Beta-Version mit Asterisk 1.4 verfügbar
Digium stellt mit Asterisk Now nun ein fertiges Paket, bestehend aus einer Linux-Distribution und der freien Telefonanlage Asterisk, zum Download bereit. Damit soll sich die Software in etwa 30 Minuten komplett konfigurieren lassen, ohne dass Linux-Kenntnisse vonnöten sind.
Die verfügbare Beta-Version von Asterisk Now enthält bereits das kürzlich veröffentlichte Asterisk 1.4, das unter anderem mit Funktionen wie einer Jabber-Integration aufwartet. Um Asterisk zu nutzen, besteht Asterisk Now aus einer kompletten Linux-Distribution, zu der Digium allerdings keine weiteren Angaben macht. Das Betriebssystem soll hier mehr Mittel zum Zweck sein, mit dem sich der Anwender nicht weiter auseinandersetzen braucht.
Stattdessen ist ein Einrichtungswerkzeug enthalten, das den Nutzer durch das Setup leitet. Auch ohne Linux-Kenntnisse soll sich eine Asterisk-Telefonanlage somit in 30 Minuten aufsetzen lassen, verspricht Digium. Die eingesetzte Linux-Variante soll nur die notwendigen Komponenten mitbringen, um so Sicherheits- und Leistungsprobleme zu umgehen. Das Ziel von Asterisk Now sei es, die Komplexität bei der Installation und Nutzung von Asterisk zu verringern und die Lösung damit weiter zu verbreiten, so Asterisk-Erfinder Mark Spencer.
Asterisk Now steht in der zweiten Beta-Version als ISO-Image für x86 und x86-64, als Live-CD und als Image für die Virtualisierungslösungen Xen und VMware kostenlos zum Download bereit. Digium verlangt allerdings eine Registrierung.
http://www.asterisknow.org/downloads
Quelle : www.golem.de
Linux-Distribution mit VoIP-Telefonanlage
Die Linux-Distribution Trixbox ist nun in der Version 2.2 verfügbar, die einige Teile der Version 2.0 verbessern soll. Unter anderem sollen sich spätere Updates leichter durchführen lassen. Die auf CentOS aufsetzende Distribution ist zum schnellen und unkomplizierten Einsatz einer Voice-over-IP-Telefonanlage mittels Asterisk gedacht.
Trixbox ist eine angepasste Version der auf den Red-Hat-Enterprise-Linux-Quellen basierenden Linux-Distribution CentOS und richtet sich speziell an Anwender, die unkompliziert eine VoIP-Telefonanlage einrichten möchten. Die Aufgabe der Telefonanlage übernimmt die Open-Source-Software Asterisk, für die Trixbox Verwaltungssoftware enthält, so dass Konfigurationsdateien nicht mehr manuell geändert werden müssen.
Die neue Trixbox-Version soll die letzte Asterisk-Ausgabe enthalten, womit Version 1.4.4 gemeint sein dürfte. Die Versionsnummer wird jedoch nicht explizit aufgelistet. Zusätzlich wurden die Berichte mit den Gesprächsdaten erweitert und es gibt einen neuen Manager, um die Netzwerkschnittstellen zu verwalten. Auch die Systemstatus-Anzeige überarbeiteten die Entwickler. Weiterhin gab es Verbesserungen bei der Hardware-Unterstützung.
Trixbox 2.2 steht als ISO-Image zum Download bereit. Hinter der ehemals als Asterisk@Home bekannten Distribution steht mittlerweile die Firma Fonality, die ebenfalls eine IP-Telefonanlage auf Basis von Asterisk entwickelt.
http://www.trixbox.org/downloads
Quelle : www.golem.de
Die unter der GPL-Lizenz erschienene Telefonsoftware Gemeinschaft ist in einer Home-Edition in der Version 2.3 erschienen. Gemeinschaft basiert auf Asterisk und Lamp. Die Home-Edition ist für den Privatgebrauch und die teilgewerbliche Nutzung kostenlos.
Die aktuelle Version 2.3 der auf Asterisk und Lamp basierenden Telefonsoftware Gemeinschaft bringt eine vereinfachte Benutzeroberfläche für Privatanwender mit, die die Konfiguration erleichtern soll. Gemeinschaft 2.3 unterstützt weitere IP-Telefone, etwa der Firma Aastra, Grandstream, Siemens oder Snom. Mit der Home-Edition können bis zu 20 Anschlüsse verwaltet werden. Die Software verwendet Sprachbausteine im GSM-Format. Gemeinschaft steht unter amooma.de zum Download bereit.
(http://scr3.golem.de/screenshots/0911/Gemeinschaft/thumb480/add-user.png)
Im Gegensatz zur Home-Edition enthält die kostenpflichtige Business-Edition zusätzliche Voiceprompts und bringt kostenlosen Support mit. Außerdem verwaltet die Business-Edition bis zu 1.000 Anschlüsse. Für soziale Einrichtungen bietet die Firma Amooma die Business-Edition kostenlos an, allerdings ohne Support. Amooma hat noch weitere Pakete mit der Telefonsoftware im Portfolio: eine Call-Center-Edition und eine Cluster-Edition.
Quelle : www.golem.de
Changelog
The release of Asterisk 13.5.0 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
Bug
[ASTERISK-19277] - [patch]endlessly repeating error: "poll failed: Bad file descriptor"
[ASTERISK-22559] - gcc 4.6 and higher supports weakref attribute but asterisk doesn't detect it.
[ASTERISK-22805] - res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
[ASTERISK-24344] - CDR_PROP(disable) disables CDR only for first dialed party
[ASTERISK-24443] - CDR fields (dst, dcontext) empty in transfer call started from Macro
[ASTERISK-24550] - res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake
[ASTERISK-24651] - [patch] Fix race condition in DTLS
[ASTERISK-24717] - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10}
[ASTERISK-24782] - StasisEnd event not present for channel that was swapped out for another after completing attended transfer
[ASTERISK-24832] - [patch]DTLS-crashes within openssl
[ASTERISK-24853] - Documentation claims chan_sip outbound registrations support WS or WSS as valid transports (not true)
[ASTERISK-24867] - Docs for 'e' option in ResetCDR say to use CDR_PROP instead, CDR_PROP docs are unclear
[ASTERISK-24900] - Manager event ParkedCallSwap is not documented
[ASTERISK-24907] - res_pjsip_outbound_registration: crash during unload if registration attempts are still occuring
[ASTERISK-24934] - [patch]Asterisk manager output does not escape control characters
[ASTERISK-24963] - ASAN: heap-use-after-free with PJSIP and WSS
[ASTERISK-24983] - IAX deadlock between hangup and scheduled actions (ex. largrq)
[ASTERISK-24988] - func_talkdetect: Test is bouncing sporadically
[ASTERISK-25087] - Asterisk segfault when using Directory application with alias option and specific mailbox configuration
[ASTERISK-25091] - Asterisk REST API - bridge.addChannel crash asterisk when calling channel hangup while adding to bridge
[ASTERISK-25094] - PBX core: Investigate thread safety issues
[ASTERISK-25096] - [patch]Segfault when registering over websockets with PJSIP (in ast_sockaddr_isnull at /include/asterisk/netsock2.h)
[ASTERISK-25100] - asterisk coredump if host has an IPv6 address that end with ::80
[ASTERISK-25103] - Roundup - investigate Asterisk DTLS crashes
[ASTERISK-25105] - res_pjsip: Possible incompatibility between qualify_timeout and pjproject-2.4
[ASTERISK-25115] - Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c
[ASTERISK-25116] - res_pjsip: Two PeerStatus AMI messages are sent for every status change
[ASTERISK-25117] - res_mwi_external_ami: Fix manager action registrations.
[ASTERISK-25121] - Stasis: Fix unsafe use of stasis_unsubscribe in modules.
[ASTERISK-25122] - Large SIP packet received via pjsip over websocket crashes Asterisk
[ASTERISK-25127] - DTLS crashes following "Unable to cancel schedule ID" in dtls_srtp_check_pending
[ASTERISK-25131] - chan_pjsip: In-dialog authentication not handled.
[ASTERISK-25137] - endpoint stasis messages are delivered twice
[ASTERISK-25148] - res_pjsip NULL channel audit
[ASTERISK-25154] - [patch]fromtag may need to be updated after successful call dialog match
[ASTERISK-25156] - chan_pjsip’s CHAN_START cel event lacks the correct context and exten
[ASTERISK-25157] - bridging: Performing a blonde transfer does not result in connected line updates
[ASTERISK-25158] - res_pjsip: Add option to use AAL2 packing when negotiating g.726
[ASTERISK-25162] - func_pjsip_aor: Leak of contact in iterator
[ASTERISK-25163] - Deadlock in chan_sip between reload of sip peer container and MWI Stasis callback
[ASTERISK-25165] - Testsuite - Sorcery memory cache leaks
[ASTERISK-25168] - Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c
[ASTERISK-25171] - Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound.
[ASTERISK-25172] - Crash in channels/sip/sip blind transfer/caller_refer_only test in ast_format_cap_append_from_cap during ast_request
[ASTERISK-25180] - res_pjsip_mwi: Unsolicited MWI requires reload
[ASTERISK-25182] - [patch] on CLI sip reload, new codecs get appended only
[ASTERISK-25183] - PJSIP: Crash on NULL channel in chan_pjsip_incoming_response despite previous checks for NULL channel
[ASTERISK-25189] - AMI: Add Linkedid header to standard channel snapshot information.
[ASTERISK-25196] - res_pjsip_nat: rewrite_contact should not be applied to Contact header when Record-Route headers are present
[ASTERISK-25201] - Crash in PJSIP distributor on already free'd threadpool
[ASTERISK-25202] - Hints extension state broken between 13.3.2 and 13.4
[ASTERISK-25204] - res_pjsip_refer: Duplicated Referred-By or Replaces headers on outbound INVITEs.
[ASTERISK-25212] - [patch]Segfault when using DEBUG_FD_LEAKS
[ASTERISK-25219] - [patch]Source and destination overlap in memcpy in rtp_engine.c
[ASTERISK-25220] - [patch]Closing of fd -1 in chan_mgcp.c
[ASTERISK-25226] - chan_sip: Channel leak in branch 13 on early replaces call pickup
[ASTERISK-25240] - bridge_native_rtp: Direct media wrongfully started when completing attended transfer
[ASTERISK-25242] - PJSIP: No audio when Asterisk inside NAT and endpoints outside NAT - implement functionality similar to chan_sip 'rtpkeepalive'?
[ASTERISK-25247] - choppy audio when spying on a g722 channel, chan_sip or chan_pjsip
[ASTERISK-25250] - chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback
[ASTERISK-25253] - confbridge volume options and other volume controls such as func_volume don't work
[ASTERISK-25254] - Crash if dialplan sets ATTENDEDTRANSFER to an empty string before Park.
[ASTERISK-25255] - Missing AMI VarSet events when setting to an empty string.
[ASTERISK-25257] - [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope
[ASTERISK-25258] - chan_pjsip: Incorrect format switch on received RTP packet
Improvement
[ASTERISK-25040] - pbx: Improve performance of reloads by making hint destruction more performant
[ASTERISK-25067] - Sorcery Caching: Implement a new caching module
[ASTERISK-25072] - res_pjsip_outbound_registration: line functionality. Additional check for using the request URI
[ASTERISK-25114] - res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes
[ASTERISK-25256] - [patch]Post AMI VarSet to empty string events when Asterisk deletes a dialplan variable.
New Feature
[ASTERISK-25173] - ARI: Add the ability to load/reload/unload an Asterisk module
[ASTERISK-25238] - ARI: Support push configuration
[ASTERISK-25259] - chan_pjsip: Add rtptimeout support
http://www.asterisk.org/
Changelog
Improvements made in this release:
ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue
ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample
ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
Bugs fixed in this release:
ASTERISK-26608 - Compile and link failures on OpenBSD
ASTERISK-26520 - codec_opus: Generated fmtp line has no content
ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
ASTERISK-26516 - pjsip: Memory corruption with possible memory leak.
ASTERISK-26556 - manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes
ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi
ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
ASTERISK-26571 - res_pjsip: Resolution incorrect when explicit IPv6 transport configuredASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls
ASTERISK-24400 - ooh323 sends wrong hangup code
ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions
ASTERISK-26412 - build: Prepare for gcc 6.2
ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10
ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state
ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types.
ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
ASTERISK-25070 - Fix FTBFS on Hurd
ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum
ASTERISK-26537 - AMI: NewConnectedLine event is not documented
ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash
ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls.
ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf
ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak
ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change
ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
ASTERISK-26482 - [patch] chan_pjsip: segfault on already disconnected session
ASTERISK-26421 - Segmentation Fault with ARI originate into mixing bridge with 43 clients
ASTERISK-26444 - 'features show' command in CLI does not return prompt.
ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
ASTERISK-26356 - menuselect: invalid test for GTK2
ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
ASTERISK-26439 - chan_rtp: Crash when originating
ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered.
ASTERISK-26618 - build: Backport addition of librt check to configure.ac
New Features made in this release:
ASTERISK-26595 - ARI: Add the ability to control the source of video in a multi-party mixing bridge
ASTERISK-26492 - ARI: Add ability to specify channel variables on websocket events
ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing events
http://www.asterisk.org/
Changelog
Security bugs fixed in this release:
ASTERISK-27818 - Username bruteforce is possible when using ACL with PJSIP
ASTERISK-27807 - iostreams: Potential DoS when client connection closed prematurely
Bugs fixed in this release:
ASTERISK-27783 - res_pjsip_pubsub: apparent crash on shutdown
ASTERISK-27870 - app_confbridge: Conference bridge and announcer channels are not removed if conference is ended as soon as it starts
ASTERISK-27943 - AMI: Action SendText needs to use the correct thread.
ASTERISK-27942 - res_pjsip_messaging doesn't accept application/* content-types.
ASTERISK-27909 - cdr: Deadlock with submit_scheduled_batch and submit_unscheduled_batch
ASTERISK-27936 - res_pjsip_session doesn't update media when a 200 comes in with a different port than a 183
ASTERISK-26987 - pbx_dundi: Asterisk crashes when unloading module pbx_dundi.so with dundi peers
ASTERISK-27933 - [patch] uuid: Enable UUID in Solaris 11.
ASTERISK-27625 - channels: CHECK_BLOCKING is ineffective
ASTERISK-27931 - [patch] BuildSystem: Enable ./configure in Solaris 11.
ASTERISK-27926 - [patch] bootstrap.sh: find -maxdepth is not POSIX compatible.
ASTERISK-27903 - menuselect: GCC 8: restrict-qualified parameter passed and aliased.
ASTERISK-27914 - [patch] tests/test_utils: Repair ./configure --with-ssl=PATH.
ASTERISK-27705 - chan_iax2: Stops listening for traffic
ASTERISK-27908 - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
ASTERISK-27905 - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
ASTERISK-27888 - SQL fetch error on query which return 0 columns
ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX responses
ASTERISK-27901 - [patch] ooh323c: GCC 8: output truncated before terminating nul.
ASTERISK-27872 - res_pjsip: Modified qualify_frequency doesn't effect until pjsip reload
ASTERISK-27094 - res_fax: Deadlock when using Local channels and fax gateway
ASTERISK-27848 - rtp: DTMF Breaks With telephony-event/16000
ASTERISK-25261 - Manager events for MeetMe have incorrectly documented key name 'Usernum' - should be 'User'
ASTERISK-27878 - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
ASTERISK-27876 - [patch] tcptls: Allow OpenSSL configured with no-dh.
ASTERISK-27874 - [patch] tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated.
ASTERISK-27845 - Codec-Change Re-INVITE during DTMF can cause marker bit error
ASTERISK-27831 - res_rtp_asterisk: Add support for abs-send-time RTP extension
ASTERISK-27863 - config/ast_destroy_realtime_fields: successful DELETE is treated as failed
ASTERISK-27865 - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
ASTERISK-27760 - Asterisk ODBC Voicemail Prompt storage fails with recent MariaDB version.
ASTERISK-27853 - Incorrect error reported when leaving/retrieving a ODBC voicemail
ASTERISK-27726 - chan_mobile: presents incorrect inbound Caller-ID names
ASTERISK-27861 - [patch] res_pjsip_endpoint_identifier_ip: Unregister the module for headers.
ASTERISK-27860 - [patch] res_pjsip: Register pjsip_transport_management not externally but internally.
ASTERISK-27852 - cli: "manager show settings" mislabels HTTP timeout as being minutes.
ASTERISK-27824 - Fix issues exposed by GCC 8
ASTERISK-27850 - [patch] rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again.
ASTERISK-27811 - [patch] sip_to_pjsip: Enable python3 compatibility.
ASTERISK-27841 - digest over for manager (ami) over http fails on too long uris
ASTERISK-26570 - Macro allows an infinite loop of dialplan inclusion resulting in a crash
ASTERISK-27801 - Asterisk got stuck while enabling "ari set debug all on"
ASTERISK-27795 - chan_sip: one way / no audio with srtp
ASTERISK-27800 - One way audio when calling from Asterisk(sip trunk) to another number where both are connected to a SBC using TLS+SRTP
ASTERISK-26806 - pjsip_options: rework to make more efficient
ASTERISK-27814 - translate: interpolated frames are not passed through
ASTERISK-27812 - When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk.
ASTERISK-26893 - No "alert" or "progress" in chan_ooh323 if debug is enabled only on the module
ASTERISK-27639 - [patch] BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD.
ASTERISK-27804 - bridge_softmix / app_confbridge: Add support for combining REMB reports
ASTERISK-27418 - app_confbridge: "core show profile bridge" does not output "sfu" when video_mode is sfu
ASTERISK-27808 - [patch] chan_vpb: Avoid GNU old-style field designator extension.
Improvements made in this release:
ASTERISK-27929 - [patch] BuildSystem: Enable autotools in Solaris 11.
ASTERISK-27752 - Ten seconds of silence after mp3 playback
ASTERISK-27910 - [patch] res_rtp_asterisk: Allow OpenSSL configured with no-deprecated.
ASTERISK-27906 - [patch] res_crypto: Allow OpenSSL configured with no-deprecated.
ASTERISK-27877 - app_confbridge: Add talking indicator for ConfBridgeList AMI response
ASTERISK-27873 - documentation: Error on wiki description of Asterisk 13 "MeetmeMute" event
ASTERISK-27846 - ast_coredumper: Fix OUTPUT directory
ASTERISK-27867 - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated.
ASTERISK-27796 - res_hep: Allow create_address to resolve a provided hostname
ASTERISK-27820 - [patch] Add DragonFly BSD.
ASTERISK-27793 - cppcheck identifies redundant "if"
http://www.asterisk.org/
Release Notes
The Asterisk Development Team would like to announce the release of Asterisk 16.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.0.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! For brevity I am omitting the list of issues. For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
www.asterisk.org/
Changelog
New Features:
ASTERISK-28375 - res_pjsip: New configuration setting to allow disabling norefersub
ASTERISK-28320 - Added ARI resource /ari/channels/{channelid}/rtp_statistics
Bugs fixed:
ASTERISK-28427 - new mwi.h include missing from some dahdi source files, causes build failure
ASTERISK-28412 - GCC 9 catches more string formatting issues
ASTERISK-28379 - pjsip: show channelstats incorrect information output
ASTERISK-28399 - channel.c: Exceptionally long queue length queuing
ASTERISK-28392 - The no-partial-inlining flag isn't passed to the bundled pjproject or jansson builds
ASTERISK-28402 - res_pjsip_registrar: SEGV in registrar_find_contact
ASTERISK-27756 - bridge: Failure to impart a channel results in bad data causing crash
ASTERISK-26718 - ARI: Bridge destroying doesn't work as expected
ASTERISK-28143 - app_amd: Infinite loop on silent calls
ASTERISK-28353 - stasis: Crash at shutdown when statistics enabled
ASTERISK-28374 - latest asterisk unconditionally launch gcc --version, even if the compiler is different
ASTERISK-28391 - res_indications: Crash requesting autocomplete on indications cli command
ASTERISK-27935 - app_voicemail: emailbody per user can't contain commas
ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with '-' in them
ASTERISK-17799 - AEL reload causes loss of control in a macro
ASTERISK-18593 - AEL for loops use Macro app and pipe delimiter
ASTERISK-14939 - AEL parsers does not find existing label
ASTERISK-20182 - Parsing a label beginning with a numeric character in all Goto/GotoIf/GotoIfTime application causes unexpected behavior
ASTERISK-28348 - Failed to initialize OOH323 endpoint-OOH323 Disabled
ASTERISK-28371 - chan_pjsip: DTMF Mode auto_info fallback lead to both inband and info
ASTERISK-28319 - musl: Crash on startup when loading modules
ASTERISK-28362 - strtok_r() makes gcc compile warning
ASTERISK-28255 - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
Improvements:
ASTERISK-28401 - app_confbridge: Add *_all remb behavior variants
ASTERISK-28400 - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for transport-cc
ASTERISK-28363 - Millisecond-resolution call stats including PDD in channel variables
ASTERISK-20207 - Asterisk should clear out any .lock files in the voice mail directory on startup.
ASTERISK-28111 - build: CHANGES/UPGRADE are irritating to work with.
ASTERISK-28343 - Added app_name, app_data to channel type
ASTERISK-28264 - Added topic_all container
www.asterisk.org
Changelog
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-28574
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. ASTERISK-28575
www.asterisk.org