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-------------------------------------------------------------------
Mon Apr 17 10:51:03 UTC 2023 - Jan Engelhardt <jengelh@inai.de>
- Enable chan_mobile
-------------------------------------------------------------------
Thu May 19 17:16:31 UTC 2022 - Michael Ströder <michael@stroeder.com>
- Updated pjproject to 2.12
- Update to release 18.12.1
* Release 18.12.1:
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing requests
* Release 18.12.0:
+ Security fixes
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
+ New Features
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter but everyone else can
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at prefix
- [ASTERISK-29486] - Hint-like extension value lookup function without device state
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events
- [ASTERISK-29820] - cli: Add command to evaluate a function
- [ASTERISK-29876] - app_queue: Add music on hold option
+ Bugs fixes
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera available
- [ASTERISK-29638] - res_pjsip_session: No video after early media
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up Dahdi Call On Hold
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi restart
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with missing secrets
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled are always recompiled
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when channel is NULL
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max number of groups
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R" flags. (documentation bug)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable console colorization"
- [ASTERISK-29843] - Session timers get removed on UPDATE
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if early_media already enabled
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data
- [ASTERISK-29253] - Incorrect bridging on transfer
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when async_operations is greater than 1
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not enabled
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new keyword
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database columns
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP attributes
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting channel for lack of RTP activity
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP activity in one way sessions
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context (AST_PBX_MAX_STACK - 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
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats printout
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments displayed incorreclty
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
- [ASTERISK-29674] - Adjust for 64bit time_t
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with SUBSCRIBE request
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
+ Improvements
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring cadences
- [ASTERISK-29940] - general: Add since tags to xmldocs
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP) implementation
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk
- [ASTERISK-29976] - Should Readme include information about install_prereq script?
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
- [ASTERISK-29980] - build: External binary modules don't use https
- [ASTERISK-25716] - Documentation: Document explanations and examples for possible values of DIALSTATUS
- [ASTERISK-29967] - pbx_builtins: Add missing documentation
-------------------------------------------------------------------
Tue Apr 26 17:19:07 UTC 2022 - Michael Ströder <michael@stroeder.com>
- Update to release 18.11.3
* [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not enabled
-------------------------------------------------------------------
Fri Apr 15 08:29:49 UTC 2022 - Michael Ströder <michael@stroeder.com>
- Update to release 18.11.2 with security fixes for
* AST-2022-001: res_stir_shaken: resource exhaustion with large files
* AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header
* AST-2022-003: func_odbc: Possible SQL Injection
- remove unpackaged file
-------------------------------------------------------------------
Wed Mar 30 08:16:33 UTC 2022 - Michael Ströder <michael@stroeder.com>
- 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
-------------------------------------------------------------------
Thu Mar 24 14:09:38 UTC 2022 - Michael Ströder <michael@stroeder.com>
- Updated to jansson-2.14
- Update to release 18.11.0:
* Security bugs fixed:
- [ASTERISK-29945] - pjproject: Security fixes for things
* New Features:
- [ASTERISK-29853] - ami: Allow events to be globally disabled
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
* Bugs fixed:
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'" error
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists
- [ASTERISK-29904] - RLS: Batched Notifications stop working
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown
- [ASTERISK-29873] - [patch] Queue Realtime load
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a call event.
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML
* Improvements:
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
- [ASTERISK-29896] - xmldocs: Add since tag
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
- [ASTERISK-29925] - func_db: Warn about malformed key names
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
- [ASTERISK-29866] - cli: add core dump information to core show settings
- [ASTERISK-29898] - documentation: Add default attributes to documentation
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits
-------------------------------------------------------------------
Sat Mar 5 10:04:14 UTC 2022 - Michael Ströder <michael@stroeder.com>
- 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
-------------------------------------------------------------------
Sun Feb 6 13:36:20 UTC 2022 - Martin Hauke <mardnh@gmx.de>
- Reenable build with support for DAHDI on supported platforms
-------------------------------------------------------------------
Sat Jan 8 11:53:07 UTC 2022 - Michael Ströder <michael@stroeder.com>
- 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
-------------------------------------------------------------------
Thu Sep 9 21:40:15 UTC 2021 - Jan Engelhardt <jengelh@inai.de>
- Update to release 18.6.0
* AST-2021-009 - pjproject-bundled: Avoid crash during
handshake for TLS
* app_reload: New Reload application
* app_waitforcond: New application
* app_dtmfstore: New application to store digits
* AST-2021-008 - chan_iax2: remote crash on unsupported
media format
-------------------------------------------------------------------
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
-------------------------------------------------------------------
Fri May 13 19:58:36 UTC 2016 - michael@stroeder.com
- Update to new upstream maintenance release 13.9.1
-------------------------------------------------------------------
Wed Mar 30 06:30:17 UTC 2016 - michael@stroeder.com
- Update to new upstream maintenance release 13.9.0
-------------------------------------------------------------------
Sat Feb 6 22:11:13 UTC 2016 - jengelh@inai.de
- Update to new upstream maintenance release 13.7.2
-------------------------------------------------------------------
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
-------------------------------------------------------------------
Thu Dec 10 12:15:00 UTC 2015 - zawel1@gmail.com
- Update to new upstream maintenance release 13.6.0
-------------------------------------------------------------------
Thu Aug 13 15:16:00 UTC 2015 - jengelh@inai.de
- Update to new upstream maintenance release 13.5.0
-------------------------------------------------------------------
Mon Jun 8 22:30:50 UTC 2015 - jengelh@inai.de
- Update to new upstream maintenenace release 13.4.0
-------------------------------------------------------------------
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
-------------------------------------------------------------------
Mon Mar 16 20:53:29 UTC 2015 - jengelh@inai.de
- Update to new upstream maintenance release 13.2
-------------------------------------------------------------------
Sat Jan 3 15:43:55 UTC 2015 - jengelh@inai.de
- Update to new upstream maintenance release 13.1
-------------------------------------------------------------------
Thu Nov 20 23:22:45 UTC 2014 - joop.boonen@opensuse.org
- Build version 13.0.1
-------------------------------------------------------------------
Thu Nov 20 21:13:00 UTC 2014 - joop.boonen@opensuse.org
- Corrected the file paths
- Added missing files
- Added excludes
-------------------------------------------------------------------
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.
-------------------------------------------------------------------
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
-------------------------------------------------------------------
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)
-------------------------------------------------------------------
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
-------------------------------------------------------------------
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
-------------------------------------------------------------------
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
-------------------------------------------------------------------
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
-------------------------------------------------------------------
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
-------------------------------------------------------------------
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
-------------------------------------------------------------------
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
-------------------------------------------------------------------
Thu Aug 15 14:56:19 UTC 2013 - jengelh@inai.de
- Use libuuid to reenable res_rtp_asterisk
-------------------------------------------------------------------
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
-------------------------------------------------------------------
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
-------------------------------------------------------------------
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)
-------------------------------------------------------------------
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
-------------------------------------------------------------------
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
-------------------------------------------------------------------
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.
-------------------------------------------------------------------
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.
-------------------------------------------------------------------
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.
-------------------------------------------------------------------
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)
-------------------------------------------------------------------
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
-------------------------------------------------------------------
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)
-------------------------------------------------------------------
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
-------------------------------------------------------------------
Fri Dec 16 00:06:31 UTC 2011 - jengelh@medozas.de
- Update to final 10.0.0
-------------------------------------------------------------------
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
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