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-------------------------------------------------------------------
Thu May 13 21:04:27 UTC 2021 - Diederik de Groot <ddegroot [at] users.sourceforge.net>

- 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.

-------------------------------------------------------------------
Fri Mar 26 12:25:27 UTC 2021 - Michael Ströder <michael@stroeder.com>

- 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.

-------------------------------------------------------------------
Sun Mar 14 22:20:25 UTC 2021 - Jan Engelhardt <jengelh@inai.de>

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

-------------------------------------------------------------------
Thu Feb 18 19:38:20 UTC 2021 - Michael Ströder <michael@stroeder.com>

- 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

-------------------------------------------------------------------
Sat Feb 13 19:58:22 UTC 2021 - Jan Engelhardt <jengelh@inai.de>

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

-------------------------------------------------------------------
Tue Feb 13 09:38:39 UTC 2021 - Asterisk Team <asteriskteam@digium.com>

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

-------------------------------------------------------------------
Tue Dec 22 09:58:39 2020 - Torrey Searle <tsearle@voxbone.com>

- 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.

-------------------------------------------------------------------
Thu Nov 19 14:01:31 UTC 2020 - Michael Ströder <michael@stroeder.com>

- update to 17.9.0:
  * Security bugs fixed:
    - [ASTERISK-29057] - pjsip: Crash on call rejection during high load
  * Improvements:
    - [ASTERISK-29055] - Create a Bridge with video_single mode
    - [ASTERISK-29056] - Increase reg_server column size for ps_contacts table realtime
  * many more bug fixes

-------------------------------------------------------------------
Fri Nov  6 09:08:03 UTC 2020 - Michael Ströder <michael@stroeder.com>

- update to 17.8.1 with security fixes:
  * AST-2020-001: Remote crash in res_pjsip_session
  * AST-2020-002: Outbound INVITE loop on challenge with different nonce.

-------------------------------------------------------------------
Fri Oct 23 08:26:25 UTC 2020 - Hans-Peter Jansen <hpj@urpla.net>

- 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)

-------------------------------------------------------------------
Mon Oct 19 15:55:34 UTC 2020 - Hans-Peter Jansen <hpj@urpla.net>

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

-------------------------------------------------------------------
Thu Sep  3 09:30:15 UTC 2020 - Michael Ströder <michael@stroeder.com>

- 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

-------------------------------------------------------------------
Fri Jul 17 16:06:36 UTC 2020 - Diederik de Groot <ddegroot@talon.nl>

- 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:

-------------------------------------------------------------------
Sat Jun  6 07:00:36 UTC 2020 - Jan Engelhardt <jengelh@inai.de>

- Update to release 17.4.0
  * ARI: Application event filtering is now supported. An
    application can now specify an "allowed" and/or "disallowed"
    list(s) of event types.
  * AttendedTransfer: A new application, this will queue up
    attended transfer to the given extension.
  * BlindTransfer: A new application, this will redirect all
    channels currently bridged to the caller channel to the
    specified destination.
  * ConfBridge: Add "average_all", "highest_all", and
    "lowest_all" values for the remb_behavior option. These
    values operate on a bridge level instead of a per-source
    level.
  * Dial: Add RINGTIME and RINGTIME_MS variables containing
    respectively seconds and milliseconds between creation of the
    dialing channel and receiving the first RINGING signal.
  * Dial: Add PROGRESSTIME and PROGRESSTIME_MS variables
    analogous to the above with respect to the PROGRESS signal.
    Shorter of these two times should be equivalent to the PDD
    (Post Dial Delay) value.
  * Dial: Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get
    millisecond resolution versions of DIALEDTIME and
    ANSWEREDTIME.

-------------------------------------------------------------------
Thu Mar 12 18:37:36 UTC 2020 - Hans-Peter Jansen <hpj@urpla.net>

- Update to new upstream release 16.8.0
  + Bugs fixed in this release:
   * ASTERISK-28766 - PJSIP blind transfer not completed after
     using Proceeding() (Reported by lvl)
   * ASTERISK-28685 - check_expr2: linking (when hardening) and
     cross-compiling troubles (Reported by Sebastian Kemper)
   * ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and
     seqno handling (Reported by Joshua C. Colp)
   * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in
     the "variables" field (Reported by Jean Aunis - Prescom)
   * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After
     Hold (Reported by Ross Beer)
   * ASTERISK-28697 - res_pjsip: Named ACL does not update on
     reload if changed (Reported by Timothy Vanderaerden)
   * ASTERISK-28746 - res_pjsip_outbound_registration keeps
     retrying the first entry in a SRV record set
     (Reported by George Joseph)
   * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to
     complete before allowing sending
     (Reported by Benjamin Keith Ford)
   * ASTERISK-28738 - Incorrect state machine used when
     MOH_PASSTHRU is used (Reported by Torrey Searle)
   * ASTERISK-28742 - res_rtp_asterisk: static for audio due to
     incomplete dtls/srtp setup (Reported by Kevin Harwell)
   * ASTERISK-28735 - Realtime MoH Unknown format '' --
     defaulting to SLIN (Reported by Ross Beer)
   * ASTERISK-28730 - res_pjsip_session: Fix out of order
     session refreshes (Reported by Joshua C. Colp)
   * ASTERISK-28718 - chan_sip: Returns 403 if RTP ports are
     depleted, should return 503 (Reported by Walter Doekes)
   * ASTERISK-28719 - Cannot remove defaultrule from queue using
     realtime queues (Reported by EDV O-TON)
   * ASTERISK-28713 - res_stasis_playback: Error building JSON
     (Reported by Sébastien Duthil)
   * ASTERISK-28714 - REGRESSION: Feature
     subscription_persistence_recreate (ASTERISK-27759) Causes
     Segfaults (Reported by Ross Beer)
   * ASTERISK-26082 - res_pjsip_messaging: MessageSend
     Content-Type can't be changed (Reported by Alex)
   * ASTERISK-28423 - ARI causes STASIS Deadlock
     (Reported by Ross Beer)
   * ASTERISK-28679 - stasis application is destroyed after its
     creation (Reported by Francois Blackburn)
   * ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS
     in spite of the error when sending
     (Reported by Dmitriy Serov)
   * ASTERISK-28686 - chan_sip strictrtp=yes fails when media
     source is changed: no audio (Reported by Walter Doekes)
   * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes
     Asterisk To Drop Calls (Reported by Paul Brooks)
   * ASTERISK-26955 - pjsip: SIP Packets with Via "received="
     Containing IPv6 Address Delimited by "[]" Rejected
     (Reported by Peter Sokolov)
  + Improvements made in this release:
    * ASTERISK-28750 - TLS/SSL Key too small error
      (Reported by Martin Zeh)
    * ASTERISK-28733 - stream: Add support for adding/removing
      streams during SFU/calls (Reported by Joshua C. Colp)
    * ASTERISK-24798 - Documentation - Clarify That Format Is Set
      By File Name Extension In MixMonitor (Reported by xrobau)
    * ASTERISK-28726 - install_prereq script uses the interactive
      mode when installing aptitude (Reported by Sylvain Afchain)

-------------------------------------------------------------------
Wed Feb  5 19:39:12 UTC 2020 - Hans-Peter Jansen <hpj@urpla.net>

- Update to new upstream release 16.8.0
  + New Features made in this release:
    * ASTERISK-17491 - CURLOPT() needs a "followlocation"
      parameter / "maxredirs" doesn't do anything
      (Reported by candrews)
    * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add
      ability to match on source port
      (Reported by Sean Bright)
  + Bugs fixed in this release:
    * ASTERISK-28679 - stasis application is destroyed after its
      creation (Reported by Francois Blackburn)
    * ASTERISK-28423 - ARI causes STASIS Deadlock
      (Reported by Ross Beer)
    * ASTERISK-28714 - REGRESSION: Feature
      subscription_persistence_recreate (ASTERISK-27759) Causes
      Segfaults (Reported by Ross Beer)
    * ASTERISK-28677 - CDR billsec is always 0 for transferred
      calls (Reported by Maciej Michno)
    * ASTERISK-28702 - chan_dahdi: holding a channel via flash to
      dialtone times out after 0:16:40 (Reported by Andrew Siplas)
    * ASTERISK-28706 - silk 24hHz doesn't show up in 'core show
      translation' output (Reported by Sean Bright)
    * ASTERISK-24484 - Update documentation for statsd module -
      usage requirements unclear (Reported by Dan Jenkins)
    * ASTERISK-28695 - core: minmemfree watermark uses free RAM,
      not available RAM (Reported by Kevin Flyn)
    * ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a
      whitespace appears empty in the dialplan
      (Reported by Frank Matano)
    * ASTERISK-23739 - [patch]Segfault forwarding voicemail with
      ODBC storage enabled and realtime voicemail_data is used
      (Reported by Stas Kobzar)
    * ASTERISK-27622 - empty voicemail.conf required for ARA
      (realtime) voicemail to leave message
      (Reported by Jim Van Meggelen)
    * ASTERISK-28349 - Pause reason not reported in QueueMember
      AMI event (Reported by Niksa Baldun)
    * ASTERISK-21794 - CLI command 'realtime update2' syntax
      failure when using according to usage help
      (Reported by Cedric BASSAGET)
    * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document
      support for hostnames (Reported by Joshua C. Colp)
    * ASTERISK-27775 - res_pjsip_notify: Multiple Event headers
      can be present instead of just one (Reported by AvayaXAsterisk)
    * ASTERISK-28682 - app_record: Lack of `beep` audio file
      causes application to return error and hangup
      (Reported by Corey Farrell)
    * ASTERISK-28507 - Wiki docs missing for MessageWaiting
      (Reported by David M. Lee)
    * ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence
      does not preserve XML <dialog-info> version number
      (Reported by Bryan Nelson)
    * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios
      with concurrent command pri show span X
      (Reported by Dirk Wendland)
    * ASTERISK-28633 - stasis bridge topic leak
      (Reported by Joeran Vinzens)
    * ASTERISK-28492 - pjsip reload not reloading wizard
      endpoint/pickup_group endpoint/call_group
      (Reported by Jean-Denis Girard)
    * ASTERISK-28562 - SIP WSS message not processed until next
      frame arrives (Reported by Robert Sutton)
    * ASTERISK-27243 - contrib: valgrind.supp doesn't suppress
      what it's supposed to due to invalid syntax
      (Reported by Richard Kenner)
    * ASTERISK-28497 - func_odbc: truncating Unicode string on
      readsql (Reported by Boris P. Korzun)
    * ASTERISK-28647 - chan_sip: RTP frames not transmitted after
      emitting a COLP (Reported by Jean Aunis - Prescom)
    * ASTERISK-28667 - Asterisk ignores parsing of config files
      if a Byte order mark is present (Reported by Robin Leffmann)
    * ASTERISK-28664 - "trustrpid" is misspelled in
      sip_to_pjsip.py (Reported by Pascal Cadotte Michaud)
    * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql
      don't build on 17.0.0 (Reported by George Joseph)
    * ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes
      non-existent media stream if codecs create additional streams
      and offer does not have them (Reported by nappsoft)
    * ASTERISK-28660 - res_fax: wrap Asterisk initiated
      negotiation with config option (Reported by Kevin Harwell)
    * ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes
      fails to deactivate CDR.  (Reported by Frederic LE FOLL)
    * ASTERISK-28626 - Missing arguments in PJSIP_CONTACT
      function documentation (Reported by Pascal Cadotte Michaud)
    * ASTERISK-28609 - Memory Leak in res_rtp_asterisk.c
      (Reported by Ted G)
    * ASTERISK-28651 - chan_sip logs errors on tx to non-existent
      TCP connections (Reported by Jaco Kroon)
    * ASTERISK-28502 - chan_pjsip incorrectly re-writes REGISTER
      200 Response Contact (Reported by Ross Beer)
    * ASTERISK-28625 - Playback of local files impacted by large
      media cache (Reported by Kevin Reeves)
  + Improvements made in this release:
    * ASTERISK-28710 - Should be able to disable the /httpstatus
      URI in the built-in HTTP server (Reported by Sean Bright)
    * ASTERISK-28638 - Simplify dialplan for Dial, Page, and
      ChanIsAvail (Reported by cmaj)
    * ASTERISK-28673 - GET FULL VARIABLE documentation
      clarification (Reported by Jonathan Harris)
    * ASTERISK-28658 - app_confbridge: Add support for setting
      maximum sample rate (Reported by Joshua C. Colp)

-------------------------------------------------------------------
Wed Dec 25 13:06:01 UTC 2019 - Hans-Peter Jansen <hpj@urpla.net>

- Update to new upstream release 16.7.0
  + Security bugs fixed in this release:
    * ASTERISK-28589 - chan_sip: Depending on configuration an
      INVITE can alter Addr of a peer (Reported by Andrey  V.T.)
    * ASTERISK-28580 - Bypass SYSTEM write permission in manager
      action allows system commands execution
      (Reported by Eliel Sardañons)
  + Improvements made in this release:
    * ASTERISK-28602 - res_pjsip_outbound_registration: Maximum
      retries reached (Reported by Daniel)
    * ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md
      (Reported by Sam Banks)
    * ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC
      backend when format differs from attachfmt column
      (Reported by cmaj)
    * ASTERISK-28567 - Problem with ASTERISK-20207: Asterisk
      should clear out any .lock files in the voice mail directory on
      startup. (Reported by Michael)
    * ASTERISK-28542 - [patch] add the ability for asterisk to
      generate on-hold re-invites (Reported by Torrey Searle)
    * ASTERISK-28512 - Add pass-through support for H.265 (HEVC)
      codec (Reported by Florian Floimair)
  + Bugs fixed in this release:
    * ASTERISK-28609 - Memory Leak in res_rtp_asterisk.c
      (Reported by Ted G)
    * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql
      don't build on 17.0.0 (Reported by George Joseph)
    * ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes
      non-existent media stream if codecs create additional streams
      and offer does not have them
      (Reported by nappsoft)
    * ASTERISK-28641 - res_pjsip Segfaults when realtime
      configuration to an AOR points to a not existent AOR
      (Reported by Ross Beer)
    * ASTERISK-28644 - Stale comment in app_queue about
      ring_entry exception (Reported by Walter Doekes)
    * ASTERISK-28445 - res_pjsip_session: ast_json_vpack: Invalid
      UTF-8 string on hangup when TEST_FRAMEWORK enabled
      (Reported by Bernhard Schmidt)
    * ASTERISK-28637 - chan_sip+native_bridge_rtp: directmedia
      compatibility check failure when negociated ptime is not
      default ptime.  (Reported by Frederic LE FOLL)
    * ASTERISK-28631 - res_parking: Doesn't park when parkee and
      parker are the same (Reported by Ross Beer)
    * ASTERISK-28621 - Enforce T.38 error correction mode at 200
      ok received (Reported by Salah Ahmed)
    * ASTERISK-28624 - res_pjsip_outbound_registration: add SRV
      failover (Reported by Kevin Harwell)
    * ASTERISK-28608 - app_amd: Use time calculation to calculate
      timeout (Reported by Michael Cargile)
    * ASTERISK-28615 - chan_dahdi: PRI span status may stay
      "Down, Active" after a short alarm
      (Reported by Frederic LE FOLL)
    * ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash
      when sent packet length doesn't match
      (Reported by Joshua Elson)
    * ASTERISK-26481 - FILE function grabs garbage along with
      read data when target line has no newline
      (Reported by Jonathan Harris)
    * ASTERISK-28618 - bridge_softmix: hold not cleared when
      joining a softmix bridge (Reported by Kevin Harwell)
    * ASTERISK-28616 - parking: Deadlock when multi call parking
      (Reported by Joshua C. Colp)
    * ASTERISK-28423 - ARI causes STASIS Deadlock
      (Reported by Ross Beer)
    * ASTERISK-28572 - Memory leaks in res_calendar_exchange and
      res_calendar_icalendar (Reported by Yoooooo Ha)
    * ASTERISK-28585 - ari/resource_events: Crash in event
      session cleanup (Reported by Kevin Harwell)
    * ASTERISK-28590 - utils.c throws repeated warnings
      "pthread_attr_setstacksize: Invalid argument"
      (Reported by Speed Dial Dave)
    * ASTERISK-28578 - race condition on pjsip channelstats
      command (Reported by Salah Ahmed)
    * ASTERISK-28571 - cdr_pgsql: accesses obsolete (and finally
      removed) column (Reported by Christoph Moench-Tegeder)
    * ASTERISK-28575 - MWI Send Notify Crash on 16.6
      (Reported by Joshua Elson)
    * ASTERISK-28574 - pjproject fails to build on 16.6.0, works
      on 16.5 (Reported by Niklas Larsson)
    * ASTERISK-28561 - Asterisk Deadlocks
      (Reported by Aheliotech)
    * ASTERISK-28552 - res_pjsip_mwi: Frack during unload on
      unsolicited_mwi container (Reported by Kevin Harwell)
    * ASTERISK-28566 - CDR backend unload problem during active
      call(s) (Reported by Marian Piater)
    * ASTERISK-28553 - stasis.c: Crash during unload
      (Reported by Kevin Harwell)
    * ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF
      over AMI (Reported by Jeremiah Gadd)
    * ASTERISK-28544 - Wrong contact representation in ipv6 mode
      (Reported by Jørgen H)
    * ASTERISK-28534 - Segmentation fault when there is no
      priority for an extension (Reported by Timothy Vanderaerden)
    * ASTERISK-28463 - res_pjsip_path: Crash when invalid contact
      is configured (Reported by Juan Martin)
    * ASTERISK-28521 - pjsip: Memory Leak
      (Reported by Mark)
    * ASTERISK-28523 - Asterisk 16.5.0 Memory leak
      (Reported by Cyril Ramière)
    * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
      (Reported by Joshua C. Colp)
    * ASTERISK-28536 - Asterisk release candidates fail to build
      on FreeBSD (Reported by Guido Falsi)
    * ASTERISK-23756 - setvar directive when used in template and
      a child of said template, results in duplicate variable names
     (Reported by Michael Goryainov)
  + New Features made in this release:
    * ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with
      PlayDTMF instead of only "sending" (Reported by lvl)
    * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type
      header (Reported by Martin Tomec)
    * ASTERISK-28533 - func_jitterbuffer: Add support for video
      synchronization (Reported by Joshua C. Colp)

-------------------------------------------------------------------
Fri Nov 22 10:34:03 UTC 2019 - Hans-Peter Jansen <hpj@urpla.net>

- Update to new upstream release 16.6.2
  * ASTERISK-28580
    manager.c:  Prevent the Originate action from running the Originate app
    If an AMI user without the "system" authorization calls the
    Originate AMI command with the Originate application,
    the second Originate could run the "System" command.
    Action: Originate
    Channel: Local/1111
    Application: Originate
    Data: Local/2222,app,System,touch /tmp/owned
    If the "system" authorization isn't set, we now block the
    Originate app as well as the System, Exec, etc. apps.

* ASTERISK-28589 #close
    chan_sip.c: Prevent address change on unauthenticated SIP request.
    If the name of a peer is known and a SIP request is sent using that
    peer's name, the address of the peer will change even if the request
    fails the authentication challenge. This means that an endpoint can
    be altered and even rendered unusuable, even if it was in a working
    state previously. This can only occur when the nat option is set to the
    default, or auto_force_rport.
    This change checks the result of authentication first to ensure it is
    successful before setting the address and the nat option.

- Update to new upstream release 16.6.1
  * ASTERISK-28574
    pjproject_bundled:  Replace earlier reverts with official fixes.
    Issues in pjproject 2.9 caused us to revert some of their changes
    as a work around.  This introduced another issue where pjproject
    wouldn't build with older gcc versions such as that found on
    CentOS 6.  This commit replaces the reverts with the official
    fixes for the original issues and allows pjproject to be built
    on CentOS 6 again.

* ASTERISK-28575
    res_pjsip_mwi: potential double unref, and potential unwanted double link
    When creating an unsolicited MWI aggregate subscription it was possible for
    the subscription object to be double unref'ed. This patch removes the explicit
    unref as it is not needed since the RAII_VAR will handle it at function end.
    Less concerning there was also a bug that could potentially allow the aggregate
    subscription object to be added to the unsolicited container twice. This patch
    ensures it is added only once.

-------------------------------------------------------------------
Sun Oct 13 15:40:20 UTC 2019 - Hans-Peter Jansen <hpj@urpla.net>

- Update to new upstream release 16.6.0
  - Security bugs fixed in this release:
    * [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with
      declined stream causes crash (Reported by Alexei Gradinari)
  - Bugs fixed in this release:
    * [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
    * [ASTERISK-28523] - Asterisk 16.5.0 Memory leak 
      (Reported by Cyril Ramière)
    * [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection
      (Reported by Joshua C. Colp)
    * [ASTERISK-28536] - Asterisk release candidates fail to build
      on FreeBSD (Reported by Guido Falsi)
    * [ASTERISK-28511] - codec_resample: Bad sound quality when up
      sampling from SLIN16 to SLIN32 (Reported by Ruddy G)
    * [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when
      a PRI channel hangs up (Reported by Frederic LE FOLL)
    * [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes
      is set in cdr.conf (Reported by Frederic LE FOLL)
    * [ASTERISK-28499] - translate: Crash when frame does not have a
      "src" field set (Reported by Gregory Massel)
    * [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized
      type not at end of a struct (Reported by Alexander Traud)
    * [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register
      (Reported by Chris Savinovich)
    * [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36
      characters, NEC only supports up to 32 characters
      (Reported by Dan Cropp)
    * [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail
      because not checking mailstream (Reported by Alexei Gradinari)
    * [ASTERISK-28487] - compile menuselect on gentoo
      (Reported by Kilburn)
    * [ASTERISK-28472] - Asterisk occasionally passes a NULL as
      srtp->session to srtp_protect/unprotect causing SEGV
      (Reported by Jonas Swiatek)
    * [ASTERISK-28498] - cel / cdr: Event times may be incorrect
      (Reported by Joshua C. Colp)
    * [ASTERISK-28480] - json integer overflow in ssrc and timestamp
      (Reported by Salah Ahmed)
    * [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double
      entries (Reported by Ian Jones)
    * [ASTERISK-28483] - packet lost on UDPTL wrap around
      (Reported by Torrey Searle)
    * [ASTERISK-28477] - Crash when not specifying "dbfile" in
      res_config_sqlite3.conf (Reported by Dennis)
    * [ASTERISK-28478] - Crash performing "core reload" with modified
      res_config_sqlite3.conf (Reported by Dennis)
    * [ASTERISK-26968] - chan_pjsip: Transfer() does not result in
      TRANSFERSTATUS reflecting SIP response to transfer
      (Reported by Dan Cropp)
    * [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self
      deadlocks (in chan_sip) (Reported by Walter Doekes)
  - New Features made in this release:
    * [ASTERISK-17808] - [patch] Unregister a realtime moh class
      (Reported by Byron Clark)
    * [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip
      to setup From header URI domain (Reported by Stas Kobzar)

-------------------------------------------------------------------
Fri Sep 20 08:53:46 UTC 2019 - Hans-Peter Jansen <hpj@urpla.net>

- Update to new upstream release 16.5.1
  - Security bugs fixed in this release:
    * AST-2019-005 - translate: Don't assume all frames will have a src.
      This change removes the assumption that a frame will always have
      a src set on it. This assumption is incorrect.
    * AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media
      After receiving a 200 OK with a declined stream in response to a T.38
      initiated re-invite Asterisk would crash when attempting to dereference
      a NULL session media object.

-------------------------------------------------------------------
Tue Aug 13 07:04:15 UTC 2019 - Hans-Peter Jansen <hpj@urpla.net>

- Update to new upstream release 16.5.0
  - Security bugs fixed in this release:
    * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with
      no body causes crash (Reported by Gil Richard)
    * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38
      reINVITE (Reported by Francesco Castellano)
  - Bugs fixed in this release:
    * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit
      systems caused by ASTERISK-28317 (Reported by abelbeck)
    * ASTERISK-28458 - res_pjsip_sdp_rtp: Remove unused variable
      (Reported by Michael Maier)
    * ASTERISK-26006 - Show offending IP for TLS setup failures in
      logs (Reported by Oleksandr Natalenko)
    * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors
      not logged (Reported by Bernhard Schmidt)
    * ASTERISK-28419 - app_amd: Does not work with silence
      suppression (Reported by Nasir Iqbal)
    * ASTERISK-28018 - IP Fragmentation happening instead of DTLS
      fragmentation on handshake server hello certificate
      (Reported by vijay kumar)
    * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when
      Asterisk attempts to generate hangup event
      (Reported by Abhay Gupta)
    * ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work
      (Reported by Dmitry Svyatogorov)
    * ASTERISK-27981 - res_fax: Fax session leak with fax
      gatewaying (Reported by pasandev)
    * ASTERISK-28427 - new mwi.h include missing from some dahdi
      source files, causes build failure
      (Reported by Guido Falsi)
    * ASTERISK-28421 - Wrong type used for timestamp in
      res_rtp_asterisk (Reported by Morten Tryfoss)
    * ASTERISK-27994 - PJSIP: Early media ringback not indicated
      after Progress() (Reported by Gregory Massel)
  - Improvements made in this release:
   * ASTERISK-28234 - pbx_dundi: Add IPv4/IPv6 dual bind support
    for DUNDi (Reported by Kirsty Tyerman)

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.0

- Update bundled pjproject tarball to 2.9

-------------------------------------------------------------------
Thu Jul 25 16:13:06 UTC 2019 - Hans-Peter Jansen <hpj@urpla.net>

- Add postgresql-server-devel dependency for Factory

-------------------------------------------------------------------
Fri Jul 12 09:41:03 UTC 2019 - Hans-Peter Jansen <hpj@urpla.net>

- Update to new upstream release 16.4.1
  * AST-2019-002: Remote crash vulnerability with MESSAGE messages
    A specially crafted SIP in-dialog MESSAGE message can cause Asterisk
    to crash.

* AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver
    When T.38 faxing is done in Asterisk a T.38 reinv ite may be sent to an
    endpoint to switch it to T.38. If the endpoint responds with an improperly
    formatted SDP answer including both a T.38 UDPTL stream and an audio or video
    stream containing only codecs not allowed on the SIP peer or user a crash will
    occur. The code incorrectly assumes that there will be at least one common
    codec when T.38 is also in the SDP answer. Fixes CVE-2019-13161.

https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1

- Update to new upstream release 16.4.0
  https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.0

- Update to new upstream release 16.3.0
  https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.3.0

- Remove dependency on jansson (locally supplied)
-------------------------------------------------------------------
Fri Mar  1 10:11:49 UTC 2019 - Adam Majer <adam.majer@suse.de>

- Update to new upstream release 16.2.1
  * Fix remote crash vulnerability in SDP protocol violation
    (CVE-2019-7251)

-------------------------------------------------------------------
Sat Feb 23 05:53:11 UTC 2019 - seanlew@opensuse.org

- Update to new upstream release 16.2.0

-------------------------------------------------------------------
Wed Dec 26 22:51:36 UTC 2018 - Michael Ströder <michael@stroeder.com>

- Update to new upstream release 16.1.1
  * Fix for Regression: MWI polling no longer works

-------------------------------------------------------------------
Wed Dec 12 22:59:05 UTC 2018 - Jan Engelhardt <jengelh@inai.de>

- Update to new upstream release 16.1.0
  * Fix a buffer overflow for DNS SRV/NAPTR records
  * Fix a crash when reading HTTP Upgrade request

-------------------------------------------------------------------
Thu Nov 22 19:47:07 UTC 2018 - Hans-Peter Jansen <hpj@urpla.net>

- Enable app_macro build
  deprecated now, but missing it will break too many diaplans

-------------------------------------------------------------------
Tue Nov 20 17:32:52 UTC 2018 - Jan Engelhardt <jengelh@inai.de>

- Update to new upstream release 16.0.1
  * webrtc: Both REMB and NACK are now supported.
  * Text messages sent through a conference bridge using
    ConfBridge will now be relayed to the other participants.
  * app_originate: The 'a' option has been added which
    asynchronously places calls. The application will return
    immediately instead of waiting for the originated channel
    to answer.
  * app_queue: A wrapup time can now be configured on a per-member
    basis instead of on a per-queue basis for static members as
    defined in the configuration file.
  * app_queue: Predial handler support has also been added so that
    subroutines can be invoked on the callee or caller channels.

-------------------------------------------------------------------
Thu Oct 25 10:45:46 UTC 2018 - Hans-Peter Jansen <hpj@urpla.net>

- Add missing /usr/share/asterisk/keys directory for res_crypto
- Adjust permissions of /var/lib/asterisk/phoneprov/*

-------------------------------------------------------------------
Wed Oct 24 17:01:01 UTC 2018 - Hans-Peter Jansen <hpj@urpla.net>

- Improve systemd unit to wait for network online state

-------------------------------------------------------------------
Sun Oct 21 10:56:40 UTC 2018 - Hans-Peter Jansen <hpj@urpla.net>

- Don't install /etc/init.d script, if systemd driven

-------------------------------------------------------------------
Mon Oct 15 15:19:39 UTC 2018 - Hans-Peter Jansen <hpj@urpla.net>

- Make adjusting asterisk.conf actually work, and prevent that perl
  expression from failing unnoticed ever again.

-------------------------------------------------------------------
Fri Sep 21 10:32:43 UTC 2018 - Michael Ströder <michael@stroeder.com>

- Update to new upstream release 15.6.1
  * AST-2018-009: Remote crash vulnerability in HTTP websocket upgrade

-------------------------------------------------------------------
Thu Jul 12 19:27:24 UTC 2018 - michael@stroeder.com

- Update to new upstream release 15.5.0 with following
  security fixes:
  * [ASTERISK-27818]
    Username bruteforce is possible when using ACL with PJSIP
  * [ASTERISK-27807]
    iostreams: Potential DoS when client connection closed prematurely

-------------------------------------------------------------------
Tue Jun 19 10:04:31 UTC 2018 - adam.majer@suse.de

- drop pwlib-devel from BR as it is not going to be ported to
  OpenSSL 1.1 (boo#1074796)

-------------------------------------------------------------------
Tue Jun 12 06:32:13 UTC 2018 - michael@stroeder.com

- Update to new upstream release 15.4.1 with following
  security fixes:
  * AST-2018-007: Infinite loop when reading iostreams
  * AST-2018-008: PJSIP endpoint presence disclosure when using ACL

-------------------------------------------------------------------
Sat May 26 14:32:42 UTC 2018 - dev@stellardeath.org

- Switch to bundled pjproject to avoid segmentation faults when
  using channel PJSIP

-------------------------------------------------------------------
Mon May 14 10:07:59 UTC 2018 - michael@stroeder.com

- Update to new upstream release 15.4.0 with following
  security fixes:
  * [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
  * [ASTERISK-27583] - Segmentation fault occurs in asterisk with an
    invalid SDP fmtp attribute
  * [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an
    invalid SDP media format description
  * [ASTERISK-27618] - Crash occurs when sending a repeated number of
    INVITE messages over TCP or TLS transport
  * [ASTERISK-27640] - SUBSCRIBE message with a large Accept value
    causes stack corruption

-------------------------------------------------------------------
Sun May  6 14:44:26 UTC 2018 - dev@stellardeath.org

- Remove sqlite2-devel as BuildRequires, is no longer available

-------------------------------------------------------------------
Sun Mar  4 14:25:11 UTC 2018 - jengelh@inai.de

- Update to new upstream release 15.2.2
  * The "Data Retrieval API" has been removed. This API was not
    actively maintained, was not added to new modules (such as
    res_pjsip), and there exist better alternatives to acquire
    the same information, such as the ARI. As a result, the
    "DataGet" AMI action as well as the "data get" CLI command
    have been removed.
  * Streams, as a new concept for media flows, have been
    introduced.
  * To simplify configuration for users a new option, webrtc, has
    been created which controls configuration options that are
    required for WebRTC.

-------------------------------------------------------------------
Wed Feb 21 23:40:10 UTC 2018 - michael@stroeder.com

- Update to new upstream release 14.7.6
 * AST-2018-001: Crash when receiving unnegotiated dynamic payload
 * AST-2018-002: Crash when given an invalid SDP media format description
 * AST-2018-003: Crash with an invalid SDP fmtp attribute
 * AST-2018-004: Crash when receiving SUBSCRIBE request
 * AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
 * AST-2018-006: WebSocket frames with 0 sized payload causes DoS

-------------------------------------------------------------------
Sat Dec 23 09:41:36 UTC 2017 - michael@stroeder.com

- Update to new upstream release 14.7.5
  * AST-2017-014: Crash in PJSIP resource when missing a contact header

-------------------------------------------------------------------
Thu Dec 14 22:06:50 UTC 2017 - michael@stroeder.com

- Update to new upstream release 14.7.4
  AST-2017-012: Remote Crash Vulnerability in RTCP Stack

-------------------------------------------------------------------
Fri Dec  1 21:40:51 UTC 2017 - michael@stroeder.com

- Update to new upstream release 14.7.3
  * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny

-------------------------------------------------------------------
Tue Sep 19 19:23:13 UTC 2017 - michael@stroeder.com

- Update to new upstream release 14.6.2 with security fix:
  AST-2017-008: RTP/RTCP information leak
- HTTPS source download links

-------------------------------------------------------------------
Wed Sep  6 13:06:14 UTC 2017 - michael@stroeder.com

- Update to new upstream release 14.6.1 with security fixes:
  * AST-2017-007: Remote Crash Vulerability in res_pjsip
  * AST-2017-006: Shell access command injection in app_minivm
  * AST-2017-005: Media takeover in RTP stack

-------------------------------------------------------------------
Wed Jul 12 14:07:26 UTC 2017 - michael@stroeder.com

- Update to new upstream release 14.6.0

-------------------------------------------------------------------
Tue May 30 18:44:30 UTC 2017 - michael@stroeder.com

- Update to new upstream release 14.5.0

-------------------------------------------------------------------
Mon May 22 19:58:24 UTC 2017 - hpj@urpla.net

- separate doc package (with ~26 MB)

-------------------------------------------------------------------
Sat May 20 07:58:46 UTC 2017 - michael@stroeder.com

- Update to new upstream release 14.4.1 with security fixes:
  * AST-2017-002: Ensure transaction key buffer is large enough
  * AST-2017-003: Handle zero-length body parts correctly
  * AST-2017-004: chan_skinny:  Add EOF check in skinny_session

-------------------------------------------------------------------
Fri Apr  7 17:22:26 UTC 2017 - michael@stroeder.com

- Update to new upstream release 14.4.0

-------------------------------------------------------------------
Tue Feb 21 22:13:30 UTC 2017 - jengelh@inai.de

- asterisk won't start without the sound directory
  (so make sure it is always there)

-------------------------------------------------------------------
Mon Feb 13 22:47:10 UTC 2017 - michael@stroeder.com

- Update to new upstream release 14.3.0

-------------------------------------------------------------------
Tue Jan 17 13:39:09 UTC 2017 - jengelh@inai.de

- Enable app_meetme

-------------------------------------------------------------------
Thu Dec  8 21:59:57 UTC 2016 - michael@stroeder.com

- Update to new upstream release 14.2.1 with security fixes:
  * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
  * AST-2016-009: Remote unauthenticated sessions in chan_sip

-------------------------------------------------------------------
Wed Nov 23 18:25:45 UTC 2016 - michael@stroeder.com

- Update to new upstream release 14.2.0

-------------------------------------------------------------------
Thu Nov 10 20:53:36 UTC 2016 - michael@stroeder.com

- Update to new upstream release 14.1.2

-------------------------------------------------------------------
Wed Oct 26 18:21:07 UTC 2016 - michael@stroeder.com

- Update to new upstream release 14.1.1

-------------------------------------------------------------------
Sat Oct  1 15:24:14 UTC 2016 - jengelh@inai.de

- Add asterisk-cflags.diff to drop -march=native again
  [boo#1002419]

-------------------------------------------------------------------
Sat Oct  1 00:14:18 UTC 2016 - michael@stroeder.com

- Update to new upstream release 14.0.2

-------------------------------------------------------------------
Tue Sep 27 19:56:20 UTC 2016 - michael@stroeder.com

- Update to new upstream release 14.0.1

-------------------------------------------------------------------
Mon Sep 26 20:01:56 UTC 2016 - jengelh@inai.de

- Update to new upstream release 14.0.0

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Fri May 13 19:58:36 UTC 2016 - michael@stroeder.com

- Update to new upstream maintenance release 13.9.1

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Wed Mar 30 06:30:17 UTC 2016 - michael@stroeder.com

- Update to new upstream maintenance release 13.9.0

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Sat Feb  6 22:11:13 UTC 2016 - jengelh@inai.de

- Update to new upstream maintenance release 13.7.2

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Fri Feb  5 18:33:22 UTC 2016 - michael@stroeder.com

- Update to new upstream maintenance release 13.7.1
  with security fixes:
  * AST-2016-001: BEAST vulnerability in HTTP server
  * AST-2016-002: File descriptor exhaustion in chan_sip
  * AST-2016-003: Remote crash vulnerability when receiving UDPTL FAX data.
- Added build dependencies:
  * libv4l-devel
  * libSDL2-devel

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Thu Dec 10 12:15:00 UTC 2015 - zawel1@gmail.com

- Update to new upstream maintenance release 13.6.0

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Thu Aug 13 15:16:00 UTC 2015 - jengelh@inai.de

- Update to new upstream maintenance release 13.5.0

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Mon Jun  8 22:30:50 UTC 2015 - jengelh@inai.de

- Update to new upstream maintenenace release 13.4.0

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Thu Apr  9 10:14:52 UTC 2015 - jengelh@inai.de

- Update to new upstream maintenance release 13.3.2
* fix for CVE-2015-3008 asterisk: TLS Certificate Common name NULL
  byte exploit

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Mon Mar 16 20:53:29 UTC 2015 - jengelh@inai.de

- Update to new upstream maintenance release 13.2

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Sat Jan  3 15:43:55 UTC 2015 - jengelh@inai.de

- Update to new upstream maintenance release 13.1

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Thu Nov 20 23:22:45 UTC 2014 - joop.boonen@opensuse.org

- Build version 13.0.1

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Thu Nov 20 21:13:00 UTC 2014 - joop.boonen@opensuse.org

- Corrected the file paths
- Added missing files
- Added excludes

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Mon Nov 17 20:51:45 UTC 2014 - jengelh@inai.de

- Update to new upstream release 13
* Asterisk security events are now provided via AMI, allowing end
  users to monitor their Asterisk system in real time for security
  related issues.
* Both AMI and ARI now allow external systems to control the state
  of a mailbox. Using AMI actions or ARI resources, external
  systems can programmatically trigger Message Waiting Indicators
  (MWI) on subscribed phones. This is of particular use to those
  who want to build their own VoiceMail application using ARI.
* ARI now supports the reception/transmission of out of call text
  messages using any supported channel driver/protocol stack
  through ARI. Users receive out of call text messages as JSON
  events over the ARI websocket connection, and can send out of
  call text messages using HTTP requests.
* The PJSIP stack now supports RFC 4662 Resource Lists, allowing
  Asterisk to act as a Resource List Server. This includes defining lists of presence state, mailbox state, or lists of presence state/mailbox state; managing subscriptions to lists; and batched delivery of NOTIFY requests to subscribers.
* The PJSIP stack can now be used as a means of distributing device
  state or mailbox state via PUBLISH requests to other Asterisk
  instances. This is analogous to Asterisk's clustering support
  using XMPP or Corosync; unlike existing clustering mechanisms,
  using the PJSIP stack to perform the distribution of state does
  not rely on another daemon or server to perform the work.

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Fri Aug 22 10:29:26 UTC 2014 - jengelh@inai.de

- Update to new upstream maintenance release 12.5.0
* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.5.0-summary.txt

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Sun Jul 13 00:07:28 UTC 2014 - jengelh@inai.de

- Update to new upstream release 12.4.0
* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.4.0-summary.txt
- Reenable SS7 support in chan_dahdi (for libss7-2.0)

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Fri Jun 27 23:49:07 UTC 2014 - jengelh@inai.de

- Update to new upstream release 12.3.2
* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.3.2-summary.txt

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Wed Apr 23 17:16:00 UTC 2014 - marcelloceschia@users.sourceforge.net

- Update to new upstream release 12.2.0
* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.2.0-summary.txt

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Thu Mar 22 22:32:00 UTC 2014 - marcelloceschia@users.sourceforge.net

- Update to new upstream release 12.1.1 (security release)
* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.1.1-summary.txt

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Thu Mar  6 07:34:00 UTC 2014 - jengelh@inai.de

- Update to new upstream release 12.1.0 (bugfix release)
* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.1.0-summary.txt

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Tue Dec 24 10:33:13 UTC 2013 - jengelh@inai.de

- Update to new upstream release 12.0.0
* A more flexible bridging core based on the Bridging API
* A new internal message bus, Stasis
* Major standardization and consistency improvements to AMI
* Addition of the Asterisk REST Interface (ARI)
* A new SIP channel driver, chan_pjsip
* https://wiki.asterisk.org/wiki/display/AST/New+in+12

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Tue Dec 24 10:29:52 UTC 2013 - jengelh@inai.de

- Update to new upstream release 11.7.0 (bugfix release)
* See http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.7.0-summary.html
  for details

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Sun Nov 24 11:57:30 UTC 2013 - jengelh@inai.de

- Update to new upstream release 11.6.0 (bugfix release)
* See http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.6.0-summary.html
  for details

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Thu Aug 15 14:56:19 UTC 2013 - jengelh@inai.de

- Use libuuid to reenable res_rtp_asterisk

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Thu Aug  8 11:32:21 UTC 2013 - jengelh@inai.de

- Update to new upstream release 11.5.0 (bugfix release)
* See http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.5.0-summary.html
  for details

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Sun Jun  2 23:05:18 UTC 2013 - jengelh@inai.de

- Update to new upstream release 11.4.0 (bugfix release)
* See http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.4.0-summary.html
  for details

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Sun Feb 17 09:27:46 UTC 2013 - jengelh@inai.de

- Enable building res_corosync (replaces res_ais from asterisk-10)
- Order asterisk after network (bnc#796148)

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Sat Feb 16 02:32:13 UTC 2013 - jengelh@inai.de

- Enable building chan_ooh323
- Put config sample files into their respective subpackages
- Split off asterisk-freetds
- Make libasteriskssl.so symlink point to actual file
- Call ldconfig for libasteriskssl1

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Thu Jan 24 18:02:07 UTC 2013 - jengelh@inai.de

- Update to new upstream release 11.2.1 (bugfix release)
* Fixed stuck DTMF when using ChannelRedirect to split a two
  channel bridge
* Asterisk deadlocked during startup with mutex errors
* Resolved segfault in chan_sip while performing connected line
  update

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Fri Dec 21 22:50:50 UTC 2012 - joop.boonen@opensuse.org

- Update to new upstream release 11.1.0
* chan_local: Fix local_pvt ref leak in local_devicestate().
* Fix a SIP request memory leak with TLS connections.
* Fix a bug where our Motif ICE candidates were not quite proper,
  and make us more forgiving.

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Wed Dec  5 11:07:31 UTC 2012 - joop.boonen@opensuse.org

- Update to new upstream release 11.0.1
* Fix a bug which made ConfBridge not record conferences when the
  record command was initiated from AMI/CLI commands
* Fix a bug causing SIP reloads to remove all entries from the registry
* Fix an issue with res_http_websocket where the chan_sip WebSocket
  handler could not be registered.

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Sat Nov  3 02:33:18 UTC 2012 - jengelh@inai.de

- Update to new upstream release 11.0.0
* WebRTC Support with WebSocket transport over SIP.
* DTLS-SRTP - A secure transport for RTP media streams used by
  WebRTC and SIP endpoints.
* ICE, STUN and TURN – A set of related technologies for
  establishing live media streams between software agents running
  behind network address translators (NATs) and firewalls. ICE,
  STUN and TURN have been incorporated into the Asterisk RTP engine.

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Sun Apr  8 18:44:34 UTC 2012 - jengelh@medozas.de

- Update to new upstream release 10.3.0
* http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-10.3.0-summary.html
- Make /var/lib/asterisk writable, so that the sqlite db can
  be automatically created
- Replace init script by something less convoluted;
  also add a systemd service file (bnc#750762, bnc#750763)

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Fri Mar 16 19:28:25 UTC 2012 - jengelh@medozas.de

- Update to new upstream release 10.2.1
* Fix AST-2012-002, AST-2012-003 security vulnerabilities

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Sun Mar 11 21:54:37 UTC 2012 - jengelh@medozas.de

- Update to new upstream release 10.2.0
* http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-10.2.0-summary.html
- Restore spandsp support (bnc#731943)
- Set permissions on files (bnc#750761)

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Wed Feb  1 15:10:07 UTC 2012 - jengelh@medozas.de

- Update to new upstream release 10.1.0
* http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-10.1.0-summary.html
- Add autotools BuildRequires for factory/12.2

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Fri Dec 16 00:06:31 UTC 2011 - jengelh@medozas.de

- Update to final 10.0.0

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Sat Oct  8 22:12:16 UTC 2011 - jengelh@medozas.de

- New package, for a change list see
  https://wiki.asterisk.org/wiki/display/AST/New+in+10

Places

File asterisk.changes of Package asterisk (Revision 6c7f02105c0e785c50507a0a0a247aa9)

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