The release of Asterisk 15.0.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release:
Improvements made in this release:
-----------------------------------
| [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup | (Reported by Alexei Gradinari) | |
| Core/BuildSystem: Add defines to fix build with LibreSSL | (Reported by Guido Falsi) | |
| Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file | (Reported by Guido Falsi) | |
| audiohooks: Remove redundant codec translations when using audiohooks | (Reported by Michael Walton) | |
| libsrtp-2.x.x support | (Reported by Alex) | |
| configurable busy_timeout in sqlite backends | (Reported by Marek Cervenka) | |
| res_agi: Set audio format for EAGI audio stream | (Reported by John Fawcett) | |
| Investigate heavy memory utilization by res_pjsip_pubsub | (Reported by Richard Mudgett) | |
| res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip | (Reported by Nir Simionovich (GreenfieldTech - Israel)) | |
| [patch] SIP/SDP: No rtpmap for static RTP payload IDs | (Reported by Alexander Traud) | |
| res_pjsip_session: Add support for overlap dialling | (Reported by Richard Begg) | |
| chan_sip: Add rtcp-mux support | (Reported by Sean Bright) | |
| pbx_spool: OUTGOING_RETRY variable | (Reported by Roman Shubovich) | |
| app_confbridge: 3D-Conferencing via Binaural Synthesis | (Reported by Dennis Guse) | |
| pjsip - Need a command to list active SIP subscriptions | (Reported by Rusty Newton) | |
| app_queue: New service level calculation | (Reported by Sebastian Gutierrez) | |
| Add ability for dialplan show to display filenames/line numbers of registered extensions | (Reported by Jonathan R. Rose) | |
| Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec | (Reported by Badalian Vyacheslav) | |
| [patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel | (Reported by Anthony Messina) | |
| res_calendar_caldav: Add support for gmail | (Reported by Eduardo Scudeller Libardi) | |
| app_controlplayback: Transmit Silence on ControlPlayback pause | (Reported by Mikheili Dautashvili) | |
| TLS support for Solaris, Ming and non-glibc Linux systems | (Reported by Timo Teräs) | |
| cdr_radius: use radcli instead of freeradius-client | (Reported by Tzafrir Cohen) | |
| app_queue: add variable to know if the call is not answered after a queue | (Reported by Sebastian Gutierrez) | |
| chan_sip: Add AccountCode to AMI PeerEntry | (Reported by Sebastian Gutierrez) | |
| [patch] Codec 2 Mode 2400 | (Reported by Alexander Traud) | |
| codec_opus: Add sample to configs/samples/codecs.conf.sample | (Reported by Kevin Harwell) | |
| ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands | (Reported by Matt Jordan) | |
| res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP | (Reported by Michael Walton) | |
| [patch] Force calendars to do new fetch after module reload | (Reported by Ludovic Gasc (Eyepea)) | |
| core: Remove ABI differences of LOW_MEMORY | (Reported by Corey Farrell) | |
| codec_opus: Update Asterisk to support the translation codec. | (Reported by Kevin Harwell) | |
| Announcer channels in ConfBridges cause inefficiencies | (Reported by Mark Michelson) | |
| ARI : Add reason answered_elsewhere to channel hangup | (Reported by Jean Aunis - Prescom) | |
| [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used | (Reported by Alexei Gradinari) | |
| [patch] iLBC 20 | (Reported by Alexander Traud) | |
| [patch] SRTP: Enable AES-256 and AES-GCM. | (Reported by Alexander Traud) | |
| Add support for noreturn function attributes. | (Reported by Corey Farrell) |
Bugs fixed in this release:
-----------------------------------
| Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash | (Reported by Ross Beer) | |
| core: ast_safe_system command injection possible. | (Reported by Corey Farrell) | |
| res_rtp_asterisk: Media can be hijacked even with strict RTP enabled | (Reported by Joshua Colp) | |
| res_rtp_asterisk: Allow remote SSRC to change due to renegotiation | (Reported by Joshua Colp) | |
| core: Don't queue up multiple video update frames. | (Reported by Joshua Colp) | |
| bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU | (Reported by Richard Mudgett) | |
| If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed | (Reported by Seán C. McCord) | |
| manager: hook event is not being raised | (Reported by Kevin Harwell) | |
| res_pjsip_session: Handling of 'msid' is incorrect | (Reported by Kevin Harwell) | |
| bridge: Crash when mapping streams | (Reported by Joshua Colp) | |
| Make --with-pjproject-bundled the default for Asterisk 15 | (Reported by George Joseph) | |
| channel: requester leaks joint_cap on success. | (Reported by Corey Farrell) | |
| Asterisk 15.0.0-Beta1 does not compile | (Reported by Ira Emus) | |
| res_pjsip: parse/add msid attribute when webrtc is enabled | (Reported by Kevin Harwell) | |
| bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. | (Reported by Joshua Colp) | |
| sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5 | (Reported by Corey Farrell) | |
| say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds. | (Reported by Nicolas Riendeau) | |
| bridge_softmix: Don't reorder SFU streams | (Reported by Joshua Colp) | |
| bridge_softmix: Reuse any removed streams for video | (Reported by Joshua Colp) | |
| res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use | (Reported by Joshua Colp) | |
| confbridge: Name recordings are left on filesystem | (Reported by Sergej Kasumovic) | |
| chan_iax2: On reload MWI taskprocessors keep adding up | (Reported by Sergej Kasumovic) | |
| sounds: New 3-D Binaural audio features require new sound prompts | (Reported by Rusty Newton) | |
| French conf-adminmenu, conf-usermenu prompts differ in content from the English files | (Reported by Benoit Duverger) | |
| Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?) | (Reported by Rusty Newton) | |
| res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE | (Reported by Joshua Colp) | |
| res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@' | (Reported by Maxim Vasilev) | |
| res_rtp_asterisk: Deadlock when TURN session in use | (Reported by Jatin Jain) | |
| [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. | (Reported by Alexander Traud) | |
| ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting | (Reported by James Terhune) | |
| channel: ast_waitfordigit_full fails to clear flag in an error branch. | (Reported by Corey Farrell) | |
| PJSIP: Deadlock using TCP transport | (Reported by Richard Mudgett) | |
| Create an StreamEcho dialplan application | (Reported by Kevin Harwell) | |
| chan_pjsip: Add support for multiple streams | (Reported by Joshua Colp) | |
| res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation | (Reported by Joshua Colp) | |
| rtp: Crash in ast_rtp_codecs_payload_code() | (Reported by Ross Beer) | |
| Duplicate logging in queue log for EXITEMPTY events | (Reported by Ove Aursand) | |
| call hangup after leaving app_queue | (Reported by Marek Cervenka) | |
| app_voicemail reloads result in leaked IMAP sockets. | (Reported by Louis Jocelyn Paquet) | |
| core_local: local channel data not being properly unref'ed and unlocked | (Reported by Kevin Harwell) | |
| bridge: stuck channel(s) after failed attended transfer | (Reported by Kevin Harwell) | |
| Comment typo format_g729.c | (Reported by Matthew Fredrickson) | |
| Core/PBX: [patch] Deadlock between dialplan execution and application unregistration | (Reported by Frederic LE FOLL) | |
| res_corosync segfaults at startup with corosync version > 2.x | (Reported by mdu113) | |
| res_ari: Crash when no ari.conf configuration file exists | (Reported by Ronald Raikes) | |
| Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times. | (Reported by Chris Howard) | |
| Seg Fault in ast_sorcery_object_get_id at sorcery.c | (Reported by Ryan Smith) | |
| nat/external_media settings ignored in 14.4.1 | (Reported by Christopher van de Sande) | |
| res_rtp_asterisk: Incorrect SSRC change for RTCP component | (Reported by Michael Walton) | |
| bridging: T.38 request is lost when channels are added to bridge | (Reported by Torrey Searle) | |
| res_pjsip_refer/session: Calls dropped during transfer | (Reported by Kevin Harwell) | |
| Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network | (Reported by alex) | |
| res_pjsip_transport_websocket: segfault in get_write_timeout | (Reported by Jørgen H) | |
| chan_pjsip: Device state is idle when channel from endpoint is in early media | (Reported by Joshua Colp) | |
| chan_pjsip: Flipping between codecs | (Reported by Michael Maier) | |
| chan_pjsip would send INVITE to 'Unreachable' endpoints | (Reported by Jacek Konieczny) | |
| bridge: Crash when freeing frame and snooping | (Reported by Michel R. Vaillancourt) | |
| Background in realtime | (Reported by Andrew Nowrot) | |
| channel / meetme: Fix missing parentheses | (Reported by Joshua Colp) | |
| GET /recordings/stored returns 500 Internal Server Error | (Reported by Tim Morgan) | |
| [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec | (Reported by Frankie Chin) | |
| Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded | (Reported by Tzafrir Cohen) | |
| srtp's crypto_get_random deprecated | (Reported by Tzafrir Cohen) | |
| AGI - RECORD FILE - documentation doesn't describe BEEP argument | (Reported by Rusty Newton) | |
| Async AGI crashes Asterisk when issuing "set variable" command without args | (Reported by Antoine Pitrou) | |
| Malformed AGI 520 Usage response | (Reported by Tony Mountifield) | |
| res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space | (Reported by John Harris) | |
| app_queue: Agent not called when caller is parked | (Reported by wushumasters) | |
| app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime | (Reported by Etienne Lessard) | |
| app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel | (Reported by David Brillert) | |
| app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call | (Reported by Lorne Gaetz) | |
| app_confbridge: ConfBridge sometimes does not play user name recording while leaving | (Reported by Robert Mordec) | |
| res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 | (Reported by Javier Riveros ) | |
| chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable | (Reported by Stefan Engström) | |
| Out of bound memory access in PJSIP multipart parser crashes Asterisk | (Reported by Sandro Gauci) | |
| Asterisk Skinny memory exhaustion vulnerability leads to DoS | (Reported by Sandro Gauci) | |
| Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP | (Reported by Sandro Gauci) | |
| Audit manipulation of channel flags without locks | (Reported by Joshua Colp) | |
| res_pjsip_session: INVITE retransmissions could still setup the same call again. | (Reported by Richard Mudgett) | |
| res_rtp_asterisk: One way audio when transcoding | (Reported by Henning Holtschneider) | |
| Problems with Blind Transfer, PJSIP (Aastra 6869i) | (Reported by Matthias Binder) | |
| tcptls: Incorrect OpenSSL function call leads to misleading error report | (Reported by Bob Ham) | |
| Crash in Manager Reload when TLS Config Changes | (Reported by Joshua Elson) | |
| [patch]cel_odbc sometimes inserts CEL with wrong eventtime | (Reported by Etienne Lessard) | |
| func_cdr: CDR function does not permit empty values to be assigned | (Reported by gkloepfer) | |
| [patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages. | (Reported by Frederic LE FOLL) | |
| Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used | (Reported by Corey Farrell) | |
| bridge_simple: Add support for streams | (Reported by Kevin Harwell) | |
| res_pjsip: Deadlock in T.38 framehook | (Reported by Richard Mudgett) | |
| res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked. | (Reported by Richard Mudgett) | |
| dial: Allow topology of dialing channel to influence dialed channel | (Reported by Joshua Colp) | |
| SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory. | (Reported by Andreas Krüger) | |
| func_speex: Crash caused by frame with no datalen | (Reported by Richard Kenner) | |
| res_pjsip_session: Wrong From on reinvite when request and To URI differ | (Reported by Yasin CANER) | |
| pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux | (Reported by abelbeck) | |
| chan_sip: tcpbind uses wrong source address | (Reported by Ksenia) | |
| pjsip: Add database tables for RLS | (Reported by Joshua Colp) | |
| sdp: Implement T.38 | (Reported by Joshua Colp) | |
| Asterisk crash if hep.conf have some missing parameters | (Reported by Joel Vandal) | |
| STUN server with non-default-route transport causes INVITE delay | (Reported by George Joseph) | |
| chan_sip: ACK with SDP does not update a direct media bridge | (Reported by Jean Aunis - Prescom) | |
| res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) | (Reported by Sebastian Gutierrez) | |
| res_rtp_asterisk: Crash when freeing RTCP address string | (Reported by Niklas Larsson) | |
| res_rtp_asterisk: Crash in pjnath when receiving packet | (Reported by Adagio) | |
| format_wav: wav16 format read file only by 320 - half of frame | (Reported by Vitaly K) | |
| format_ogg_vorbis: Memory leak using OGG in MixMonitor | (Reported by Ivan Myalkin) | |
| STUN never works when asterisk started without internet access | (Reported by Jeremy Kister) | |
| Audible clicks when playing sox encoded au file with STREAM FILE AGI command | (Reported by Roman S.) | |
| [UBSAN] strings.h:signed integer overflow in ast_str_case_hash | (Reported by Badalian Vyacheslav) | |
| res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport | (Reported by Richard Begg) | |
| Listening TCP/TLS sockets stop when temporarily out of open files | (Reported by Walter Doekes) | |
| pjsip: Add database tables for PUBLISH support | (Reported by Joshua Colp) | |
| pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). | (Reported by Alexander Traud) | |
| pjproject_bundled: Merge 3 upstream deadlock patches into bundled | (Reported by Ross Beer) | |
| app_queue: PAUSEALL/UNPAUSEALL does not log reason | (Reported by Troy Bowman) | |
| chan_sip: Security vulnerability with client code header | (Reported by Alex VillacÃs Lasso) | |
| Unused realtime MOH classes not purged on 'moh reload' | (Reported by Sébastien Couture) | |
| res_pjsip: Excessive refcount reached on transport ao2 object | (Reported by Ross Beer) | |
| SIP Failed to parse multiple Supported: headers | (Reported by Olle Johansson) | |
| chan_sip: Session Timers required but refused wrongly. | (Reported by Alexander Traud) | |
| res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code | (Reported by Yaacov Akiba Slama) | |
| Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT | (Reported by twisted) | |
| libasteriskssl.so not found when asterisk is installed for the 1st time | (Reported by George Joseph) | |
| sdp: Add support for connection address management and topology updating | (Reported by Joshua Colp) | |
| xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client | (Reported by Marcello Ceschia) | |
| [patch]SDP crypto tag is validated incorrectly | (Reported by Joerg Sonnenberger) | |
| channel: Support dynamic number of file descriptors | (Reported by Joshua Colp) | |
| res_musiconhold: format option is not documented adequately | (Reported by Jens Bürger) | |
| No core dumps because of res_musiconhold chdir. | (Reported by Walter Doekes) | |
| xmpp: starttls problem causes connection spew | (Reported by Matthias Urlichs) | |
| pjproject_bundled build fails to download pjproject source when using cURL | (Reported by Gergely Dömsödi) | |
| JABBER_STATUS fails with improper code 7 for unavailable clients | (Reported by Anthony Critelli) | |
| Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available | (Reported by Jeremy Kister) | |
| WARNING for "JABBER: socket read error" should be more specific | (Reported by Sean Darcy) | |
| rtp_engine: Allocate RTP payloads on a per-session basis | (Reported by Joshua Colp) | |
| cdr: Problem setting variables in h exten | (Reported by Sebastian Gutierrez) | |
| res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets | (Reported by Max Norba) | |
| res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument. | (Reported by Vinod Dharashive) | |
| res_pjsip_pubsub: Crash when generating xpidf content | (Reported by Andrew Green) | |
| Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled | (Reported by Kirsty Tyerman) | |
| app_mixmonitor: Recording out of sync when 183 but no RTP | (Reported by Aaron An) | |
| app_queue: Queue stops calling members with local interface after forwarding in previous call | (Reported by Robert Mordec) | |
| res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome | (Reported by Dan Jenkins) | |
| PJSIP external_media_address ignored if no local_net options are provided | (Reported by Matt Jordan) | |
| autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade). | (Reported by Krzysztof Trempala) | |
| res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension | (Reported by Torrey Searle) | |
| core: Malformed pattern matching extension (various factors) results in crash | (Reported by xrobau) | |
| chan_iax2: Reload of iax peer results in loss of host address/port | (Reported by Richard Begg) | |
| Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal) | (Reported by Matt Jordan) | |
| Document the fact that Asterisk HEP support only works with the PJSIP channel driver | (Reported by Olivier Krief) | |
| Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk | (Reported by Roman Bedros) | |
| stasis_cache.c:845 caching_topic_exec: - misleading ERROR message | (Reported by Smirnov Aleksey) | |
| chan_pjsip: Dialplan function race condition | (Reported by Joshua Colp) | |
| pjsip/cli_commands: pjsip show channelstats shows wrong codec | (Reported by Kevin Harwell) | |
| res_musiconhold: musiconhold seems to think that the general section is a class and issues warning | (Reported by Jonathan Harris) | |
| res_pjsip: Crash when using IPv6 and Transport ws,wss | (Reported by Michael Balen) | |
| app_voicemail: Cannot set fromstring on a per-mailbox basis | (Reported by Mark Scholten) | |
| Websocket becomes disconnected when trying to place call from browser | (Reported by Mark Michelson) | |
| chan_sip: Call not cancelled after receiving a 422 response | (Reported by Jean Aunis - Prescom) | |
| core: Implement stream topology changing in channels | (Reported by Joshua Colp) | |
| Saynumber is trying to get "and" from "digits/" subfolder | (Reported by Jonathan Harris) | |
| Long lines in call files cause spurious syntax error | (Reported by Dave Olszewski) | |
| res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS' | (Reported by Jørgen H) | |
| Implement ast_read_stream in channels | (Reported by Joshua Colp) | |
| res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging | (Reported by Dmitry Wagin) | |
| core: Playback URL fails after some time | (Reported by Igor Gamayunov) | |
| pjsip.conf.sample: user_agent: still refers to branch 12 | (Reported by Tzafrir Cohen) | |
| PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist | (Reported by Mark Michelson) | |
| res_pjsip: Crash when calling PJSIPShowEndpoint | (Reported by Jørgen H) | |
| res_pjsip_outbound_registration doesn't know about network change events | (Reported by George Joseph) | |
| bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio | (Reported by Sean Bright) | |
| res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication | (Reported by Peter Sokolov) | |
| Implement ast_write_stream in channels | (Reported by George Joseph) | |
| [patch] Fix download_externals To Allow The Use Of curl Or wget | (Reported by Michael L. Young) | |
| Pattern matching with res_config_mysql extensions does not behave as expected | (Reported by Charlie Smurthwaite) | |
| stream: Add streams to "core show channel" | (Reported by Joshua Colp) | |
| [patch] DUNDi weight parameter not processed correctly | (Reported by Peter Racz) | |
| res_pjsip: Using an auth object for inbound and outbound authentication fails. | (Reported by Richard Mudgett) | |
| PJSIP Segfault 13.13.1 (Bundled PJSIP) | (Reported by Nic Colledge) | |
| Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c | (Reported by Michael Maier) | |
| Function vmauthenticate accesses uninitialized memory | (Reported by Filip Jenicek) | |
| [patch] Error during LDAP modify action when user unregisters | (Reported by Nicholas John Koch) | |
| [patch] Integrity Check Of PJSIP Download Fails | (Reported by Michael L. Young) | |
| [patch] Fix query with double backslash in string literals and stop log warnings | (Reported by Humberto Figuera) | |
| res_config_sqlite3 uses incorrect query - unnecessary escape | (Reported by Stepan) | |
| SQlite3: Realtime queue loading fails after PRAGMA query result | (Reported by Scott Griepentrog) | |
| http: Crash on Reload Only in ast_tcptls_server_start | (Reported by Joshua Elson) | |
| Phone default have not ringing on ARM | (Reported by Igor Goncharovsky) | |
| pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh | (Reported by Zach R) | |
| res_pjsip_mwi: Asterisk does not terminate MWI subscription | (Reported by Carl Fortin) | |
| Implement stream topology (non-change request) API usage in channels | (Reported by George Joseph) | |
| VoiceMailPlayMsg not playing messages via realtime | (Reported by Ryan Rittgarn) | |
| [patch] 'Silence' is truncated in Record() | (Reported by var) | |
| app_queue: reset abandoned in service level | (Reported by Sebastian Gutierrez) | |
| Implement ast_stream_topology API | (Reported by George Joseph) | |
| chan_pjsip: Error when calling PJSIP client with domain specified | (Reported by Norbert Varga) | |
| core: Protect flags during ast_waitfor | (Reported by Joshua Colp) | |
| pbx: AMI Originate ignore "failed" extension on call failure | (Reported by Nasir Iqbal) | |
| stream: Add basic API | (Reported by Joshua Colp) | |
| configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample | (Reported by Torrey Searle) | |
| Crash in srv.c on startup with pjsip | (Reported by nappsoft) | |
| res_stasis_device_state: Duplicate subscriptions when multiple received at same time | (Reported by Joshua Colp) | |
| ARI channelvars cause memory leak | (Reported by Sébastien Duthil) | |
| ari: Channels with pre-dial handlers cannot be hung up via ARI | (Reported by Tom Pawelek) | |
| core: Possibility of a frame "imbalance" leading to stuck channels. | (Reported by Mark Michelson) | |
| res_agi: run_agi eats frames it shouldn't | (Reported by George Joseph) | |
| ASTERISK-25951 causes issues for callerid manipulation through agi | (Reported by Morten Tryfoss) | |
| res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'. | (Reported by Anthony Messina) | |
| res_resolver_unbound: FRACK! Excessive ref count trap tripped. | (Reported by Richard Mudgett) | |
| MixMonitorMute mutes through stream if already slinear (e.g. Originate) | (Reported by David Woolley) | |
| Crash on invalid contact domain (pjsip aor) | (Reported by Dmitriy) | |
| res_pjsip: Assertion when sending OPTIONS request to endpoint | (Reported by Ross Beer) | |
| build_tools: make_build_h does not handle \ in user name | (Reported by Kirill Katsnelson) | |
| app_queue: Random queues disappear on "core reload queue all" | (Reported by Kirill Katsnelson) | |
| res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect | (Reported by Michael Maier) | |
| res_pjsip_endpoint_identifier_ip: Add support for SRV | (Reported by Joshua Colp) | |
| PJPROJECT: Detecting compiled max log level does not work. | (Reported by Richard Mudgett) | |
| res_sorcery_memory_cache: memory leak on every sorcery memory cache populate | (Reported by Ustinov Artem) | |
| voicemail API test: confuses expected and actual values | (Reported by Tzafrir Cohen) | |
| voicemail API test: uses varlibdir instead of datadir for a sound file | (Reported by Tzafrir Cohen) | |
| app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER? | (Reported by Marek Cervenka) | |
| [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 | (Reported by Aaron An) | |
| Crash when setting remote address on RTP instance | (Reported by Richard Mudgett) | |
| [patch] Outgoing SIP-URI Dialing via PJSIP | (Reported by Alexander Traud) | |
| Remember SDP negotiation on SIP_CODEC_INBOUND. | (Reported by Alexander Traud) | |
| chan_pjsip: Crash when using CHANNEL dialplan function around masquerade | (Reported by Joshua Colp) | |
| res_pjsip: Various issues with compact SIP headers | (Reported by Joshua Elson) | |
| res_calendar: Calendars duplicated after module reload | (Reported by Martin Tomec) | |
| [patch]pjsip: Transfers Broken with Compact Headers Enabled | (Reported by JoshE) | |
| app_queue: Queue application does not ring members with Local interface | (Reported by Jonas Kellens) | |
| chan_sip: Segfaults upon reload if client with MWI wasn't registered | (Reported by Michael Kuron) | |
| build: GCC 5.1.x catches some new const, array bounds and missing paren issues | (Reported by George Joseph) | |
| Need more explicit debug when PJSIP dialstring is invalid | (Reported by Rusty Newton) | |
| Message.c: Message channel becomes saturated with frames leading to spammy log messages | (Reported by Jonathan Rose) | |
| pjproject_bundled doesn't verify already downloaded tarballs | (Reported by George Joseph) | |
| chan_sip: Allows To-tag checks to be bypassed, setting up new calls | (Reported by Walter Doekes) | |
| codec_opus: Recursiveness when parsing fmtp line | (Reported by Jørgen H) | |
| PJSIPShowRegistrationsInbound just dumps all aors | (Reported by George Joseph) | |
| Support older DNS style for OpenBSD | (Reported by snuffy) | |
| res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_" | (Reported by Juris Breicis) | |
| tests/manager: 4 test failures as a result of iostream change | (Reported by Joshua Colp) | |
| Asterisk fails building with OpenSSL 1.1.0 | (Reported by Tzafrir Cohen) | |
| res_rtp_asterisk: Can't bind on systems without IPv6 | (Reported by Guido Falsi) | |
| [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no | (Reported by Alexei Gradinari) | |
| Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 | (Reported by Marek Cervenka) | |
| res_rtp_asterisk: RTT miscalculation in RTCP | (Reported by Hector Royo Concepcion) | |
| chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. | (Reported by Michael Kuron) | |
| Compile and link failures on OpenBSD | (Reported by snuffy) | |
| codec_opus: Generated fmtp line has no content | (Reported by Sebastian Gutierrez) | |
| codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. | (Reported by Richard Mudgett) | |
| pjsip: Memory corruption with possible memory leak. | (Reported by Richard Mudgett) | |
| Unconditional use of fopencookie() / funopen() is non-portable | (Reported by Timo Teräs) | |
| manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes | (Reported by Michelle Dupuis) | |
| Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage | (Reported by George Joseph) | |
| testsuite: Need to check PJSIP functionality when res_srtp is not loaded. | (Reported by Joshua Colp) | |
| chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set | (Reported by Jason) | |
| Some typos in documentation of chan_sip.c | (Reported by C.J. Collier) | |
| res_pjsip: Resolution incorrect when explicit IPv6 transport configured | (Reported by Joshua Colp) | |
| ari: Bridge events stop working after this sequence of ARI calls | (Reported by Daniele Pallastrelli) | |
| ooh323 sends wrong hangup code | (Reported by Dmitry Melekhov) | |
| Multi-party Video: Fix some post Asterisk-11 regressions | (Reported by Matt Jordan) | |
| build: Prepare for gcc 6.2 | (Reported by George Joseph) | |
| A few non-critical deprecation warnings when building on Ubuntu 16.10 | (Reported by Jonathan Harris) | |
| chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression | (Reported by Michael Keuter) | |
| app_dial: When PickupChan() is used some channels may have incorrect device state | (Reported by Joshua Colp) | |
| [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used | (Reported by Frankie Chin) | |
| [patch] rtp_engine: Allow more than 32 dynamic payload types. | (Reported by Alexander Traud) | |
| mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2' | (Reported by Tzafrir Cohen) | |
| res_pjsip_sdp_rtp: Restrict number of formats to maximum | (Reported by Joshua Colp) | |
| chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" | (Reported by Sergey Grachev) | |
| Fix FTBFS on Hurd | (Reported by Gabriele Giacone) | |
| AMI: NewConnectedLine event is not documented | (Reported by Etienne Lessard) | |
| [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy | (Reported by Badalian Vyacheslav) | |
| astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled. | (Reported by Corey Farrell) | |
| Asterisk 13.11.0 + PJSIP crash | (Reported by Ian Gilmour) | |
| Asterisk segfaults shortly after starting even with no active calls. | (Reported by Harley Peters) | |
| [patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c | (Reported by Matt Krokosz) | |
| tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance | (Reported by Joshua Colp) | |
| Super Awesome Company: Don't specify transport in pjsip.conf | (Reported by Rusty Newton) | |
| pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions | (Reported by George Joseph) | |
| Embedded pjproject: build.mak contains hardcoded full path to version.mak | (Reported by Matt Jordan) | |
| [patch] CLI: core set debug: Auto-completes File not Module | (Reported by Alexander Traud) | |
| res_pjsip_caller_id: Crash on outgoing change | (Reported by Bill Brigden) | |
| app_voicemail: Asterisk crashes when MailboxExists is used | (Reported by Doug Lytle) | |
| res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness | (Reported by Andreas Wetzel) | |
| [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations. | (Reported by Alexander Traud) | |
| [patch] chan_pjsip: segfault on already disconnected session | (Reported by Alexei Gradinari) | |
| cdr_radius / cel_radius: try fix memory leak | (Reported by Badalian Vyacheslav) | |
| Segmentation Fault with ARI originate into mixing bridge with 43 clients | (Reported by Andrew Nagy) | |
| 'features show' command in CLI does not return prompt. | (Reported by John Kiniston) | |
| menuselect: invalid test for GTK2 | (Reported by Tzafrir Cohen) | |
| pjproject: SEGV during SSL operations | (Reported by George Joseph) | |
| [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage | (Reported by Leandro Dardini) | |
| chan_rtp: Crash when originating | (Reported by Kayode) | |
| [patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio. | (Reported by effie mouzeli) | |
| pjproject-bundled: configure fails to check for all required utilities | (Reported by Corey Farrell) | |
| core: Be forgiving on external callerid that may be flawed so we don't drop events | (Reported by Richard Mudgett) | |
| res_config_mysql: Broken after 13.10 | (Reported by Carlos Chavez) | |
| app_dial: There's no way to override the hangupcause on unanswered channels | (Reported by George Joseph) | |
| [patch] force_rport,auto_comedia: No NAT detection triggered. | (Reported by Alexander Traud) | |
| res_pjsip_config_wizard: Memory leak in module_unload | (Reported by Badalian Vyacheslav) | |
| core: Asterisk 14 doesn't show the header in the console or verbose when starting | (Reported by Dan Jenkins) | |
| Populating database via Alembic fails when using same database for multiple schema sets | (Reported by Dafi Ni) | |
| [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. | (Reported by Alexander Traud) | |
| app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members | (Reported by Etienne Lessard) | |
| format_ogg_opus: remove from source | (Reported by Kevin Harwell) | |
| Broken REGISTER sent to IPv4 server when bindaddr=[::] | (Reported by Jacek) | |
| Deadlock in chan_sip - core show locks shows do_monitor lock | (Reported by Barry Flanagan) | |
| manager: PresenceState action crashes Asterisk 14 | (Reported by Andrew Nagy) | |
| res_odbc: Clean up pooling options | (Reported by Joshua Colp) | |
| core: Won't compile when LOW_MEMORY is enabled | (Reported by Anthony Messina) | |
| Consoles do not display verbose logger messages even when requested. | (Reported by Marcelo Terres) | |
| Astcanary dies when doing "core restart" | (Reported by Walter Doekes) | |
| asterisk fails to lower its priority when astcanary dies | (Reported by Xavier Hienne) | |
| SQL error when using realtime and registering extension / inserting into ps_contacts | (Reported by Jeppe Ryskov Larsen) | |
| rtp: Offer with multiple payloads for same codec is incorrectly handled | (Reported by Joshua Colp) | |
| res_pjsip_multihomed: Contact port is rewritten for connectionful protocols | (Reported by Joshua Colp) | |
| [patch] cdr_mysql: fails to use UTC if so instructed | (Reported by Tzafrir Cohen) | |
| rtp: Timestamps broken when video frame is across multiple RTP packets | (Reported by Joshua Colp) | |
| res_pjsip_transport_management: Log message states seconds, but time value is milliseconds | (Reported by Joshua Colp) | |
| TCP Session-Timers not dropping call | (Reported by Aaron Hamstra) | |
| res_pjsip: Don't assume a request will have target addresses | (Reported by Joshua Colp) | |
| app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. | (Reported by Richard Mudgett) | |
| chan_sip: Contact is updated on re-200, but not on re-INVITE | (Reported by Walter Doekes) | |
| res_pjsip_callerid: Irregular URI causes unexpected callerid | (Reported by Kevin Harwell) | |
| 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed | (Reported by Dmitry Melekhov) | |
| res_pjsip_session: Add ability to use preferred codec only | (Reported by Aaron An) | |
| res_pjsip: Crash when applying ACL from non-existent endpoint | (Reported by nappsoft) | |
| chan_sip: File descriptors leak (UDP sockets) | (Reported by Etienne Lessard) | |
| SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!" | (Reported by tootai) | |
| ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list | (Reported by Matt Jordan) | |
| res_pjsip_session: segfault on already disconnected session | (Reported by Alexei Gradinari) | |
| [patch]SDP offer/answer fails if crypto keys added to non-crypto offer | (Reported by Olle Johansson) | |
| Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up. | (Reported by Etienne Lessard) | |
| Crash on â??core show channeltype Surrogateâ?? in ast_format_cap_get_names | (Reported by CGI.NET) | |
| app_mp3: results in timeout for streams | (Reported by Jens Bürger) | |
| res_pjsip: Wrong state for aors without registered contacts after startup | (Reported by nappsoft) | |
| pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts | (Reported by Etienne Lessard) | |
| app_queue: Queue application sometimes stops calling members with Local interface | (Reported by Etienne Lessard) | |
| pjproject-bundled: Fails to compile on Debian 6 | (Reported by George Joseph) | |
| channel: Hang-up crashes, chan_pjsip not cleaning up properly | (Reported by Alexander Traud) | |
| res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels | (Reported by Etienne Lessard) | |
| Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code | (Reported by David Brillert) | |
| Deadlock potential in res_fax and CCSS with local channels. | (Reported by Richard Mudgett) | |
| AEL: macro-call in Dial application, macro "lacks 's' extension" | (Reported by chris de rock) | |
| [patch] Plaintext auth is still supported in IAX2 | (Reported by Eugene) | |
| Finish mapping the sip.conf parameters to res_sip.conf parameters | (Reported by Matt Jordan) | |
| [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) | (Reported by abelbeck) | |
| res_pjsip_sdp_rtp: G729A does not include annexb=no attribute. | (Reported by Ali Ghavidel) | |
| Swagger scripts are not replacing format variable in file brief | (Reported by Corey Farrell) | |
| res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it | (Reported by József Dudás) | |
| Asterisk 14: Two resolver unbound testsuite tests fail | (Reported by Richard Mudgett) | |
| followme: fails to reset config items to default values on reload | (Reported by Tzafrir Cohen) | |
| [patch] BuildSystem: ca_list_path capabilities not detected in PJProject. | (Reported by Alexander Traud) | |
| ARI: Path parameters are case sensitive | (Reported by Joshua Colp) | |
| XMPP no longer triggers NOTIFY to device via chan_pjsip | (Reported by Ross Beer) | |
| pbx: Failure to remove inconsistent extension names | (Reported by Corey Farrell) | |
| Security: Privilege escalation by AMI adding dialplan extensions. | (Reported by Richard Mudgett) | |
| ast_register_atexit callbacks should be run on failed startup. | (Reported by Corey Farrell) | |
| res_pjsip: When using compact headers, rpid and pai are incorrectly generated | (Reported by George Joseph) | |
| app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension | (Reported by Etienne Lessard) | |
| res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname | (Reported by George Joseph) | |
| res_pjsip: Empty global default_from_user causes crash | (Reported by Joshua Colp) | |
| alembic: 'auth_username' not in PJSIP 'identify_by' enum | (Reported by Joshua Colp) | |
| sdp_srtp: libsrtp now a required dependency, shouldn't be | (Reported by Ben Merrills) | |
| pjsip: Deadlock with suspend + masquerade + indicate | (Reported by Ross Beer) | |
| alembic: error when using sqlalchemy version 1.1.0b2 | (Reported by Kevin Harwell) | |
| res_resolver_unbound: fails configure on older Ubuntu and CentOS | (Reported by George Joseph) | |
| DNS lookups can block channel media paths | (Reported by Mark Michelson) | |
| asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM. | (Reported by Corey Farrell) | |
| [patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c | (Reported by Richard Mudgett) | |
| Errors ignored from some parts of system initialization. | (Reported by Corey Farrell) | |
| [patch] res_pjsip: Use more compatible regex for get all | (Reported by Dmitry Wagin) | |
| [patch] SIP/SDP origin (o=) contains brackets with IP6 | (Reported by Alexander Traud) | |
| Remove "live_dangerously" requirement on DB(read) | (Reported by Andrew Nagy) | |
| pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..." | (Reported by Hans van Eijsden) | |
| Fax is detected on regular calls. | (Reported by Richard Mudgett) | |
| sqlalchemy error due to long identifier name | (Reported by Mark Michelson) | |
| asterisk leaves zombie mpg123 | (Reported by dcarr) | |
| [patch] Deadlock between 'sip show channels' command and attended transfer handling | (Reported by Ben Smithurst) | |
| PJSIP: tx_data_destroy called twice | (Reported by Scott Griepentrog) | |
| res_pjsip_pubsub: Crash when decrementing reference count of message | (Reported by Ross Beer) | |
| res_pjsip: Crash when freeing cloned message in distributor | (Reported by Ross Beer) | |
| res_fax: Deadlock when detect fax while channel executing Playback | (Reported by Richard Mudgett) | |
| Allow arbitrary time for fax detection to end on a channel | (Reported by Richard Mudgett) |
New Features made in this release:
-----------------------------------
| Add support for systemd socket activation | (Reported by Corey Farrell) | |
| core: Add support for timelen parsing to ast_parse_arg and ACO. | (Reported by Corey Farrell) | |
| ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait. | (Reported by Corey Farrell) | |
| Add QUEUE_FLOAT_PENALTY to app_queue | (Reported by Steve Davies) | |
| func_channel: Add ability to get the callid so dialplan has access to it. | (Reported by Richard Mudgett) | |
| res_pjsip: Add endpoint identification scheme based on a configured SIP header/value | (Reported by Matt Jordan) | |
| [patch] Allow "Comedian Mail" branding to be removed | (Reported by John Covert) | |
| [patch] RTCP feedback for codec modules | (Reported by Lorenzo Miniero) | |
| app_queue: Update Data of Queues (use queues as outbound calls container) | (Reported by Sebastian Gutierrez) | |
| Make logging PJPROJECT messages a bit easier | (Reported by Richard Mudgett) | |
| app_originate: Add option to execute gosub prior to dial | (Reported by dkerr) | |
| ARI: Add the ability to control the source of video in a multi-party mixing bridge | (Reported by Matt Jordan) | |
| ARI: Add ability to specify channel variables on websocket events | (Reported by Mark Michelson) | |
| ARI: Add an 'asterisk_id' field to outgoing events | (Reported by Matt Jordan) | |
| Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment | (Reported by Matt Jordan) |
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
Thank you for your continued support of Asterisk!
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