T.38
  * AST-2021-003: Remote attacker could prematurely tear down SRTP calls
  * AST-2021-004: An unsuspecting user could crash Asterisk with multiple
  * AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver

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- Cut build recipe parts for platforms older than SLE/Leap 15

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fix missed bullet points

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- Fix the year in changes file
- Remove insserv dependency on newer systems using systemd

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- update to 18.2.0:
  * Security
    - [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
  * Bug
    - [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
    - [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy
    - [ASTERISK-29155] - app_queue: Deadlock between queues container and individual queues
    - [ASTERISK-29161] - Incorrect setup of recall channels
    - [ASTERISK-29168] - Asterisk crashes during call transfer
    - [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable
    - [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
    - [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses
    - [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
    - [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send
    - [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the Transfer result
    - [ASTERISK-29210] - res_pjsip: Crash when examining transport
    - [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
    - [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are accepted.
    - [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
    - [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled user-agent.
    - [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server.
    - [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are accepted.
    - [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
    - [ASTERISK-29209] - Debug messages printed by scope trace might be missing newlines
    - [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels spuriously
    - [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend
    - [ASTERISK-29188] - null media causing the Asterisk crash
    - [ASTERISK-29173] - Media cache URL requests allow infinite redirects
    - [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without entries
    - [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS response
    - [ASTERISK-29191] - tel: URI in Diversion header causes crash
    - [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered
    - [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable
    - [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription
    - [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description
    - [ASTERISK-29191] - tel: URI in Diversion header causes crash
    - [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
  * Improvement
    - [ASTERISK-29118] - VoiceMail() should have an option to play greetings as Early Media
    - [ASTERISK-28549] - Two repeated 183
    - [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other  newer linux versions
    - [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in /tmp
    - [ASTERISK-28549] - Two repeated 183

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- Update for 18.1.1:
  * Security bugs fixed:
    - [AST-2020-001] - res_pjsip: Return dialog locked and referenced
    - [AST-2020-002] - res_pjsip: Stop sending INVITEs after challenge limit.

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