Asterisk News

Asterisk Releases

Asterisk 13.11.0-rc1 Now Available

Jul 28, 2016

The Asterisk Development Team has announced the first release candidate of Asterisk 13.11.0.

This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.11.0-rc1 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 candidate:

Bug

  • [ASTERISK-19968] - TCP Session-Timers not dropping call
  • [ASTERISK-23013] - [patch] Deadlock between 'sip show channels' command and attended transfer handling
  • [ASTERISK-25289] - Build System does not respect CFLAGS and CXXFLAGS when building menuselect
  • [ASTERISK-25659] - res_rtp_asterisk: ECDH not negotiated causing DTLS failure occurred on RTP instance
  • [ASTERISK-25772] - res_pjsip: Unexpected two BYE when answered
  • [ASTERISK-26021] - Build codecs siren7 and siren14 for Asterisk 13
  • [ASTERISK-26038] - 'make install' doesn't seem to install OS/X init files
  • [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf.
  • [ASTERISK-26099] - res_pjsip_pubsub: Crash when sending request due to server timeout
  • [ASTERISK-26119] - [patch] fix: memory leaks, resource leaks, out of bounds and bugs
  • [ASTERISK-26133] - app_queue: Queue members receive multiple calls
  • [ASTERISK-26144] - Crash on loading codecs g729/g723
  • [ASTERISK-26157] - Build: Fix errors highlighted by GCC 6.x
  • [ASTERISK-26160] - pjsip: Updated->Reachable during qualify
  • [ASTERISK-26177] - func_odbc: Database handle is kept when it should be released
  • [ASTERISK-26179] - chan_sip: Second T.38 request fails
  • [ASTERISK-26180] - PJSIP: provide valid tcp nodelay option for reuse
  • [ASTERISK-26181] - REF_DEBUG: Node object incorrectly logged during duplicate replacement
  • [ASTERISK-26184] - chan_sip: Reference leaks in error paths.
  • [ASTERISK-26191] - threadpool: Leak on duplicate taskprocessor for ast_threadpool_serializer_group
  • [ASTERISK-26193] - chan_sip: reference leak in mwi_event_cb
  • [ASTERISK-26196] - pbx: Time based includes can leak timezone string
  • [ASTERISK-26207] - [patch] sRTP: Count a roll-over of the sequence number even on lost packets.
  • [ASTERISK-26211] - Unit tests: AST_TEST_DEFINE should be used in conditional code.
  • [ASTERISK-26212] - [patch] Makefile: Retain XML Declaration and DTD in docs.
  • [ASTERISK-26214] - Allow arbitrary time for fax detection to end on a channel
  • [ASTERISK-26216] - res_fax: Deadlock when detect fax while channel executing Playback
  • [ASTERISK-26227] - sqlalchemy error due to long identifier name
  • [ASTERISK-26237] - Fax is detected on regular calls.

Improvement

  • [ASTERISK-22131] - Update the make dependencies script to pull, build, and install the correct pjproject
  • [ASTERISK-25471] - [patch]Add subscribe_context to res_pjsip
  • [ASTERISK-26159] - res_hep: enabled by default and information sent to default address
  • [ASTERISK-26220] - Add support for noreturn function attributes.

New Feature

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.0-rc1

Thank you for your continued support of Asterisk!


Asterisk 14.0.0-beta1 Now Available

Jul 27, 2016

The Asterisk Development Team has announced the first beta of Asterisk 14.0.0.

This beta is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.0.0-beta1 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 beta: Release Notes - Asterisk - Version 14.0.0

Bug

  • [ASTERISK-7803] - [patch] Update the maximum packetization values in frame.c
  • [ASTERISK-13271] - menuselect sets defaults too late
  • [ASTERISK-13797] - [patch] relax badshell tilde test
  • [ASTERISK-14233] - [patch] Buddies are always auto-registered when processing the roster
  • [ASTERISK-15242] - transmit_refer leaks sip_refer structures
  • [ASTERISK-15434] - [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller
  • [ASTERISK-15879] - [patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
  • [ASTERISK-16115] - [patch] problem with ringinuse=no, queue members receive sometimes two calls
  • [ASTERISK-16779] - Cannot disallow unknown format ''
  • [ASTERISK-17588] - Caller ID on TDM410P *UK* PSTN
  • [ASTERISK-17608] - func_aes.so cannot be loaded if res_crypto / openssl not compiled
  • [ASTERISK-17721] - Incoming SRTP calls that specify a key lifetime fail
  • [ASTERISK-18032] - [patch] - IPv6 and IPv4 NAT not working
  • [ASTERISK-18105] - most of asterisk modules are unbuildable in cygwin environment
  • [ASTERISK-18252] - queue_log mysql time column data format
  • [ASTERISK-18708] - func_curl hangs channel under load
  • [ASTERISK-18923] - res_fax_spandsp usage counter is wrong
  • [ASTERISK-19277] - [patch]endlessly repeating error: "poll failed: Bad file descriptor"
  • [ASTERISK-19470] - Documentation on app_amd is incorrect
  • [ASTERISK-19608] - Asterisk-1.8.x starts rejecting calls with cause code 44 after some time.
  • [ASTERISK-20127] - [Regression] Config.c config_text_file_load() unescapes semicolons ("\;" -> ";") turning them into comments (corruption) on rewrite of a config file
  • [ASTERISK-20233] - SRTP not working with some devices (Eg Grandstream gxv3175) - Message "Can't provide secure audio requested in SDP offer"
  • [ASTERISK-20399] - Compilation on some systems requires the -fnested-functions flag
  • [ASTERISK-20524] - AMI improperly handles lines of exactly 1025 characters
  • [ASTERISK-20567] - bashism in autosupport
  • [ASTERISK-20744] - [patch] Security event logging does not work over syslog
  • [ASTERISK-20784] - Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
  • [ASTERISK-20850] - [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
  • [ASTERISK-20987] - non-admin users, who join muted conference are not being muted
  • [ASTERISK-21038] - Bad command completion of "core set debug channel"
  • [ASTERISK-21211] - chan_iax2 - unprotected access of iaxs[peer->callno] potentially results in segfault
  • [ASTERISK-21721] - SIP Failed to parse multiple Supported: headers
  • [ASTERISK-21765] - [patch] - FILE function's length argument counts from beginning of file rather than the offset
  • [ASTERISK-21777] - Asterisk tries to transcode video instead of audio
  • [ASTERISK-21845] - maxcalls exceeded, Asterisk sends out 480 and also BYE
  • [ASTERISK-21893] - Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c
  • [ASTERISK-22252] - res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks
  • [ASTERISK-22352] - [patch] IAX2 custom qualify timer is not taken into account
  • [ASTERISK-22455] - Asterisk 12 on Ubuntu Lucid deadlocks with DEBUG_THREADS+OPTIONAL_API enabled
  • [ASTERISK-22559] - gcc 4.6 and higher supports weakref attribute but asterisk doesn't detect it.
  • [ASTERISK-22670] - Asterisk crashes when processing ISDN AoC Events
  • [ASTERISK-22708] - res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work
  • [ASTERISK-22748] - SRTP Crypto Offer With Lifetime Not Accepted
  • [ASTERISK-22790] - check_modem_rate() may return incorrect rate for V.27
  • [ASTERISK-22791] - asterisk sends Re-INVITE after receiving a BYE
  • [ASTERISK-22805] - res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
  • [ASTERISK-22945] - [patch] Memory leaks in chan_sip.c with realtime peers
  • [ASTERISK-23013] - [patch] Deadlock between 'sip show channels' command and attended transfer handling
  • [ASTERISK-23214] - chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases
  • [ASTERISK-23231] - Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load
  • [ASTERISK-23319] - Segmentation fault in queue_exec at app_queue.c
  • [ASTERISK-23390] - NewExten Event with application AGI shows up before and after AGI runs
  • [ASTERISK-23508] - Memory Corruption in __ast_string_field_ptr_build_va
  • [ASTERISK-23577] - res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL
  • [ASTERISK-23634] - With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls
  • [ASTERISK-23651] - Reloading some modules that are loaded already, results in 'No such module' before a successful reload
  • [ASTERISK-23666] - CLONE - nested functions aren't portable
  • [ASTERISK-23733] - 'reload acl' fails if acl.conf is not present on startup
  • [ASTERISK-23767] - [patch] Dynamic IAX2 registration stops trying if ever not able to resolve
  • [ASTERISK-23768] - [patch] Asterisk man page contains a (new) unquoted minus sign
  • [ASTERISK-23781] - outgoing missing as enum from contrib/ast-db-manage/config
  • [ASTERISK-23841] - DTMF atxfer doesn't set CallerID for the recall calls to the transferrer.
  • [ASTERISK-23846] - Unistim multilines. Loss of voice after second call drops (on a second line).
  • [ASTERISK-23850] - Park Application does not respect Return Context Priority
  • [ASTERISK-23991] - [patch]asterisk.pc file contains a small error in the CFlags returned
  • [ASTERISK-23994] - res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname
  • [ASTERISK-23997] - chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer
  • [ASTERISK-24011] - [patch]safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM
  • [ASTERISK-24015] - app_transfer fails with PJSIP channels
  • [ASTERISK-24019] - When a Music On Hold stream starts it restarts at beginning of file.
  • [ASTERISK-24027] - MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up
  • [ASTERISK-24032] - Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined
  • [ASTERISK-24043] - ARI /continue fails to actually continue into the dialplan
  • [ASTERISK-24048] - [patch] contrib/scripts/install_prereq selects 32-bit packages on 64-bit hosts
  • [ASTERISK-24049] - Asterisk Manager Interface: A number of list type responses aren't using astman_send_listack
  • [ASTERISK-24063] - [patch]Asterisk does not respect outbound proxy when sending qualify requests
  • [ASTERISK-24085] - Documentation - We should remove or further document the 'contact' section in pjsip.conf
  • [ASTERISK-24097] - Documentation - CHANNEL function help text missing 'linkedid' argument
  • [ASTERISK-24106] - WebSockets Automatically decides what driver it will use
  • [ASTERISK-24122] - Documentaton for res_pjsip option use_avpf needs to be fixed
  • [ASTERISK-24134] - ARI: GET /channels/{channel_id}/variable for channel in dialplan returns 409 conflict
  • [ASTERISK-24136] - Security: Crash in Asterisk's PJSIP code when subscribing to an event with an unexpected body type
  • [ASTERISK-24138] - dial: Call forwarding information presented through AMI/ARI is wrong
  • [ASTERISK-24142] - CCSS: crash during shutdown due to device lookup in destroyed container
  • [ASTERISK-24143] - pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK
  • [ASTERISK-24146] - [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec
  • [ASTERISK-24147] - ARI: channel hangup crashes asterisk process
  • [ASTERISK-24155] - [patch]Non-portable and non-reliable recursion detection in ast_malloc
  • [ASTERISK-24161] - PJSIPShowEndpoint gives inaccurate count of list items
  • [ASTERISK-24178] - [patch]fromdomainport used even if not set
  • [ASTERISK-24181] - RLS: Large lists don't get sent because they exceed the PJSIP message length limit
  • [ASTERISK-24190] - IMAP voicemail causes segfault
  • [ASTERISK-24195] - bridge_native_rtp: Removing mixmonitor from a native RTP capable smart bridge doesn't cause the bridge to resume being a native rtp bridge
  • [ASTERISK-24199] - 'ALL' is specified in pjsip.conf.sample for TLS cipher but it is not valid
  • [ASTERISK-24208] - Channels with CDR Information Remain Active Even After ConfBrige Is Ended
  • [ASTERISK-24212] - testsuite: Sporadic crash due to assert on stopping RTP engine
  • [ASTERISK-24215] - testsuite: ARI Live Dangerously test fails due to wrong response code from Asterisk
  • [ASTERISK-24222] - PJSIP: Failed assertions when placing a call with no allow= specified
  • [ASTERISK-24223] - Gibberish Call-ID on Local channel on origination
  • [ASTERISK-24224] - When using Bridge() dialplan application, surrogate channel appears in list and call count is inflated.
  • [ASTERISK-24225] - Dial option z is broken
  • [ASTERISK-24229] - ARI: playback of sounds implicitly answers channel, preventing early media playback
  • [ASTERISK-24231] - crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable
  • [ASTERISK-24234] - app_meetme: Crash on conference shutdown due to NULL channel passed to meetme_stasis_generate_msg()
  • [ASTERISK-24236] - res_hep_rtcp: Module incorrectly depends on pjsip
  • [ASTERISK-24237] - CDR: FRACK With PJSIP blonde transfer.
  • [ASTERISK-24241] - crash: CDRs recursively attempt to update Party B information in a multi-party bridge, overrunning the stack
  • [ASTERISK-24245] - gcc 4.1.2 complains of files that do not end with newlines
  • [ASTERISK-24246] - Quiet warning about type qualifiers ignored on function return type
  • [ASTERISK-24249] - SIP debugs do not stop
  • [ASTERISK-24250] - [patch] Voicemail with multi-recipients To: header fix
  • [ASTERISK-24254] - CDRs: Application/args/dialplan CEP updated during dial operation
  • [ASTERISK-24257] - agent must dial acceptdtmf twice to bridge to queue caller
  • [ASTERISK-24262] - AMI CoreShowChannel missing several output fields and event documentation
  • [ASTERISK-24264] - ARI: Adding a channel to a holding bridge automatically starts MOH
  • [ASTERISK-24265] - segfault in asterisk when try to make call to IAX
  • [ASTERISK-24267] - Queue variables associated with setinterfacevar, setqueueentryvar, setqueuevar are not passed to local channel
  • [ASTERISK-24271] - Unable to make WebRTC call through chan_PJSIP nor chan_SIP
  • [ASTERISK-24274] - [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
  • [ASTERISK-24280] - Add 'rtpbindaddr' setting for chan_sip
  • [ASTERISK-24281] - When bridging 2 chan_sip channels, MOH not removed from on-hold channels and bridge is never destroyed after hangup.
  • [ASTERISK-24288] - [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup
  • [ASTERISK-24290] - Endpoint identifier match value fails to parse when CIDR network format is specified
  • [ASTERISK-24295] - crash: creating out of dialog OPTIONS request crashes
  • [ASTERISK-24301] - Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk
  • [ASTERISK-24304] - asterisk crashing randomly because of unistim channel
  • [ASTERISK-24307] - Unintentional memory retention in stringfields
  • [ASTERISK-24312] - SIGABRT when improperly configured realtime pjsip
  • [ASTERISK-24321] - SIP deadlock when running automated queues tests
  • [ASTERISK-24323] - Bug in documentation AGI STREAM FILE CONTROL
  • [ASTERISK-24325] - res_calendar_ews: cannot be used with neon 0.30
  • [ASTERISK-24326] - res_rtp_asterisk: ICE-TCP candidates are incorrectly attempted
  • [ASTERISK-24327] - bridge_native_rtp: Smart bridge operation to softmix sometimes fails to properly re-INVITE remotely bridged participants
  • [ASTERISK-24328] - Use of MixMonitor 'm' option results in 0 duration vm description file
  • [ASTERISK-24331] - Unexpected Errors in Asterisk Manager Interface Output
  • [ASTERISK-24335] - [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls
  • [ASTERISK-24336] - PJSIP timer_min_se value under 90 causes crash
  • [ASTERISK-24337] - Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped'
  • [ASTERISK-24339] - Swagger API Docs have incorrect basePath
  • [ASTERISK-24342] - PJSIP: Qualifying endpoints attempts to do them all at the same time.
  • [ASTERISK-24344] - CDR_PROP(disable) disables CDR only for first dialed party
  • [ASTERISK-24348] - Built-in editline tab complete segfault with MALLOC_DEBUG
  • [ASTERISK-24350] - PJSIP shows commands prints unneeded headers
  • [ASTERISK-24354] - AMI sendMessage closes AMI connection on error
  • [ASTERISK-24355] - [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser'
  • [ASTERISK-24356] - PJSIP: Directed pickup causes deadlock
  • [ASTERISK-24357] - [fax] Out of bounds error in update_modem_bits
  • [ASTERISK-24362] - res_hep leaks reference to configuration
  • [ASTERISK-24367] - PJSIP: allow all results in failure to send INVITE
  • [ASTERISK-24368] - res_pjsip_pubsub: Subscription persistence causes crash when re-constructing stored subscription
  • [ASTERISK-24369] - res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations
  • [ASTERISK-24370] - res_pjsip/pjsip_options: OPTIONS request sent to Asterisk with no user in request is always 404'd
  • [ASTERISK-24376] - res_pjsip_refer: REFER request for remote session attempts to direct channel to external_replaces extension instead of context, without providing for the Referred-To SIP URI
  • [ASTERISK-24378] - Release AMI connections on shutdown
  • [ASTERISK-24380] - core: Native formats are set to h264 with certain audio/video codec configuration, resulting in path translation WARNINGs
  • [ASTERISK-24381] - res_pjsip_sdp_rtp: Declined media streams are interpreted, leading to erroneous 488 rejections
  • [ASTERISK-24382] - chan_pjsip: Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel results in an invalid reference of a channel pvt and a FRACK
  • [ASTERISK-24383] - res_rtp_asterisk: Crash if no candidates received for component
  • [ASTERISK-24384] - chan_motif: format capabilities leak on module load error
  • [ASTERISK-24387] - res_pjsip: rport sent from UAS MUST include the port that the UAC sent the request on
  • [ASTERISK-24389] - chan_iax2: Unit test on Bamboo failing
  • [ASTERISK-24392] - res_fax: fax gateway sessions leak
  • [ASTERISK-24393] - rtptimeout=0 doesn't disable rtptimeout
  • [ASTERISK-24394] - CDR: FRACK with PJSIP directed pickup.
  • [ASTERISK-24398] - Initialize auth_rejection_permanent on client state to the configuration parameter value
  • [ASTERISK-24406] - Some caller ID strings are parsed differently since 11.13.0
  • [ASTERISK-24411] - [patch] Status of outbound registration is not changed upon unregistering.
  • [ASTERISK-24413] - parking/parking_tests: Crash due to assertion in unit tests when MoH is started on channel in holding bridge
  • [ASTERISK-24415] - Missing AMI VarSet events when channels inherit variables.
  • [ASTERISK-24419] - Incorrect syntax for setting language in configs/extensions.conf.sample
  • [ASTERISK-24425] - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
  • [ASTERISK-24426] - CDR Batch mode: size used as time value after first expire
  • [ASTERISK-24430] - missing letter "p" in word response in OriginateResponse event documentation
  • [ASTERISK-24432] - Install refcounter.py when REF_DEBUG is enabled
  • [ASTERISK-24435] - Asterisk 13 with TC400P segfault
  • [ASTERISK-24436] - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
  • [ASTERISK-24437] - Review implementation of ast_bridge_impart for leaks and document proper usage
  • [ASTERISK-24438] - res_pjsip_multihomed.so blocks Asterisk reload when DNS settings invalid
  • [ASTERISK-24442] - Outgoing call files don't work properly when set in the future
  • [ASTERISK-24443] - CDR fields (dst, dcontext) empty in transfer call started from Macro
  • [ASTERISK-24444] - PBX: Crash when generating extension for pattern matching hint
  • [ASTERISK-24447] - Bridge DTMF hooks: Audio doesn't pass when waiting for more matching digits.
  • [ASTERISK-24449] - Reinvite for T.38 UDPTL fails if SRTP is enabled
  • [ASTERISK-24451] - chan_iax2: reference leak in sched_delay_remove
  • [ASTERISK-24453] - manager: acl_change_sub leaks
  • [ASTERISK-24454] - app_queue: ao2_iterator not destroyed, causing leak
  • [ASTERISK-24455] - func_cdr: CDR_PROP leaks payload
  • [ASTERISK-24457] - res_fax: fax gateway frames leak
  • [ASTERISK-24458] - chan_phone fails to build on big endian systems
  • [ASTERISK-24459] - bridge_native_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible
  • [ASTERISK-24462] - res_pjsip: Stale qualify statistics after disablementation
  • [ASTERISK-24463] - Voicemail email address corrupt or not sent when message is in the process of being recorded during reload
  • [ASTERISK-24465] - audiohooks list leaks reference to formats
  • [ASTERISK-24466] - app_queue: fix a couple leaks to struct call_queue
  • [ASTERISK-24468] - Incoming UCS2 encoded SMS truncated if SMS length exceeds 50 (roughly) national symbols
  • [ASTERISK-24469] - Security Vulnerability: Mixed IPv4/IPv6 ACLs allow blocked addresses through
  • [ASTERISK-24471] - Crash - assert_fail in libc in pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
  • [ASTERISK-24472] - Asterisk Crash in OpenSSL when calling over WSS from JSSIP
  • [ASTERISK-24474] - sip_to_pjsip.py lacks documentation and does not function
  • [ASTERISK-24476] - main/app.c / app_voicemail: ast_writestream leaks
  • [ASTERISK-24479] - Enable REF_DEBUG for module references
  • [ASTERISK-24480] - res_http_websockets: Module reference decrease below zero
  • [ASTERISK-24482] - func_talkdetect: Fix stasis message leak in audiohook callback
  • [ASTERISK-24485] - res_pjsip cannot be unloaded or shutdown
  • [ASTERISK-24487] - configuration: sections should be loadable as template even when not marked
  • [ASTERISK-24489] - Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1
  • [ASTERISK-24490] - Security Vulnerability: CONFBRIDGE function's record_command option allows arbitrary parameters to be passed to MixMonitor, allowing remote execution of commands
  • [ASTERISK-24491] - Memory leak in res_hep
  • [ASTERISK-24492] - main/file.c: ast_filestream sometimes causes extra calls to ast_module_unref
  • [ASTERISK-24498] - Segmentation fault in res_hep_rtcp on attended transfer
  • [ASTERISK-24499] - Need more explicit debug when PJSIP dialstring is invalid
  • [ASTERISK-24500] - Regression introduced in chan_mgcp by SVN revision r227276
  • [ASTERISK-24501] - ARI: Moving a channel between bridges followed by a hangup can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd
  • [ASTERISK-24502] - Build fails when dev-mode, dont optimize and coverage are enabled
  • [ASTERISK-24504] - chan_console: Fix reference leaks to pvt
  • [ASTERISK-24505] - manager: http connections leak references
  • [ASTERISK-24508] - pjsip - REFER request from SNOM is rejected with "400 bad request" - DEBUG shows "Received a REFER without a parseable Refer-To"
  • [ASTERISK-24513] - Local channel apparently leaked in off-nominal DTMF attended transfer
  • [ASTERISK-24514] - res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard
  • [ASTERISK-24516] - [patch]Asterisk segfaults when playing back voicemail under high concurrency with an IMAP backend
  • [ASTERISK-24522] - ConfBridge: delay occurs between kicking all endmarked users when last marked user leaves
  • [ASTERISK-24528] - res_pjsip_refer: Sending INVITE with Replaces in-dialog with invalid target causes crash
  • [ASTERISK-24531] - res_pjsip_acl: ACLs not applied on initial module load
  • [ASTERISK-24533] - 2 threads created per chan_sip entry
  • [ASTERISK-24534] - [patch]Register DB() as escalating to prevent users from writing to astdb
  • [ASTERISK-24535] - stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix
  • [ASTERISK-24536] - AMI redirect with PJSIP fails to move extra channel
  • [ASTERISK-24537] - Stasis: StasisStart/StasisEnd events are not reliably transmitted during transfers
  • [ASTERISK-24539] - Compile fails on OSX because of sem_timedwait in bridge_channel.c
  • [ASTERISK-24542] - [patch]Failure showing codecs via 'core show channeltype <tech>'
  • [ASTERISK-24543] - Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs
  • [ASTERISK-24544] - Compile fails on OSX Yosemite because of incorrect detection of htonll and ntohll
  • [ASTERISK-24550] - res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake
  • [ASTERISK-24556] - Asterisk 13 core dumps when calling from pjsip extension to another pjsip extension
  • [ASTERISK-24560] - Creating a named ARI bridge twice causes a crash
  • [ASTERISK-24563] - Direct Media calls within private network sometimes get one way audio
  • [ASTERISK-24566] - Uninit buf in WS write
  • [ASTERISK-24572] - [patch]App_meetme is loaded without its defaults when the configuration file is missing
  • [ASTERISK-24573] - [patch]Out of sync conversation recording when divided in multiple recordings
  • [ASTERISK-24591] - Stasis() side of an ARI originated channel cannot be Redirected
  • [ASTERISK-24596] - Unclear how to use Park application with res_parking 'parkeddynamic' enabled. Documentation?
  • [ASTERISK-24600] - Stuck IAX channels, Asterisk stops responding to most traffic, potential deadlock
  • [ASTERISK-24604] - res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core
  • [ASTERISK-24605] - res_parking option parkeddynamic does not work with the core Features 'parkcall' (DTMF initiated parking)
  • [ASTERISK-24607] - res_pjsip_session: re-INVITE with declined media streams results in 488
  • [ASTERISK-24612] - res_pjsip: No information if a required sorcery wizard is not loaded
  • [ASTERISK-24614] - Deadlock when DEBUG_THREADS compiler flag enabled
  • [ASTERISK-24615] - When Multiple Transports Exist in pjsip.conf, Incorrect External Addresses is Used in SIP Packets When Responding to INVITE
  • [ASTERISK-24616] - Crash in res_format_attr_h264 due to invalid string copy
  • [ASTERISK-24619] - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int
  • [ASTERISK-24624] - Transfer to invalid extension results in hung channel.
  • [ASTERISK-24626] - Voicemail passwords not being stored in ARA
  • [ASTERISK-24628] - [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment)
  • [ASTERISK-24632] - install_prereq script installs pjproject without IPv6 support
  • [ASTERISK-24635] - PJSIP outbound PUBLISH crashes when no response is ever received
  • [ASTERISK-24637] - Channel re-enters Stasis() when it should not
  • [ASTERISK-24640] - Registration pending stays forever after sip reload
  • [ASTERISK-24641] - Deadlock in Trunk
  • [ASTERISK-24646] - PJSIP changeset 4899 breaks TLS
  • [ASTERISK-24649] - Pushing of channel into bridge fails; Stasis fails to get app name
  • [ASTERISK-24651] - [patch] Fix race condition in DTLS
  • [ASTERISK-24655] - res_pjsip_outbound_publish: Hang on shutdown while attempting to publish
  • [ASTERISK-24663] - [patch] Unnamed semaphore autoconf check fails on cross compilation
  • [ASTERISK-24665] - Configure check required for pjsip_get_dest_info()
  • [ASTERISK-24666] - Security Vulnerability: RTP not closed after sip call using unsupported codec
  • [ASTERISK-24672] - [PATCH] Memory leak in func_curl CURLOPT
  • [ASTERISK-24673] - outgoing sip registers cannot be removed or modified without doing restart (or doing module unload chan_sip.so)
  • [ASTERISK-24676] - Security Vulnerability: URL request injection in libCURL (CVE-2014-8150)
  • [ASTERISK-24677] - ARI GET variable on channel provides unhelpful response on non-existent variable
  • [ASTERISK-24682] - app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT are an unexpected value
  • [ASTERISK-24683] - Crash in PBX ast_hashtab_lookup_internal during core restart now
  • [ASTERISK-24685] - "pjsip show version" CLI command
  • [ASTERISK-24689] - Segfault on hangup after outgoing PRI-Euroisdn call
  • [ASTERISK-24693] - Investigate and fix memory leaks in Asterisk
  • [ASTERISK-24700] - CRASH: NULL channel is being passed to ast_bridge_transfer_attended()
  • [ASTERISK-24701] - Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information
  • [ASTERISK-24709] - [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event
  • [ASTERISK-24711] - DTLS handshake broken with latest OpenSSL versions
  • [ASTERISK-24715] - chan_sip: stale nonce causes failure
  • [ASTERISK-24716] - Improve pjsip log messages for presence subscription failure
  • [ASTERISK-24717] - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10}
  • [ASTERISK-24719] - ConfBridge recording channels get stuck when recording started/stopped more than once
  • [ASTERISK-24721] - manager: ModuleLoad action incorrectly reports 'module not found' during a Reload operation
  • [ASTERISK-24723] - confbridge: CLI command 'confbridge list XXXX' no longer displays user menus
  • [ASTERISK-24724] - 'httpstatus' Web Page Produces Incomplete HTML
  • [ASTERISK-24727] - PJSIP: Crash experienced during multi-Asterisk transfer scenario.
  • [ASTERISK-24728] - tcptls: Bad file descriptor error when reloading chan_sip
  • [ASTERISK-24729] - Outbound registration not occuring on new registrations after reload.
  • [ASTERISK-24731] - res_pjsip_session cannot be unloaded
  • [ASTERISK-24736] - Memory Leak Fixes
  • [ASTERISK-24737] - When agent not logged in, agent status shows unavailable, queue status shows agent invalid
  • [ASTERISK-24739] - [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules
  • [ASTERISK-24740] - [patch]Segmentation fault on aoc-e event
  • [ASTERISK-24741] - dtls_handler causes Asterisk to crash
  • [ASTERISK-24742] - [patch] Fix ast_odbc_find_table function in res_odbc
  • [ASTERISK-24748] - res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur
  • [ASTERISK-24749] - ConfBridge: Wrong language on playing conf-hasjoin and conf-hasleft when played to bridge
  • [ASTERISK-24751] - Integer values in json payload to ARI cause asterisk to crash
  • [ASTERISK-24752] - Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown
  • [ASTERISK-24755] - Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge
  • [ASTERISK-24768] - res_timing_pthread: file descriptor leak
  • [ASTERISK-24769] - res_pjsip_sdp_rtp: Local ICE candidates leaked
  • [ASTERISK-24771] - ${CHANNEL(pjsip)} - segfault
  • [ASTERISK-24772] - ODBC error in realtime sippeers when device unregisters under MariaDB
  • [ASTERISK-24774] - Segfault in ast_context_destroy with extensions.ael and extensions.conf
  • [ASTERISK-24779] - Passthrough OPUS codec not working with chan_pjsip
  • [ASTERISK-24780] - [patch] - Buddies are always auto-registered when processing the roster
  • [ASTERISK-24781] - PJSIP: Unnecessary 180 Ringing messages sent with undesireabe consequences.
  • [ASTERISK-24782] - StasisEnd event not present for channel that was swapped out for another after completing attended transfer
  • [ASTERISK-24785] - 'Expires' header missing from 200 OK on REGISTER
  • [ASTERISK-24786] - [patch] - Asterisk terminates when playing a voicemail stored in LDAP
  • [ASTERISK-24787] - [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime
  • [ASTERISK-24791] - Crash in ast_rtcp_write_report
  • [ASTERISK-24796] - Codecs and bucket schema's prevent module unload
  • [ASTERISK-24797] - bridge_softmix: G.729 codec license held
  • [ASTERISK-24799] - [patch] make fails with undefined reference to SSLv3_client_method
  • [ASTERISK-24800] - Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill
  • [ASTERISK-24801] - ASAN: ast_el_read_char stack-buffer-overflow
  • [ASTERISK-24805] - [patch] - ASAN: Race condition (heap-use-after-free) on asterisk closing
  • [ASTERISK-24807] - Missing mandatory field Max-Forwards
  • [ASTERISK-24808] - res_config_odbc: Improper escaping of backslashes occurs with MySQL
  • [ASTERISK-24812] - ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding
  • [ASTERISK-24814] - asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers
  • [ASTERISK-24817] - init_logger_chain: unreachable code block
  • [ASTERISK-24825] - Caller ID not recognized using Centrex/Distinctive dialing
  • [ASTERISK-24828] - Fix Frame Leaks
  • [ASTERISK-24830] - res_rtp_asterisk.c checks USE_PJPROJECT not HAVE_PJPROJECT
  • [ASTERISK-24832] - [patch]DTLS-crashes within openssl
  • [ASTERISK-24835] - Early Media Not working with Chan SIP and Asterisk 13
  • [ASTERISK-24838] - chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling
  • [ASTERISK-24840] - res_pjsip: conflicting endpoint identifiers
  • [ASTERISK-24841] - ConfBridge: Strange sampling rates chosen when channels have multiple native formats
  • [ASTERISK-24845] - pjsip send notify not working with Cisco phone
  • [ASTERISK-24847] - [security] [patch] tcptls: certificate CN NULL byte prefix bug
  • [ASTERISK-24853] - Documentation claims chan_sip outbound registrations support WS or WSS as valid transports (not true)
  • [ASTERISK-24857] - [patch] "timing test", pjsip incoming/outgoing calls, voicemail prompts and recordings all fail when using the kqueue timer source on FreeBSD 10.x
  • [ASTERISK-24863] - res_pjsip: No endpoint events raised via AMI when contacts cannot be reached/qualified
  • [ASTERISK-24864] - app_confbridge: file playback blocks dtmf
  • [ASTERISK-24867] - Docs for 'e' option in ResetCDR say to use CDR_PROP instead, CDR_PROP docs are unclear
  • [ASTERISK-24869] - Asterisk segfaults on DAHDI attended transfer due to application (appl) being NULL on unbridged channel
  • [ASTERISK-24872] - [patch] AMI PJSIPShowEndpoint closes AMI connection on error
  • [ASTERISK-24876] - Investigate reference leaks from tests/channels/local/local_optimize_away
  • [ASTERISK-24879] - [patch]Compilation fails due to 64bit time under OpenBSD
  • [ASTERISK-24880] - [patch]Compilation under OpenBSD
  • [ASTERISK-24881] - ast_register_atexit should only be used when absolutely needed
  • [ASTERISK-24882] - chan_sip: Improve usage of REF_DEBUG
  • [ASTERISK-24887] - [patch]tags in a=crypto lines do not accept 2 or more digits
  • [ASTERISK-24894] - [patch] iax2_poke_noanswer expiration timer too short
  • [ASTERISK-24895] - After hangup on the side of the ISDN network no HangupRequest event comes for the dahdi channel.
  • [ASTERISK-24896] - [patch] Using force black background leads to colours not being reset
  • [ASTERISK-24899] - Parking fall-through behavior different in 13
  • [ASTERISK-24900] - Manager event ParkedCallSwap is not documented
  • [ASTERISK-24907] - res_pjsip_outbound_registration: crash during unload if registration attempts are still occuring
  • [ASTERISK-24910] - "timer=no" and "timer=required" settings in pjsip.conf fail
  • [ASTERISK-24914] - Division by zero in file.c when playback of voicemail with video as h264
  • [ASTERISK-24920] - Asterisk handles duplicate SIP requests as if they were each a new request
  • [ASTERISK-24928] - [patch]t38_udptl_maxdatagram in pjsip.conf not honored
  • [ASTERISK-24932] - Asterisk 13.x does not build with GCC 5.0
  • [ASTERISK-24933] - T38 fails negotiation
  • [ASTERISK-24934] - [patch]Asterisk manager output does not escape control characters
  • [ASTERISK-24935] - res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator.
  • [ASTERISK-24936] - New Feature: AO2 weakproxy objects
  • [ASTERISK-24937] - [patch]res_pjsip_messaging: Messages may be sent out of order
  • [ASTERISK-24938] - ARI Snoop Channel results in excessive escalating CPU usage
  • [ASTERISK-24944] - main/audiohook.c change prevents G722 call recording
  • [ASTERISK-24954] - Git migration: Asterisk version numbers are incompatible with the Test Suite
  • [ASTERISK-24955] - res_fax: v.27ter support baud rate of 2400, which is disallowed in res_fax's check_modem_rate
  • [ASTERISK-24958] - Forwarding loop detection inhibits certain desirable scenarios
  • [ASTERISK-24959] - [patch]CLI command cdr show pgsql status
  • [ASTERISK-24963] - ASAN: heap-use-after-free with PJSIP and WSS
  • [ASTERISK-24967] - Problem support schema for pgsql on CEL
  • [ASTERISK-24970] - Crash in res_pjsip_pubsub handling of failed notify
  • [ASTERISK-24972] - Transport Layer Security (TLS) Protocol BEAST Vulnerability - Investigate vulnerability of HTTP server
  • [ASTERISK-24975] - Enabling 'DEBUG_THREADLOCALS' Causes the Build to Fail
  • [ASTERISK-24976] - cdr_odbc not include new columns added on 1.8
  • [ASTERISK-24977] - Contacts that don't use qualify are being marked as unavailable
  • [ASTERISK-24982] - res_pjsip_mwi: Unsolicited MWI NOTIFY only sent on mailbox changes
  • [ASTERISK-24983] - IAX deadlock between hangup and scheduled actions (ex. largrq)
  • [ASTERISK-24986] - keepalive INFO packages ignored by asterisk
  • [ASTERISK-24988] - func_talkdetect: Test is bouncing sporadically
  • [ASTERISK-24991] - Check for ao2_alloc failure in __ast_channel_internal_alloc
  • [ASTERISK-24994] - dns: Query set unit tests are failing due to race condition
  • [ASTERISK-24996] - chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR Sections Exist in pjsip.conf
  • [ASTERISK-24997] - Astobj2: Some callers of __adjust_lock do not pre-check the object
  • [ASTERISK-24998] - res_corosync: res_corosync tries to load even if res_corosync.conf is missing
  • [ASTERISK-24999] - PJSIP crashes with malformed contact line
  • [ASTERISK-25003] - Asterisk crashes on attended transfer (using feature)
  • [ASTERISK-25004] - Crash in authenticated reinvite after originated T.38 FAX
  • [ASTERISK-25018] - pjsip show endpoints crashes asterisk when qualified aors present
  • [ASTERISK-25020] - Mismatched response to outgoing REGISTER request
  • [ASTERISK-25022] - Memory leak setting up DTLS/SRTP calls
  • [ASTERISK-25023] - Deadlock in chan_sip in update_provisional_keepalive
  • [ASTERISK-25025] - Periodic crashes (in ast_channel_snapshot_create at stasis_channels.c) with Certified Asterisk 13.
  • [ASTERISK-25026] - Git conversion: Non-C files not switched to ASTERISK_REGISTER_FILE
  • [ASTERISK-25027] - Build System: Many ARI modules are missing dependencies.
  • [ASTERISK-25028] - Build System: Unneeded defines in asterisk/buildopts.h
  • [ASTERISK-25033] - Asterisk 13 (branch head) won't compile without PJSip
  • [ASTERISK-25034] - chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests.
  • [ASTERISK-25037] - res_pjsip_outbound_registration: Potential crash in off-nominal failure case when sending message
  • [ASTERISK-25038] - Queue log "EXITWITHTIMEOUT" does not always contain waiting time
  • [ASTERISK-25041] - [patch]Broken column type checking in res_config_mysql addon
  • [ASTERISK-25042] - asterisk.conf options override command-line options.
  • [ASTERISK-25048] - Astobj2: Initialization order wrong when both refdebug and AO2_DEBUG are both enabled.
  • [ASTERISK-25053] - Unit test category /main/presence missing trailing slash.
  • [ASTERISK-25054] - Formats interface's cannot be unregistered, needs to hold modules until shutdown.
  • [ASTERISK-25057] - res_pjsip_pubsub: Crash in send_notify due to invalid root pointer in sub_tree
  • [ASTERISK-25061] - pbx_config: Register manager actions with module version of macro.
  • [ASTERISK-25074] - Regression: Recent clang-related change broke cross compiling of Asterisk
  • [ASTERISK-25076] - res_pjsip: Failover does not occur on connection-less transport or 503 response
  • [ASTERISK-25082] - Asterisk deletes message after doing a playback of an INBOX message using ast_vm_play when the Old folder is full for that mailbox.
  • [ASTERISK-25083] - Message.c: Message channel becomes saturated with frames leading to spammy log messages
  • [ASTERISK-25085] - [patch]Potential crash after unload of func_periodic_hook or test_message
  • [ASTERISK-25086] - [patch]PJSIP crashes if endpoint missing in Dial()
  • [ASTERISK-25087] - Asterisk segfault when using Directory application with alias option and specific mailbox configuration
  • [ASTERISK-25089] - res_pjsip_config_wizard: Variable specified in templates aren't being processed correctly
  • [ASTERISK-25090] - CLI core show channel truncates cdr variables
  • [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-25108] - configure check for older unbound library
  • [ASTERISK-25110] - res_resolver_unbound.c compilation failure: SIGURG is undeclared in func unbound_resolver_stop
  • [ASTERISK-25112] - Logger: Configuration settings are not reset to default during reload.
  • [ASTERISK-25113] - install_prereq in Debian 8 without "standard system utilities"
  • [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-25120] - Astobj2: Weakproxy subscriptions should be run in reverse order.
  • [ASTERISK-25121] - Stasis: Fix unsafe use of stasis_unsubscribe in modules.
  • [ASTERISK-25122] - Large SIP packet received via pjsip over websocket crashes Asterisk
  • [ASTERISK-25123] - Bracketed IPv6 Contact header parameter unparsable with Asterisk/PJSIP
  • [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-25135] - [patch]RTP Timeout hangup cause code missing
  • [ASTERISK-25137] - endpoint stasis messages are delivered twice
  • [ASTERISK-25146] - DNS: Create system level resolver
  • [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-25160] - [patch] Opus Codec: SIP/SDP line fmtp missing when called internally
  • [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-25179] - CDR(billsec,f) and CDR(duration,f) report incorrect values
  • [ASTERISK-25180] - res_pjsip_mwi: Unsolicited MWI requires reload
  • [ASTERISK-25181] - ARI: Channels added to Stasis application during WebSocket creation don't receive a StasisStart event
  • [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-25185] - Segfault in app_queue on transfer scenarios
  • [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-25215] - Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember
  • [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-25222] - Crash in recurring cancel callback called from ast_dns_resolve_cancel on junk pointer
  • [ASTERISK-25226] - chan_sip: Channel leak in branch 13 on early replaces call pickup
  • [ASTERISK-25227] - No audio at in-band announcements in ooh323 channel
  • [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
  • [ASTERISK-25262] - Memory leak when a caller channel does multiple dials and CEL is enabled
  • [ASTERISK-25263] - [patch]cdr_adaptive_odbc: CDR insert failure due to reversed if logic
  • [ASTERISK-25265] - [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1
  • [ASTERISK-25271] - Parking & blind transfer: Transferer channel not hung up if no MOH
  • [ASTERISK-25272] - [patch]The ICONV dialplan function sometimes returns garbage
  • [ASTERISK-25289] - Build System does not respect CFLAGS and CXXFLAGS when building menuselect
  • [ASTERISK-25292] - Testuite: tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
  • [ASTERISK-25295] - res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h
  • [ASTERISK-25296] - RTP performance issue with several channel drivers.
  • [ASTERISK-25297] - Crashes running channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
  • [ASTERISK-25304] - res_pjsip: XML sanitization may write past buffer
  • [ASTERISK-25305] - Dynamic logger channels can be added multiple times
  • [ASTERISK-25306] - Persistent subscriptions can save multiple SIP messages at once, leading to potential crashes.
  • [ASTERISK-25308] - ari: Websocket leak
  • [ASTERISK-25309] - [patch] iLBC 20 advertised
  • [ASTERISK-25312] - res_http_websocket: Terminate connection on fatal cases
  • [ASTERISK-25315] - DAHDI channels send shortened duration DTMF tones.
  • [ASTERISK-25318] - tests/rest_api/applications/subscribe-endpoint/nominal/resource: Sporadically failing
  • [ASTERISK-25320] - chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite
  • [ASTERISK-25321] - [patch]DeadLock ChanSpy with call over Local channel
  • [ASTERISK-25322] - Crash occurs when using MixMonitor with t() or r() options.
  • [ASTERISK-25325] - ARI PUT reload chan_sip HTTP response 404
  • [ASTERISK-25337] - Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub
  • [ASTERISK-25339] - res_pjsip: Empty "auth" sections from non-config backgrounds are interpreted as valid
  • [ASTERISK-25341] - bridge: Hangups may get lost when executing actions
  • [ASTERISK-25342] - res_pjsip: Repeated usage of pj_gethostip may block
  • [ASTERISK-25346] - chan_sip: Overwriting answered elsewhere hangup cause on call pickup
  • [ASTERISK-25352] - res_hep_rtcp correlation_id is different then res_hep
  • [ASTERISK-25353] - [patch] Transcoding while different in Frame size = Frames lost
  • [ASTERISK-25355] - sched: ast_sched_del may return prematurely due to spurious wakeup
  • [ASTERISK-25356] - res_pjsip_sdp_rtp: Multiple keepalive scheduled items may exist
  • [ASTERISK-25364] - [patch]Issue a TCP connection(kernel) and thread of asterisk is not released
  • [ASTERISK-25365] - Persistent subscriptions have extra Content-Length/corrupted messages
  • [ASTERISK-25367] - pbx: Long pattern match hints may cause "core show hints" to crash
  • [ASTERISK-25369] - res_parking: ParkAndAnnounce - Inheritable variables aren't applied to the announcer channel
  • [ASTERISK-25373] - add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants
  • [ASTERISK-25381] - res_pjsip: AoRs deleted via ARI (or other mechanism) do not destroy their related contacts
  • [ASTERISK-25383] - Core dumps on startup and shutdown with MALLOC_DEBUG enabled
  • [ASTERISK-25384] - Regular Asterisk crashes when using Page application. "user_data is NULL"
  • [ASTERISK-25387] - res_pjsip_nat: Malformed REGISTER request causes NAT'd Contact header to not be rewritten
  • [ASTERISK-25390] - default_from_user can crash with certain configuration backends
  • [ASTERISK-25391] - AMI GetConfigJSON returns invalid JSON
  • [ASTERISK-25394] - pbx: Incorrect device and presence state when changing hint details
  • [ASTERISK-25396] - chan_sip: Extremely long callerid name causes invalid SIP
  • [ASTERISK-25397] - [patch]chan_sip: File descriptor leak with non-default timert1
  • [ASTERISK-25399] - app_queue: AgentComplete event has wrong reason
  • [ASTERISK-25400] - Hints broken when "CustomPresence" doesn't exist in AstDB
  • [ASTERISK-25404] - segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c
  • [ASTERISK-25407] - Asterisk fails to log to multiple syslog destinations
  • [ASTERISK-25410] - app_record: RECORDED_FILE variable not being populated
  • [ASTERISK-25418] - On-hold channels redirected out of a bridge appear to still be on hold
  • [ASTERISK-25423] - Caller gets no Connected line update during call pickup.
  • [ASTERISK-25434] - Compiler flags not reported in 'core show settings' despite usage during compilation
  • [ASTERISK-25435] - Asterisk periodically hangs. UDP Recv-Q greatly exceeds zero.
  • [ASTERISK-25438] - res_rtp_asterisk: ICE role message even when ICE is not enabled
  • [ASTERISK-25441] - Deadlock in res_sorcery_memory_cache.
  • [ASTERISK-25442] - using realtime (mysql) queue members are never updated in wait_our_turn function (app_queue.c)
  • [ASTERISK-25443] - [patch]IPv6 - Potential issue in via header parsing
  • [ASTERISK-25449] - main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny
  • [ASTERISK-25451] - Broken video - erased rtp marker bit
  • [ASTERISK-25455] - Deadlock of PJSIP realtime over res_config_pgsql
  • [ASTERISK-25461] - Nested dialplan #includes don't work as expected.
  • [ASTERISK-25476] - chan_sip loses registrations after a while
  • [ASTERISK-25484] - [patch] autoframing=yes has no effect
  • [ASTERISK-25485] - res_pjsip_outbound_registration: registration stops due to 400 response
  • [ASTERISK-25486] - res_pjsip: Fix deadlock when validating URIs
  • [ASTERISK-25494] - build: GCC 5.1.x catches some new const, array bounds and missing paren issues
  • [ASTERISK-25498] - Asterisk crashes when negotiating g729 without that module installed
  • [ASTERISK-25505] - res_pjsip_pubsub: Crash on off-nominal when UAS dialog can't be created
  • [ASTERISK-25510] - [patch]Log to syslog failing
  • [ASTERISK-25513] - Crash: malloc failed with high load of subscriptions.
  • [ASTERISK-25522] - ARI: Crash when creating channel via ARI originate with requesting channel
  • [ASTERISK-25527] - Quirky xmldoc description wrapping
  • [ASTERISK-25528] - DNS: System resolver issues with TTL parse
  • [ASTERISK-25533] - [patch] buffer for ast_format_cap_get_names only 64 bytes
  • [ASTERISK-25535] - [patch] format creation on module load instead of cache
  • [ASTERISK-25537] - [patch] format-attribute module: RFC or internal defaults?
  • [ASTERISK-25545] - [patch] translation module gets cached not joint format
  • [ASTERISK-25546] - threadpool: Race condition between idle timeout and activation
  • [ASTERISK-25552] - hashtab: Improve NULL tolerance
  • [ASTERISK-25561] - app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked!
  • [ASTERISK-25565] - DNS: System resolver only returns 1 record per result
  • [ASTERISK-25569] - app_meetme: Audio quality issues
  • [ASTERISK-25573] - [patch] H.264 format attribute module: resets whole SDP
  • [ASTERISK-25575] - res_pjsip: Dynamic outbound registrations created via ARI are not loaded into memory on Asterisk start/restart
  • [ASTERISK-25582] - Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38
  • [ASTERISK-25583] - [patch] format-attribute module: RFC 7587 (Opus Codec)
  • [ASTERISK-25584] - [patch] format-attribute module: VP8 missing
  • [ASTERISK-25585] - [patch]rasterisk never hits most of main(), but it's assumed to
  • [ASTERISK-25590] - CLI Usage info for 'pjsip send notify' references incorrect config
  • [ASTERISK-25593] - fastagi: record file closed after sending result
  • [ASTERISK-25595] - Unescaped : in messge sent to statsd
  • [ASTERISK-25598] - res_pjsip: Contact status messages are printing a hash instead of the uri
  • [ASTERISK-25599] - [patch] SLIN Resampling Codec only 80 msec
  • [ASTERISK-25600] - bridging: Inconsistency in BRIDGEPEER
  • [ASTERISK-25601] - json: Audit reference usage and thread safety
  • [ASTERISK-25603] - [patch]udptl: Uninitialized lengths and bufs in udptl_rx_packet cause ast_frdup crash
  • [ASTERISK-25606] - Core dump when using transports in sorcery
  • [ASTERISK-25608] - res_pjsip/contacts/statsd: Lifecycle events aren't consistent
  • [ASTERISK-25609] - [patch]Asterisk may crash when calling ast_channel_get_t38_state(c)
  • [ASTERISK-25610] - Asterisk crash during "sip reload"
  • [ASTERISK-25611] - core: threadpool thread_timeout_thrash unit test sporadically failing
  • [ASTERISK-25614] - DTLS negotiation delays
  • [ASTERISK-25615] - res_pjsip: Setting transport async_operations > 1 causes segfault on tls transports
  • [ASTERISK-25616] - Warning with a Codec Module which supports PLC with FEC
  • [ASTERISK-25619] - res_chan_stats not sending the correct information to StatsD
  • [ASTERISK-25624] - AMI Event OriginateResponse bug
  • [ASTERISK-25625] - res_sorcery_memory_cache: Add full backend caching
  • [ASTERISK-25632] - res_pjsip_sdp_rtp: RTP is sent from wrong IP address when multihomed
  • [ASTERISK-25637] - Multi homed server using wrong IP
  • [ASTERISK-25640] - pbx: Deadlock on features reload and state change hint.
  • [ASTERISK-25641] - bridge: GOTO_ON_BLINDXFR doesn't work on transfer initiated channel
  • [ASTERISK-25647] - bug of cel_radius.c: wrong point of ADD_VENDOR_CODE
  • [ASTERISK-25659] - res_rtp_asterisk: ECDH not negotiated causing DTLS failure occurred on RTP instance
  • [ASTERISK-25664] - ast_format_cap_append_by_type leaks a reference
  • [ASTERISK-25668] - res_pjsip: Deadlock in distributor
  • [ASTERISK-25669] - [patch]CURL incorrect trim for non ASCII characters
  • [ASTERISK-25673] - res_crypto leaks CLI entries
  • [ASTERISK-25675] - Endpoint not listed as Unreachable
  • [ASTERISK-25677] - pbx_dundi: leaks during failed load.
  • [ASTERISK-25679] - res_calendar leaks scheduler.
  • [ASTERISK-25680] - manager: manager_channelvars is not cleaned at shutdown
  • [ASTERISK-25681] - devicestate: Engine thread is not shut down
  • [ASTERISK-25683] - res_ari: Asterisk fails to start if compiled with MALLOC_DEBUG
  • [ASTERISK-25685] - infrastructure: Run alembic in Jenkins build script
  • [ASTERISK-25686] - PJSIP: qualify_timeout is a double, database schema is an integer
  • [ASTERISK-25687] - res_musiconhold: Concurrent invocations of 'moh reload' cause a crash
  • [ASTERISK-25690] - Hanging up when executing connected line sub does not cause hangup
  • [ASTERISK-25696] - bridge_basic: don't cache xferfailsound during a transfer
  • [ASTERISK-25697] - bridge_basic: don't play an attended transfer fail sound after target hangs up
  • [ASTERISK-25700] - main/config: Clean config maps on shutdown.
  • [ASTERISK-25702] - PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2
  • [ASTERISK-25707] - Long contact URIs or hostnames can crash pjproject/Asterisk under certain conditions
  • [ASTERISK-25709] - ARI: Crash can occur due to race condition when attempting to operate on a hung up channel
  • [ASTERISK-25712] - Second call to already-on-call phone and Asterisk sends "Ready"
  • [ASTERISK-25714] - ASAN:heap-buffer-overflow in logger.c
  • [ASTERISK-25721] - [patch] res_phoneprov: memory leak and heap-use-after-free
  • [ASTERISK-25722] - ASAN & testsute: stack-buffer-overflow in sip_sipredirect
  • [ASTERISK-25725] - core: Incorrect XML documentation may result in weird behavior
  • [ASTERISK-25727] - RPM build requires OPTIONAL_API cflag due to PJSIP requirement
  • [ASTERISK-25730] - build: make uninstall after make distclean tries to remove root
  • [ASTERISK-25737] - res_pjsip_outbound_registration: line option not in Alembic
  • [ASTERISK-25738] - res_pjsip_pubsub: Crash while executing OutboundSubscriptionDetail ami action
  • [ASTERISK-25742] - Secondary IFP Packets can result in accessing uninitialized pointers and a crash
  • [ASTERISK-25751] - res_pjsip: Support pjsip_dlg_create_uas_and_inc_lock
  • [ASTERISK-25771] - ARI:Crash - Attended transfers of channels into Stasis application.
  • [ASTERISK-25772] - res_pjsip: Unexpected two BYE when answered
  • [ASTERISK-25777] - data race in threadpool
  • [ASTERISK-25796] - res_pjsip: DOS/Crash when TCP/TLS sockets exceed pjproject PJ_IOQUEUE_MAX_HANDLES
  • [ASTERISK-25800] - [patch] Calculate talktime when is first call answered
  • [ASTERISK-25811] - Unable to delete object from sorcery cache
  • [ASTERISK-25814] - Segfault at f ip in res_pjsip_refer.so
  • [ASTERISK-25825] - Crashes during shutdown when running CLI commands
  • [ASTERISK-25826] - PJSIP / Sorcery slow load from realtime
  • [ASTERISK-25829] - res_pjsip: PJSIP does not accept spaces when separating multiple AORs
  • [ASTERISK-25830] - Revision 2451d4e breaks NAT
  • [ASTERISK-25849] - chan_pjsip: transfers with direct media sometimes drops audio
  • [ASTERISK-25854] - No audio after HOLD/RESUME - incorrect a=recvonly in SDP from Asterisk
  • [ASTERISK-25857] - func_aes: incorrect use of strlen() leads to data corruption
  • [ASTERISK-25867] - [patch] Video delay on app_echo
  • [ASTERISK-25868] - Sorcery "append to category" should allow filters
  • [ASTERISK-25873] - res_pjsip: Bundled pjproject: compile error, cannot find -lasteriskpj
  • [ASTERISK-25874] - app_voicemail: Stack buffer overflow in test_voicemail_notify_endl
  • [ASTERISK-25881] - pbx: Add support for autohints
  • [ASTERISK-25882] - ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2)
  • [ASTERISK-25885] - res_pjsip: Race condition between adding contact and automatic expiration
  • [ASTERISK-25888] - Frequent segfaults in function can_ring_entry() of app_queue.c
  • [ASTERISK-25890] - Asterisk 13.8.0 alembic database update fails
  • [ASTERISK-25894] - [patch] webrtc video broken due to missing marker bits in RTP streams
  • [ASTERISK-25910] - pjproject: Via headers are not parsed when "received" contains an IPv6 address
  • [ASTERISK-25912] - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding them to the local hangupcauses via ast_channel_hangupcause_hash_set
  • [ASTERISK-25917] - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself
  • [ASTERISK-25922] - res_pjsip_exten_state: Add configuration support for publishing
  • [ASTERISK-25927] - Removed option "registertrying" is still documented in sip.conf.sample
  • [ASTERISK-25928] - res_pjsip: URI validation done outside of PJSIP thread
  • [ASTERISK-25929] - res_pjsip_registrar: AOR_CONTACT_ADDED events not raised
  • [ASTERISK-25934] - chan_sip should not require sipregs or updateable sippeers table unless rt
  • [ASTERISK-25938] - res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero.
  • [ASTERISK-25941] - chan_pjsip: Crash on an immediate SIP final response
  • [ASTERISK-25942] - res_pjsip_caller_id: Transfer results in mixed ConnectedLine information
  • [ASTERISK-25947] - Protocol transfers to stasis applications are missing the StasisStart with the replace_channel object.
  • [ASTERISK-25950] - [patch]SIP channel does not send PeerStatus events for autocreated peers
  • [ASTERISK-25951] - res_agi: run_agi eats frames it shouldn't
  • [ASTERISK-25954] - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName
  • [ASTERISK-25956] - Compilation error in conditionally compiled code in config_options.c
  • [ASTERISK-25959] - http_media_cache/retrieve_cache_control_directives: Sporadic failure
  • [ASTERISK-25961] - tests/channels/SIP/sip_tls_call: Sporadic crash when running test
  • [ASTERISK-25963] - func_odbc requires reconnect checks for stale connections
  • [ASTERISK-25964] - Outbound registrations created via ARI/push configuration do not clean up outbound registrations currently in flight
  • [ASTERISK-25968] - pjproject_bundled: Configure and make need to be re-tested
  • [ASTERISK-25970] - Segfault in pjsip_url_compare
  • [ASTERISK-25978] - res_pjsip_authenticator_digest: Should not use source port in nonce verification
  • [ASTERISK-25990] - PJSIP TLS registration should respect client_uri scheme when generating Contact URI
  • [ASTERISK-25993] - pjproject: Allow bundling to not require everything it does
  • [ASTERISK-25998] - file: Crash when using nativeformats
  • [ASTERISK-25999] - res_pjsip_dialog_info_body_generator: Remove subscription requirement
  • [ASTERISK-26004] - res_pjsip: The transport/method parameter is ignored
  • [ASTERISK-26005] - res_pjsip: Multiple SIP messages are combined into 1 TCP packet
  • [ASTERISK-26007] - res_pjsip: Endpoints deleting early after upgrade from 13.8.2 to 13.9
  • [ASTERISK-26008] - app_followme does not delete recorded name prompt
  • [ASTERISK-26014] - res_sorcery_astdb: Make tolerant of unknown fields
  • [ASTERISK-26021] - Build codecs siren7 and siren14 for Asterisk 13
  • [ASTERISK-26029] - parking: ast_parking_park_call should return parking_space instead of parking_exten
  • [ASTERISK-26030] - call cut because of double Session-Expires header in re-invite after proxy authentication is required
  • [ASTERISK-26034] - T.38 passthrough problem behind firewall due to early nosignal packet
  • [ASTERISK-26038] - 'make install' doesn't seem to install OS/X init files
  • [ASTERISK-26045] - [patch]app_voicemail: fix bugs, imap mm_status log change to debug
  • [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf.
  • [ASTERISK-26047] - ARI allows certain commands to run on down channels.
  • [ASTERISK-26049] - res_pjsip: Crash when our own request timer fires
  • [ASTERISK-26053] - res_pjsip_outbound_publish: Crash when shutting down
  • [ASTERISK-26054] - Asterisk crashes (core dump)
  • [ASTERISK-26063] - ${PJSIP_HEADER(read,Call-ID)} does not work - documentation needs clarification for when read/write is possible
  • [ASTERISK-26065] - chan_pjsip: MWI NOTIFY contents not ordered properly
  • [ASTERISK-26069] - Asterisk truncates To: header, dropping the closing '>'
  • [ASTERISK-26070] - ari/channels: Creating a local channel without an originator adds all audio formats to it's capabilities
  • [ASTERISK-26074] - res_odbc: Deadlock within UnixODBC
  • [ASTERISK-26078] - core: Memory leak in logging
  • [ASTERISK-26083] - ARI: Announcer channels staying around after playback to a bridge is finished
  • [ASTERISK-26089] - Invalid security events during boot using PJSIP Realtime
  • [ASTERISK-26091] - [patch] ar cru creates warning, instead use ar cr
  • [ASTERISK-26092] - [Segfault] in res_rtp_asterisk.c:4268 after Remotely bridged channels
  • [ASTERISK-26096] - res_hep: Crash when configuration file is missing
  • [ASTERISK-26097] - [patch] CLI: show maximum file descriptors
  • [ASTERISK-26099] - res_pjsip_pubsub: Crash when sending request due to server timeout
  • [ASTERISK-26103] - cdr: Assert on 'dial end' event during a blond transfer
  • [ASTERISK-26119] - [patch] fix: memory leaks, resource leaks, out of bounds and bugs
  • [ASTERISK-26126] - [patch] leverage 'bindaddr' for TLS in http.conf
  • [ASTERISK-26127] - res_pjsip_session: Crash due to race condition between res_pjsip_session unload and timer
  • [ASTERISK-26128] - Alembic scripts are failing
  • [ASTERISK-26130] - [patch] WebRTC: Should use latest DTLS version.
  • [ASTERISK-26132] - PJSIP: provide transport type with received messages
  • [ASTERISK-26133] - app_queue: Queue members receive multiple calls
  • [ASTERISK-26138] - chan_unistim: Under FreeBSD, chan_unistim generates a compile error
  • [ASTERISK-26139] - test_res_pjsip_scheduler: Compile failure if pjproject isn't installed in a system location
  • [ASTERISK-26140] - res_rtp_asterisk: gcc 6 caught a self-comparison
  • [ASTERISK-26141] - res_fax: fax_v21_session_new leaks reference to v21_details
  • [ASTERISK-26144] - Crash on loading codecs g729/g723
  • [ASTERISK-26157] - Build: Fix errors highlighted by GCC 6.x
  • [ASTERISK-26160] - pjsip: Updated->Reachable during qualify
  • [ASTERISK-26177] - func_odbc: Database handle is kept when it should be released
  • [ASTERISK-26179] - chan_sip: Second T.38 request fails
  • [ASTERISK-26180] - PJSIP: provide valid tcp nodelay option for reuse
  • [ASTERISK-26181] - REF_DEBUG: Node object incorrectly logged during duplicate replacement
  • [ASTERISK-26184] - chan_sip: Reference leaks in error paths.
  • [ASTERISK-26191] - threadpool: Leak on duplicate taskprocessor for ast_threadpool_serializer_group
  • [ASTERISK-26193] - chan_sip: reference leak in mwi_event_cb
  • [ASTERISK-26196] - pbx: Time based includes can leak timezone string
  • [ASTERISK-26207] - [patch] sRTP: Count a roll-over of the sequence number even on lost packets.
  • [ASTERISK-26211] - Unit tests: AST_TEST_DEFINE should be used in conditional code.
  • [ASTERISK-26212] - [patch] Makefile: Retain XML Declaration and DTD in docs.
  • [ASTERISK-26214] - Allow arbitrary time for fax detection to end on a channel
  • [ASTERISK-26216] - res_fax: Deadlock when detect fax while channel executing Playback
  • [ASTERISK-26221] - chan_sip: iLBC does not include correct mode
  • [ASTERISK-26227] - sqlalchemy error due to long identifier name

Improvement

New Feature

  • [ASTERISK-17899] - Handle crypto lifetime in SDES-SRTP negotiation
  • [ASTERISK-22591] - [patch]Prevent Asterisk from writing received SMS content in log
  • [ASTERISK-23186] - [patch] Add usegmtime option to cel_pgsql
  • [ASTERISK-23823] - [patch] Option to keep queuerules in realtime
  • [ASTERISK-23871] - RLS Tests: Implement RLS off-nominal tests
  • [ASTERISK-24276] - [Patch] Option to make app MOH override channel musicclass
  • [ASTERISK-24363] - [patch] Add ability for Channel Drivers to provide Presence State information
  • [ASTERISK-24554] - AMI/ARI: Generate events on connected line changes
  • [ASTERISK-24703] - ARI: Add the ability to "transfer" (redirect) a channel
  • [ASTERISK-24834] - DNS Overhaul: Implement the proposed core API - sync/async functions, resolver registration
  • [ASTERISK-24836] - DNS Overhaul: Write a Resolver Implementation
  • [ASTERISK-24919] - res_pjsip_config_wizard: Ability to write contents to file
  • [ASTERISK-24922] - ARI: Add the ability to intercept hold and raise an event
  • [ASTERISK-24931] - dns: Add support for SRV records.
  • [ASTERISK-25006] - [patch] Add support set character for quoted identifiers
  • [ASTERISK-25173] - ARI: Add the ability to load/reload/unload an Asterisk module
  • [ASTERISK-25238] - ARI: Support push configuration
  • [ASTERISK-25252] - ARI: Add the ability to manipulate log channels
  • [ASTERISK-25259] - chan_pjsip: Add rtptimeout support
  • [ASTERISK-25377] - res_pjsip: Change default "From user" from UUID to something more palatable
  • [ASTERISK-25419] - Dialplan Application for Integration of StatsD
  • [ASTERISK-25425] - logger: Add JSON structured logging
  • [ASTERISK-25479] - Allow CDR's to be modified before being dispatched to engines
  • [ASTERISK-25480] - [patch]Add field PauseReason on QueueMemberStatus
  • [ASTERISK-25549] - Confbridge: Add participant timeout option
  • [ASTERISK-25551] - [patch]Ability to add channel to an existing bridge by specifying an existing channel prefix
  • [ASTERISK-25591] - [patch] Complete List of Header Files (#include): iwyu
  • [ASTERISK-25660] - Add sipp-sendfax.xml and spandspflow2pcap.py to contrib/scripts.
  • [ASTERISK-25670] - Add regcontext to PJSIP
  • [ASTERISK-25803] - [patch] chan_sip: Optionally supply fromuser/fromdomain in SIP dial string
  • [ASTERISK-25889] - ARI: Add separate "create" and "dial" operations for channels
  • [ASTERISK-25900] - PJSIP Endpoint IP Access Controls
  • [ASTERISK-25904] - PJSIP: add contact.updated event
  • [ASTERISK-25925] - Allow Early Bridges on ARI Dials
  • [ASTERISK-25972] - res_pjsip_exten_state: Use body generator to publish extension state
  • [ASTERISK-26042] - ARI: Allow downloading of the media associated with a stored recording
  • [ASTERISK-26058] - [Patch] Add uptime and last reloaded to FullyBooted AMI event
  • [ASTERISK-26068] - Multicast RTP Options

For a full list of changes in this beta, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.0-beta1

Thank you for your continued support of Asterisk!


Asterisk 13.10.0 Now Available

Jul 21, 2016

The Asterisk Development Team has announced the release of Asterisk 13.10.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.10.0 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-16115] - [patch] problem with ringinuse=no, queue members receive sometimes two calls
  • [ASTERISK-24436] - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
  • [ASTERISK-24463] - Voicemail email address corrupt or not sent when message is in the process of being recorded during reload
  • [ASTERISK-24986] - keepalive INFO packages ignored by asterisk
  • [ASTERISK-25262] - Memory leak when a caller channel does multiple dials and CEL is enabled
  • [ASTERISK-25352] - res_hep_rtcp correlation_id is different then res_hep
  • [ASTERISK-25777] - data race in threadpool
  • [ASTERISK-25826] - PJSIP / Sorcery slow load from realtime
  • [ASTERISK-25917] - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself
  • [ASTERISK-25938] - res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero.
  • [ASTERISK-25941] - chan_pjsip: Crash on an immediate SIP final response
  • [ASTERISK-25950] - [patch]SIP channel does not send PeerStatus events for autocreated peers
  • [ASTERISK-25954] - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName
  • [ASTERISK-25956] - Compilation error in conditionally compiled code in config_options.c
  • [ASTERISK-25961] - tests/channels/SIP/sip_tls_call: Sporadic crash when running test
  • [ASTERISK-25963] - func_odbc requires reconnect checks for stale connections
  • [ASTERISK-25964] - Outbound registrations created via ARI/push configuration do not clean up outbound registrations currently in flight
  • [ASTERISK-25968] - pjproject_bundled: Configure and make need to be re-tested
  • [ASTERISK-25970] - Segfault in pjsip_url_compare
  • [ASTERISK-25978] - res_pjsip_authenticator_digest: Should not use source port in nonce verification
  • [ASTERISK-25990] - PJSIP TLS registration should respect client_uri scheme when generating Contact URI
  • [ASTERISK-25993] - pjproject: Allow bundling to not require everything it does
  • [ASTERISK-25998] - file: Crash when using nativeformats
  • [ASTERISK-26005] - res_pjsip: Multiple SIP messages are combined into 1 TCP packet
  • [ASTERISK-26007] - res_pjsip: Endpoints deleting early after upgrade from 13.8.2 to 13.9
  • [ASTERISK-26008] - app_followme does not delete recorded name prompt
  • [ASTERISK-26014] - res_sorcery_astdb: Make tolerant of unknown fields
  • [ASTERISK-26029] - parking: ast_parking_park_call should return parking_space instead of parking_exten
  • [ASTERISK-26030] - call cut because of double Session-Expires header in re-invite after proxy authentication is required
  • [ASTERISK-26034] - T.38 passthrough problem behind firewall due to early nosignal packet
  • [ASTERISK-26038] - 'make install' doesn't seem to install OS/X init files
  • [ASTERISK-26054] - Asterisk crashes (core dump)
  • [ASTERISK-26063] - ${PJSIP_HEADER(read,Call-ID)} does not work - documentation needs clarification for when read/write is possible
  • [ASTERISK-26065] - chan_pjsip: MWI NOTIFY contents not ordered properly
  • [ASTERISK-26069] - Asterisk truncates To: header, dropping the closing '>'
  • [ASTERISK-26070] - ari/channels: Creating a local channel without an originator adds all audio formats to it's capabilities
  • [ASTERISK-26074] - res_odbc: Deadlock within UnixODBC
  • [ASTERISK-26078] - core: Memory leak in logging
  • [ASTERISK-26083] - ARI: Announcer channels staying around after playback to a bridge is finished
  • [ASTERISK-26089] - Invalid security events during boot using PJSIP Realtime
  • [ASTERISK-26091] - [patch] ar cru creates warning, instead use ar cr
  • [ASTERISK-26092] - [Segfault] in res_rtp_asterisk.c:4268 after Remotely bridged channels
  • [ASTERISK-26096] - res_hep: Crash when configuration file is missing
  • [ASTERISK-26097] - [patch] CLI: show maximum file descriptors
  • [ASTERISK-26099] - res_pjsip_pubsub: Crash when sending request due to server timeout
  • [ASTERISK-26126] - [patch] leverage 'bindaddr' for TLS in http.conf
  • [ASTERISK-26127] - res_pjsip_session: Crash due to race condition between res_pjsip_session unload and timer
  • [ASTERISK-26128] - Alembic scripts are failing
  • [ASTERISK-26130] - [patch] WebRTC: Should use latest DTLS version.
  • [ASTERISK-26138] - chan_unistim: Under FreeBSD, chan_unistim generates a compile error
  • [ASTERISK-26139] - test_res_pjsip_scheduler: Compile failure if pjproject isn't installed in a system location
  • [ASTERISK-26140] - res_rtp_asterisk: gcc 6 caught a self-comparison
  • [ASTERISK-26141] - res_fax: fax_v21_session_new leaks reference to v21_details
  • [ASTERISK-26160] - pjsip: Updated->Reachable during qualify
  • [ASTERISK-26177] - func_odbc: Database handle is kept when it should be released

Improvement

New Feature

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.10.0

Thank you for your continued support of Asterisk!


Asterisk 11.23.0 Now Available

Jul 21, 2016

The Asterisk Development Team has announced the release of Asterisk 11.23.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.23.0 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-16115] - [patch] problem with ringinuse=no, queue members receive sometimes two calls
  • [ASTERISK-24436] - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
  • [ASTERISK-24463] - Voicemail email address corrupt or not sent when message is in the process of being recorded during reload
  • [ASTERISK-25407] - Asterisk fails to log to multiple syslog destinations
  • [ASTERISK-25510] - [patch]Log to syslog failing
  • [ASTERISK-25874] - app_voicemail: Stack buffer overflow in test_voicemail_notify_endl
  • [ASTERISK-25888] - Frequent segfaults in function can_ring_entry() of app_queue.c
  • [ASTERISK-25912] - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding them to the local hangupcauses via ast_channel_hangupcause_hash_set
  • [ASTERISK-25917] - [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself
  • [ASTERISK-25934] - chan_sip should not require sipregs or updateable sippeers table unless rt
  • [ASTERISK-25954] - Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName
  • [ASTERISK-26008] - app_followme does not delete recorded name prompt
  • [ASTERISK-26030] - call cut because of double Session-Expires header in re-invite after proxy authentication is required
  • [ASTERISK-26034] - T.38 passthrough problem behind firewall due to early nosignal packet
  • [ASTERISK-26038] - 'make install' doesn't seem to install OS/X init files
  • [ASTERISK-26069] - Asterisk truncates To: header, dropping the closing '>'
  • [ASTERISK-26091] - [patch] ar cru creates warning, instead use ar cr
  • [ASTERISK-26097] - [patch] CLI: show maximum file descriptors
  • [ASTERISK-26126] - [patch] leverage 'bindaddr' for TLS in http.conf
  • [ASTERISK-26130] - [patch] WebRTC: Should use latest DTLS version.
  • [ASTERISK-26138] - chan_unistim: Under FreeBSD, chan_unistim generates a compile error
  • [ASTERISK-26140] - res_rtp_asterisk: gcc 6 caught a self-comparison
  • [ASTERISK-26141] - res_fax: fax_v21_session_new leaks reference to v21_details

Improvement

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0

Thank you for your continued support of Asterisk!


Asterisk 13.10.0-rc3 Now Available

Jul 14, 2016

The Asterisk Development Team has announced the third release candidate of Asterisk 13.10.0.

This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.10.0-rc3 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 candidate: Release Notes - Asterisk - Version 13.10.0

Bug

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.10.0-rc3

Thank you for your continued support of Asterisk!


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