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- update to 18.3.0
  * app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and
    MixMonitorMute when the channel monitoring is started, stopped and muted (or
    unmuted) respectively.
  * chan_iax2: You can now specify a default "auth" method in the
    [general] section of iax.conf
  * chan_pjsip, app_transfer: Added TRANSFERSTATUSPROTOCOL variable.
    performing a REFER.
  * Introduce an ARGC variable for func_odbc functions, along with a minargs
    per-function configuration option.
  * SRTP replay protection has been added to res_srtp and
    a new configuration option "srtpreplayprotection" has
    been added to the rtp.conf config file.

- Update to release 18.2.2
  * AST-2021-006 - res_pjsip_t38.c: Check for
    session_media on reinvite.

- 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

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

- update to 18.2.0:
  * Security
    - [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains

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- Update to 18.2.0:
+ Security bugs fixed in this release:
  * ASTERISK-29219 - res_pjsip_diversion: Crash if Tel URI
       contains History-Info (Reported by Torrey Searle)
+ Bugs fixed in this release:
  * ASTERISK-29229 - Stasis/messaging: text messages not
       dispatched to all subscribers when using generic subscription
       (Reported by Jean Aunis - Prescom)
  * ASTERISK-29240 - chan_pjsip: Incoming PJSIP calls set global
       SIPDOMAIN instead of a channel variable
       (Reported by Ivan Poddubny)
  * ASTERISK-29238 - chan_sip: SDP: Offers without any enabled
       stream are accepted.  (Reported by Alexander Traud)
  * ASTERISK-29237 - chan_sip: SDP: m=video is parsed even when
       disabled.  (Reported by Alexander Traud)
  * ASTERISK-29222 - chan_sip: Hold/Resume an sRTP call on a
       video enabled user-agent.  (Reported by Alexander Traud)
  * ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX
       responses (Reported by George Joseph)
  * ASTERISK-28016 - PJSIP sends duplicate 183 Progress
       responses (Reported by Alex Hermann)
  * ASTERISK-28185 - chan_pjsip: Subsequent same responses are
       not stopped (Reported by Julien)
  * ASTERISK-29230 - pjsip: Asterisk goes crazy and massively
       spams logfile if registration can't be send
       (Reported by Michael Maier)
  * ASTERISK-29231 - pjsip: SIGSEGV in CLI if no trunk is
       registered (Reported by Michael Maier)
  * ASTERISK-29217 - LOCK() can grant the same lock to multiple
       channels spuriously (Reported by Jaco Kroon)

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auto commit by copy to link target

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Asterisk 17.8.0
  * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
    events (Reported by Ove Aursand)
  * ASTERISK-29043 - app_queue: Leave empty sometimes not
    recorded as abandoned (Reported by Kfir Itzhak)
  * ASTERISK-29042 - res_parking: Parker UUID is no longer
    copied (Reported by Misha Vodsedalek)
  * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken
    asterisk 16 (Reported by Joseph Ades)
  * ASTERISK-29046 - pbx: Deadlock when doing a reload, while
    simultaneously doing an ExtensionState on a pattern match hint
    that ends up adding an extension (Reported by Ramarajan)
  * ASTERISK-29040 - res_speech: Assertion on format
    (Reported by Nickolay V. Shmyrev)
  * ASTERISK-29001 - chan_pjsip does not process or forward 181
    responses (Reported by Torrey Searle)
  * ASTERISK-29034 - Lastpause of realtime members is reseting
    (Reported by Evandro César Arruda)
  * ASTERISK-27273 - app_voicemail: When a voicemail is marked as
    "Urgent", it is not sent by email/processed by the mailcmd
    command (Reported by Leandro Dardini)
  * ASTERISK-29033 - res_pjsip_session: Aggressively terminates
    session on failed re-INVITE (Reported by Joshua C. Colp)
  * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have
    appended RTP string to each message block.
    (Reported by Thomas Johnson)

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- Add dahdi build conditional

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fix app_zapateller handling

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fix bcond typo

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- Add dahdi build conditional 
  dahdi-linux is bitrotten, and TW kernel is moving too fast to catch up
- Use proper gmime dependency

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- Update to release 17.7.0
  * [ASTERISK-29042] - res_parking: Parker UUID is no longer copied
  * [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
    doing an ExtensionState on a pattern match hint that ends up adding an extension
  * [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload
  * [ASTERISK-29021] - Fix VERSION(ASTERISK_VERSION_NUM) on certified versions
  * [ASTERISK-28927] - Asterisk crash in music on hold
  * [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
    INVITE when NAT is active (UDP transport with external_media_address)
  * [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured contacts is not correct
  * [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
  * [ASTERISK-28978] - acl: named_acl rule misconfiguration results in
    segfault on reading rule from realtime
  * [ASTERISK-28975] - res_http_websocket:
    Text payload data doesn't necessary include trailing zero

- Update to release 17.6.0
 * AMI: You can now specify an optional 'Content-Type' as an argument
   for the Asterisk SendText manager action.
 * res_pjsip: Added a new PJSIP system setting called disable_rport.
 * res_sorcery_memory_cache: The SorceryMemoryCacheExpireObject AMI
   action and CLI command allow expiring of a specific object within
   the sorcery memory cache.
 * res_ari_channels: When creating a channel in ARI using the create
   call you can now specify dialplan variables to be set as part of the
   same operation.
 * res_pjsip_logger: The PJSIP packet logger now has the following CLI
   commands:

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- Update to release 17.5.0
  * ASTERISK-28940 - /channels/create doesn't get any parameters
    from the body (Reported by sungtae kim)
  * ASTERISK-28932 - res_pjsip_logger writing too big packets
    (Reported by nappsoft)
  * ASTERISK-28921 - Wrong return value check for fwrite when
    writing to pcap file (Reported by nappsoft)
  * ASTERISK-28794 - res_pjsip: Crash when escaping during URI
    printing (Reported by nappsoft)
  * ASTERISK-28884 - x-ast-orig-host not filtered out from
    request URI and To header (Reported by nappsoft)
  * ASTERISK-28871 - res_pjsip_session: Unnecessary re-Invite on
    call answer (Reported by Alexei Gradinari)
  * ASTERISK-28903 - res_srtp: Answered Crypto Suite might be
    wrong in SDP/SDES. (Reported by Alexander Traud)
  * ASTERISK-28898 - bridge_softmix: Conference bridge not
    passing silent rtp packets (Reported by Jonathan Hunter)
  * ASTERISK-28892 - res_musiconhold: Module res_musiconhold
    throws false warning (Reported by Nicholas John Koch)
  * ASTERISK-28904 - RTP ICE leaks the memory
    (Reported by sungtae kim)
  * ASTERISK-26780 - res_pjsip: PJSIP Registration Fails when
    transport=transport-udp6 (Reported by Peter Sokolov)
  * ASTERISK-28854 - SIGSEGV when pjsip show history encounters
    IPV6 address (Reported by Roger James)
  * ASTERISK-28797 - [patch] tcptls: Fix notice when TLS is
    enabled but not configured. (Reported by Alexander Traud)
  * ASTERISK-28804 - [patch] app_osplookup.c: Avoid a format
    truncation. (Reported by Alexander Traud)
  * ASTERISK-28776 - Non async-signal-safe syscalls used after

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add asterisk include

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- Fix postgres build dependency for 15.2 and later

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