Asterisk 14.0.0-beta1 Now Available

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] – Update the maximum packetization values in frame.c
• [ASTERISK-13271] – menuselect sets defaults too late
• [ASTERISK-13797] – relax badshell tilde test
• [ASTERISK-14233] – Buddies are always auto-registered when processing the roster
• [ASTERISK-15242] – transmit_refer leaks sip_refer structures
• [ASTERISK-15434] – When ast_pbx_start failed, both an error response and BYE are sent to the caller
• [ASTERISK-15879] – Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
• [ASTERISK-16115] – 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] – – 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] – 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] – 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] – 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] – – 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] – 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] – Memory leaks in chan_sip.c with realtime peers
• [ASTERISK-23013] – 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] – Dynamic IAX2 registration stops trying if ever not able to resolve
• [ASTERISK-23768] – 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] – 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] – 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] – 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] – 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] – No audio on WebRtc caller side when answer waiting time is more than ~7sec
• [ASTERISK-24147] – ARI: channel hangup crashes asterisk process
• [ASTERISK-24155] – Non-portable and non-reliable recursion detection in ast_malloc
• [ASTERISK-24161] – PJSIPShowEndpoint gives inaccurate count of list items
• [ASTERISK-24178] – 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] – 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] – 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] – – 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] – 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] – 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] – 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] – 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] – 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] – 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] – App_meetme is loaded without its defaults when the configuration file is missing
• [ASTERISK-24573] – 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] – 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] – 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] – Fix race condition in DTLS
• [ASTERISK-24655] – res_pjsip_outbound_publish: Hang on shutdown while attempting to publish
• [ASTERISK-24663] – 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] – 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] – – Out of files — call fails — numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules
• [ASTERISK-24740] – Segmentation fault on aoc-e event
• [ASTERISK-24741] – dtls_handler causes Asterisk to crash
• [ASTERISK-24742] – 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] – – 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] – – Asterisk terminates when playing a voicemail stored in LDAP
• [ASTERISK-24787] – – 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] – 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] – – 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] – 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] 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] – “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] – AMI PJSIPShowEndpoint closes AMI connection on error
• [ASTERISK-24876] – Investigate reference leaks from tests/channels/local/local_optimize_away
• [ASTERISK-24879] – Compilation fails due to 64bit time under OpenBSD
• [ASTERISK-24880] – 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] – tags in a=crypto lines do not accept 2 or more digits
• [ASTERISK-24894] – 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] – 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] – 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] – 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] – 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] – 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] – 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] – Potential crash after unload of func_periodic_hook or test_message
• [ASTERISK-25086] – 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] – 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] – 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] – 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] – 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] – 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] – Segfault when using DEBUG_FD_LEAKS
• [ASTERISK-25215] – Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember
• [ASTERISK-25219] – Source and destination overlap in memcpy in rtp_engine.c
• [ASTERISK-25220] – 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] – 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] – cdr_adaptive_odbc: CDR insert failure due to reversed if logic
• [ASTERISK-25265] – 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] – 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] – 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] – 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] – 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] – 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] – 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] – 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] – 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] – 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] – buffer for ast_format_cap_get_names only 64 bytes
• [ASTERISK-25535] – format creation on module load instead of cache
• [ASTERISK-25537] – format-attribute module: RFC or internal defaults?
• [ASTERISK-25545] – 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] – 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] – format-attribute module: RFC 7587 (Opus Codec)
• [ASTERISK-25584] – format-attribute module: VP8 missing
• [ASTERISK-25585] – 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] – SLIN Resampling Codec only 80 msec
• [ASTERISK-25600] – bridging: Inconsistency in BRIDGEPEER
• [ASTERISK-25601] – json: Audit reference usage and thread safety
• [ASTERISK-25603] – 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] – 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] – 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] – 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] – 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] – 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] – 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] – 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] – 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] – app_voicemail: fix bugs, imap mm_status log change to debug
• [ASTERISK-26046] – 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] – 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] – 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] – fix: memory leaks, resource leaks, out of bounds and bugs
• [ASTERISK-26126] – 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] – 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] – 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] – 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

• [ASTERISK-22131] – Update the make dependencies script to pull, build, and install the correct pjproject
• [ASTERISK-23324] – – QLOOG commiting Japanese translated prompts
• [ASTERISK-23512] – Inaccurate comment in manager.conf.sample
• [ASTERISK-23953] – Testsuite: Off-nominal Authenticate test
• [ASTERISK-24038] – device state: Report ONHOLD device state if channel driver defers device state calculation to core
• [ASTERISK-24045] – Voicemail to email at multiple email addresses
• [ASTERISK-24128] – [Patch] Adding default dtls settings
• [ASTERISK-24133] – Please support Clang; Allow no-exec stacks
• [ASTERISK-24171] – Provide a manpage for the aelparse utility
• [ASTERISK-24279] – Documentation: Clarify the behaviour of the CDR property ‘unanswered’
• [ASTERISK-24283] – Microseconds precision in the eventtime column in the cel_odbc module
• [ASTERISK-24316] – For httpd server, need option to define server name for security purposes
• [ASTERISK-24351] – Allow passing options and command to MixMonitor when recording in ConfBridge
• [ASTERISK-24365] – [Patch] Dialplan function to get first/head caller channel on queue
• [ASTERISK-24412] – Incomplete channel originate/continue handling with ARI
• [ASTERISK-24530] – app_record stripping 1/4 second from recordings
• [ASTERISK-24552] – ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes
• [ASTERISK-24553] – ARI/AMI: Include language in standard channel snapshot output
• [ASTERISK-24575] – Make capath work for res_pjsip
• [ASTERISK-24577] – Speed up loopback switches by avoiding unneeded lookups
• [ASTERISK-24671] – Missing docs for the CDR AMI Event
• [ASTERISK-24678] – [PATCH] Added atxfer* settings to features.conf.sample
• [ASTERISK-24706] – add auto-dtmf mode for pjsip
• [ASTERISK-24718] – Add inital support of “sanitize” to configure
• [ASTERISK-24744] – Swedish Core Voice prompts
• [ASTERISK-24745] – Add no_answer to ARI hangup causes
• [ASTERISK-24790] – Reduce spurious noise in logs from voicemail – Couldn’t find mailbox %s in context
• [ASTERISK-24802] – stasis: set a channel variable on websocket disconnect error
• [ASTERISK-24811] – asterisk-publication sorcery object does not use realtime
• [ASTERISK-24813] – asterisk.c: #if statement in listener() confuses code folding editors
• [ASTERISK-24815] – Enable TLS Dual-Certificates (ECC+RSA)
• [ASTERISK-24862] – Support in-dialog OPTIONS
• [ASTERISK-24870] – ARI: Subscriptions to bridges generally not super useful
• [ASTERISK-24892] – Super Awesome Company sound prompts
• [ASTERISK-24917] – clang compilation warnings
• [ASTERISK-24918] – pjsip: add CLI options to display global and system configuration
• [ASTERISK-24939] – IAX make calltoken expiration time configurable
• [ASTERISK-24947] – res_pjsip: Add a PJSIP resolver using core DNS
• [ASTERISK-24960] – Build System: Create MOD_ADD_SOURCE macro for module Makefiles
• [ASTERISK-24965] – cel_pgsql – log_error string references CDR instead of CEL
• [ASTERISK-24974] – Astobj2: Allow reference debugging to be enabled/disabled by config.
• [ASTERISK-24980] – cdr_adaptive_odbc: refactor lines to concatenate of columns name
• [ASTERISK-25040] – pbx: Improve performance of reloads by making hint destruction more performant
• [ASTERISK-25043] – Avoiding ERR_remove_state in OpenSSL
• [ASTERISK-25044] – sorcery: Add ability to insert a new wizard into an object type’s list
• [ASTERISK-25045] – vector: Add new capabilities and unit tests
• [ASTERISK-25049] – CLI: Enable automatic references to modules
• [ASTERISK-25051] – Remove unneeded uses of optional_api providers.
• [ASTERISK-25056] – Modules: Make ast_module_info->self available to auxiliary sources.
• [ASTERISK-25063] – add X.509 subject alternative name support to Asterisk TLS support
• [ASTERISK-25067] – Sorcery Caching: Implement a new caching module
• [ASTERISK-25068] – Move commonly used FreePBX extra sounds to the core set
• [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] – Post AMI VarSet to empty string events when Asterisk deletes a dialplan variable.
• [ASTERISK-25310] – on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED
• [ASTERISK-25405] – CLI: core show fd: add timestamp
• [ASTERISK-25444] – Music On Hold Warning misleading
• [ASTERISK-25471] – Add subscribe_context to res_pjsip
• [ASTERISK-25477] – pjsip show “command” like [criteria]
• [ASTERISK-25495] – Prevent old-update packages on repository Debian systems
• [ASTERISK-25518] – taskprocessor: Add high water mark
• [ASTERISK-25558] – chan_sip option ‘notifyringing’ doc fix and addition of ‘notifyringingprio’
• [ASTERISK-25571] – PJSIP: Add StatsD stats for some common PJSIP objects
• [ASTERISK-25572] – Endpoints: Add StatsD stats for Asterisk endpoints
• [ASTERISK-25578] – SIP/SDP: No rtpmap for static RTP payload IDs
• [ASTERISK-25581] – Add value reason a pause on CLI
• [ASTERISK-25618] – res_pjsip: Check for readability of TLS files at startup
• [ASTERISK-25627] – Easily Preventable Compile Warning
• [ASTERISK-25767] – Add check to configure for sanitizes
• [ASTERISK-25791] – res_pjsip_caller_id: Lack of support for Anonymous <anonymous@anonymous.invalid>
• [ASTERISK-25835] – Authentication using ‘Username’ field from Digest
• [ASTERISK-25846] – Gracefully deal with Absent Stasis Apps
• [ASTERISK-25865] – Message-Account Missing From PJSIP MWI
• [ASTERISK-25930] – PJSIP: disable multi domain to improve realtime performace
• [ASTERISK-25931] – PJSIP: add “reg_server” to contacts.
• [ASTERISK-25965] – res_pjsip_outbound_publish: Allow multiple clients per configuration
• [ASTERISK-25994] – res_pjsip: module load priority
• [ASTERISK-26011] – PJSIP: add “via_addr”, “via_port”, “call_id” to contacts
• [ASTERISK-26088] – Investigate heavy memory utilization by res_pjsip_pubsub
• [ASTERISK-26159] – res_hep: enabled by default and information sent to default address
• [ASTERISK-26190] – SRTP: Enable AES-256 and AES-GCM.
• [ASTERISK-26218] – iLBC 20
• [ASTERISK-26220] – Add support for noreturn function attributes.

New Feature

• [ASTERISK-17899] – Handle crypto lifetime in SDES-SRTP negotiation
• [ASTERISK-22591] – Prevent Asterisk from writing received SMS content in log
• [ASTERISK-23186] – Add usegmtime option to cel_pgsql
• [ASTERISK-23823] – 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] – 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] – 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] – Add field PauseReason on QueueMemberStatus
• [ASTERISK-25549] – Confbridge: Add participant timeout option
• [ASTERISK-25551] – Ability to add channel to an existing bridge by specifying an existing channel prefix
• [ASTERISK-25591] – 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] – 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!