The Asterisk Development Team would like to announce the first beta of Asterisk 15.0.0.
This beta is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 15.0.0-beta1 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 beta:
Improvements made in this release:
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res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup (Reported by Alexei Gradinari) |
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Core/BuildSystem: Add defines to fix build with LibreSSL (Reported by Guido Falsi) |
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Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file (Reported by Guido Falsi) |
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audiohooks: Remove redundant codec translations when using audiohooks (Reported by Michael Walton) |
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libsrtp-2.x.x support (Reported by Alex) |
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configurable busy_timeout in sqlite backends (Reported by Marek Cervenka) |
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res_agi: Set audio format for EAGI audio stream (Reported by John Fawcett) |
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Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) |
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res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech – Israel)) |
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SIP/SDP: No rtpmap for static RTP payload IDs (Reported by Alexander Traud) |
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res_pjsip_session: Add support for overlap dialling (Reported by Richard Begg) |
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chan_sip: Add rtcp-mux support (Reported by Sean Bright) |
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pbx_spool: OUTGOING_RETRY variable (Reported by Roman Shubovich) |
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app_confbridge: 3D-Conferencing via Binaural Synthesis (Reported by Dennis Guse) |
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pjsip – Need a command to list active SIP subscriptions (Reported by Rusty Newton) |
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app_queue: New service level calculation (Reported by scgm11) |
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Add ability for dialplan show to display filenames/line numbers of registered extensions (Reported by Jonathan R. Rose) |
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Testsuite: increase timeout to check “core fullybooted wait” up to 30 sec (Reported by Badalian Vyacheslav) |
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Asterisk app_originate doesn’t allow setting Caller*ID on the originating channel (Reported by Anthony Messina) |
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res_calendar_caldav: Add support for gmail (Reported by Eduardo Scudeller Libardi) |
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app_controlplayback: Transmit Silence on ControlPlayback pause (Reported by Mikheili Dautashvili) |
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TLS support for Solaris, Ming and non-glibc Linux systems (Reported by Timo Teräs) |
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cdr_radius: use radcli instead of freeradius-client (Reported by Tzafrir Cohen) |
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app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11) |
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chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11) |
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Codec 2 Mode 2400 (Reported by Alexander Traud) |
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codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell) |
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ARI: Add ‘ari show app’, ‘ari show apps’, and ‘ari set debug’ CLI commands (Reported by Matt Jordan) |
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res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP (Reported by Michael Walton) |
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Force calendars to do new fetch after module reload (Reported by Ludovic Gasc (Eyepea)) |
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core: Remove ABI differences of LOW_MEMORY (Reported by Corey Farrell) |
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codec_opus: Update Asterisk to support the translation codec. (Reported by Kevin Harwell) |
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Announcer channels in ConfBridges cause inefficiencies (Reported by Mark Michelson) |
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ARI : Add reason answered_elsewhere to channel hangup (Reported by Jean Aunis – Prescom) |
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res_fax: set FAXMODE variable to let dialplan know what fax transport was used (Reported by Alexei Gradinari) |
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iLBC 20 (Reported by Alexander Traud) |
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SRTP: Enable AES-256 and AES-GCM. (Reported by Alexander Traud) |
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Add support for noreturn function attributes. (Reported by Corey Farrell) |
Bugs fixed in this release:
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bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. (Reported by Joshua Colp) |
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sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5 (Reported by Corey Farrell) |
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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) |
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bridge_softmix: Don’t reorder SFU streams (Reported by Joshua Colp) |
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bridge_softmix: Reuse any removed streams for video (Reported by Joshua Colp) |
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res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use (Reported by Joshua Colp) |
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confbridge: Name recordings are left on filesystem (Reported by Sergej Kasumovic) |
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chan_iax2: On reload MWI taskprocessors keep adding up (Reported by Sergej Kasumovic) |
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sounds: New 3-D Binaural audio features require new sound prompts (Reported by Rusty Newton) |
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French conf-adminmenu, conf-usermenu prompts differ in content from the English files (Reported by Benoit Duverger) |
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Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?) (Reported by Rusty Newton) |
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res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE (Reported by Joshua Colp) |
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res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing ‘@’ (Reported by Maxim Vasilev) |
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res_rtp_asterisk: Deadlock when TURN session in use (Reported by Jatin Jain) |
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autodomain (SIP Domain Support): Add only really different domain with TLS. (Reported by Alexander Traud) |
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ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting (Reported by James Terhune) |
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channel: ast_waitfordigit_full fails to clear flag in an error branch. (Reported by Corey Farrell) |
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PJSIP: Deadlock using TCP transport (Reported by Richard Mudgett) |
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Create an StreamEcho dialplan application (Reported by Kevin Harwell) |
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chan_pjsip: Add support for multiple streams (Reported by Joshua Colp) |
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res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation (Reported by Joshua Colp) |
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rtp: Crash in ast_rtp_codecs_payload_code() (Reported by Ross Beer) |
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Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) |
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call hangup after leaving app_queue (Reported by Marek Cervenka) |
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app_voicemail reloads result in leaked IMAP sockets. (Reported by Louis Jocelyn Paquet) |
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core_local: local channel data not being properly unref’ed and unlocked (Reported by Kevin Harwell) |
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bridge: stuck channel(s) after failed attended transfer (Reported by Kevin Harwell) |
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Comment typo format_g729.c (Reported by Matthew Fredrickson) |
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Core/PBX: Deadlock between dialplan execution and application unregistration (Reported by Frederic LE FOLL) |
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res_corosync segfaults at startup with corosync version > 2.x (Reported by mdu113) |
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res_ari: Crash when no ari.conf configuration file exists (Reported by Ronald Raikes) |
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Crash occurs when a channel in a ‘mixing,dtmf_events’ bridge is muted multiple times. (Reported by Chris Howard) |
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Seg Fault in ast_sorcery_object_get_id at sorcery.c (Reported by Ryan Smith) |
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nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) |
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res_rtp_asterisk: Incorrect SSRC change for RTCP component (Reported by Michael Walton) |
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bridging: T.38 request is lost when channels are added to bridge (Reported by Torrey Searle) |
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res_pjsip_refer/session: Calls dropped during transfer (Reported by Kevin Harwell) |
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Asterisk build process fails with flag –with-pjproject-bundled with curl download command and slow network (Reported by alex) |
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res_pjsip_transport_websocket: segfault in get_write_timeout (Reported by Jørgen H) |
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chan_pjsip: Device state is idle when channel from endpoint is in early media (Reported by Joshua Colp) |
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chan_pjsip: Flipping between codecs (Reported by Michael Maier) |
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chan_pjsip would send INVITE to ‘Unreachable’ endpoints (Reported by Jacek Konieczny) |
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bridge: Crash when freeing frame and snooping (Reported by Michel R. Vaillancourt) |
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Background in realtime (Reported by Andrew Nowrot) |
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channel / meetme: Fix missing parentheses (Reported by Joshua Colp) |
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GET /recordings/stored returns 500 Internal Server Error (Reported by Tim Morgan) |
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Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) |
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Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded (Reported by Tzafrir Cohen) |
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srtp’s crypto_get_random deprecated (Reported by Tzafrir Cohen) |
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AGI – RECORD FILE – documentation doesn’t describe BEEP argument (Reported by Rusty Newton) |
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Async AGI crashes Asterisk when issuing “set variable” command without args (Reported by Antoine Pitrou) |
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Malformed AGI 520 Usage response (Reported by Tony Mountifield) |
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res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space (Reported by John Harris) |
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app_queue: Agent not called when caller is parked (Reported by wushumasters) |
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app_queue: Queue member stops being called after AMI “Redirect” action for queues with wrapuptime (Reported by Etienne Lessard) |
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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) |
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app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call (Reported by Lorne Gaetz) |
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app_confbridge: ConfBridge sometimes does not play user name recording while leaving (Reported by Robert Mordec) |
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res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 (Reported by Javier Riveros ) |
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chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable (Reported by Stefan Engström) |
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Out of bound memory access in PJSIP multipart parser crashes Asterisk (Reported by Sandro Gauci) |
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Asterisk Skinny memory exhaustion vulnerability leads to DoS (Reported by Sandro Gauci) |
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Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP (Reported by Sandro Gauci) |
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Audit manipulation of channel flags without locks (Reported by Joshua Colp) |
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res_pjsip_session: INVITE retransmissions could still setup the same call again. (Reported by Richard Mudgett) |
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res_rtp_asterisk: One way audio when transcoding (Reported by Henning Holtschneider) |
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Problems with Blind Transfer, PJSIP (Aastra 6869i) (Reported by Matthias Binder) |
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tcptls: Incorrect OpenSSL function call leads to misleading error report (Reported by Bob Ham) |
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Crash in Manager Reload when TLS Config Changes (Reported by Joshua Elson) |
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cel_odbc sometimes inserts CEL with wrong eventtime (Reported by Etienne Lessard) |
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func_cdr: CDR function does not permit empty values to be assigned (Reported by gkloepfer) |
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CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages. (Reported by Frederic LE FOLL) |
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Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used (Reported by Corey Farrell) |
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bridge_simple: Add support for streams (Reported by Kevin Harwell) |
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res_pjsip: Deadlock in T.38 framehook (Reported by Richard Mudgett) |
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res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked. (Reported by Richard Mudgett) |
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dial: Allow topology of dialing channel to influence dialed channel (Reported by Joshua Colp) |
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SIGSEGV, Segmentation fault. – ../sysdeps/x86_64/strlen.S: No such file or directory. (Reported by Andreas Krüger) |
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func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) |
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res_pjsip_session: Wrong From on reinvite when request and To URI differ (Reported by Yasin CANER) |
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pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux (Reported by abelbeck) |
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chan_sip: tcpbind uses wrong source address (Reported by Ksenia) |
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pjsip: Add database tables for RLS (Reported by Joshua Colp) |
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sdp: Implement T.38 (Reported by Joshua Colp) |
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Asterisk crash if hep.conf have some missing parameters (Reported by Joel Vandal) |
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STUN server with non-default-route transport causes INVITE delay (Reported by George Joseph) |
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chan_sip: ACK with SDP does not update a direct media bridge (Reported by Jean Aunis – Prescom) |
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res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) (Reported by scgm11) |
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res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) |
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res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) |
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format_wav: wav16 format read file only by 320 – half of frame (Reported by Vitaly K) |
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format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) |
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STUN never works when asterisk started without internet access (Reported by Jeremy Kister) |
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Audible clicks when playing sox encoded au file with STREAM FILE AGI command (Reported by Roman S.) |
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[UBSAN] strings.h:signed integer overflow in ast_str_case_hash (Reported by Badalian Vyacheslav) |
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res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (Reported by Richard Begg) |
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Listening TCP/TLS sockets stop when temporarily out of open files (Reported by Walter Doekes) |
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pjsip: Add database tables for PUBLISH support (Reported by Joshua Colp) |
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pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). (Reported by Alexander Traud) |
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pjproject_bundled: Merge 3 upstream deadlock patches into bundled (Reported by Ross Beer) |
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app_queue: PAUSEALL/UNPAUSEALL does not log reason (Reported by Troy Bowman) |
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chan_sip: Security vulnerability with client code header (Reported by Alex Villacís Lasso) |
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Unused realtime MOH classes not purged on ‘moh reload’ (Reported by Sébastien Couture) |
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res_pjsip: Excessive refcount reached on transport ao2 object (Reported by Ross Beer) |
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SIP Failed to parse multiple Supported: headers (Reported by Olle Johansson) |
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chan_sip: Session Timers required but refused wrongly. (Reported by Alexander Traud) |
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res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code (Reported by Yaacov Akiba Slama) |
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Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT (Reported by twisted) |
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libasteriskssl.so not found when asterisk is installed for the 1st time (Reported by George Joseph) |
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sdp: Add support for connection address management and topology updating (Reported by Joshua Colp) |
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xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client (Reported by Marcello Ceschia) |
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SDP crypto tag is validated incorrectly (Reported by Joerg Sonnenberger) |
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channel: Support dynamic number of file descriptors (Reported by Joshua Colp) |
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res_musiconhold: format option is not documented adequately (Reported by Jens Bürger) |
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No core dumps because of res_musiconhold chdir. (Reported by Walter Doekes) |
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xmpp: starttls problem causes connection spew (Reported by Matthias Urlichs) |
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pjproject_bundled build fails to download pjproject source when using cURL (Reported by Gergely Dömsödi) |
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JABBER_STATUS fails with improper code 7 for unavailable clients (Reported by Anthony Critelli) |
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Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available (Reported by Jeremy Kister) |
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WARNING for “JABBER: socket read error” should be more specific (Reported by Sean Darcy) |
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rtp_engine: Allocate RTP payloads on a per-session basis (Reported by Joshua Colp) |
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cdr: Problem setting variables in h exten (Reported by scgm11) |
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res_hep_pjsip: Asterisk insert wrong protocol name in “Protocol ID” field in HEP packets (Reported by Max Norba) |
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res_pjsip_messaging: Crash when using invalid URI in MessageSend ‘from’ argument. (Reported by Vinod Dharashive) |
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res_pjsip_pubsub: Crash when generating xpidf content (Reported by Andrew Green) |
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Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled (Reported by Kirsty Tyerman) |
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app_mixmonitor: Recording out of sync when 183 but no RTP (Reported by Aaron An) |
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app_queue: Queue stops calling members with local interface after forwarding in previous call (Reported by Robert Mordec) |
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res_rtp_asterisk: Implement RTCP Multiplexing – breaking WebRTC in Chrome (Reported by Dan Jenkins) |
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PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) |
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autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade). (Reported by Krzysztof Trempala) |
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res_pjsip_refer: blind call transfer w/o a user name doesn’t go to the s extension (Reported by Torrey Searle) |
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core: Malformed pattern matching extension (various factors) results in crash (Reported by xrobau) |
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chan_iax2: Reload of iax peer results in loss of host address/port (Reported by Richard Begg) |
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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) |
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Document the fact that Asterisk HEP support only works with the PJSIP channel driver (Reported by Olivier Krief) |
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Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk (Reported by Roman Bedros) |
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stasis_cache.c:845 caching_topic_exec: – misleading ERROR message (Reported by Smirnov Aleksey) |
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chan_pjsip: Dialplan function race condition (Reported by Joshua Colp) |
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pjsip/cli_commands: pjsip show channelstats shows wrong codec (Reported by Kevin Harwell) |
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res_musiconhold: musiconhold seems to think that the general section is a class and issues warning (Reported by Jonathan Harris) |
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res_pjsip: Crash when using IPv6 and Transport ws,wss (Reported by Michael Balen) |
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app_voicemail: Cannot set fromstring on a per-mailbox basis (Reported by Mark Scholten) |
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Websocket becomes disconnected when trying to place call from browser (Reported by Mark Michelson) |
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chan_sip: Call not cancelled after receiving a 422 response (Reported by Jean Aunis – Prescom) |
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core: Implement stream topology changing in channels (Reported by Joshua Colp) |
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Saynumber is trying to get “and” from “digits/” subfolder (Reported by Jonathan Harris) |
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Long lines in call files cause spurious syntax error (Reported by Dave Olszewski) |
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res_pjsip_transport_websocket: Via header is ‘WS’ when it should be ‘WSS’ (Reported by Jørgen H) |
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Implement ast_read_stream in channels (Reported by Joshua Colp) |
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res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging (Reported by Dmitry Wagin) |
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core: Playback URL fails after some time (Reported by Igor Gamayunov) |
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pjsip.conf.sample: user_agent: still refers to branch 12 (Reported by Tzafrir Cohen) |
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PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist (Reported by Mark Michelson) |
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res_pjsip: Crash when calling PJSIPShowEndpoint (Reported by Jørgen H) |
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res_pjsip_outbound_registration doesn’t know about network change events (Reported by George Joseph) |
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bridge: Passing the ‘p’ (play tone) flag to Bridge() application results in garbled audio (Reported by Sean Bright) |
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res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication (Reported by Peter Sokolov) |
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Implement ast_write_stream in channels (Reported by George Joseph) |
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Fix download_externals To Allow The Use Of curl Or wget (Reported by Michael L. Young) |
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Pattern matching with res_config_mysql extensions does not behave as expected (Reported by Charlie Smurthwaite) |
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stream: Add streams to “core show channel” (Reported by Joshua Colp) |
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DUNDi weight parameter not processed correctly (Reported by Peter Racz) |
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res_pjsip: Using an auth object for inbound and outbound authentication fails. (Reported by Richard Mudgett) |
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PJSIP Segfault 13.13.1 (Bundled PJSIP) (Reported by Nic Colledge) |
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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) |
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Function vmauthenticate accesses uninitialized memory (Reported by Filip Jenicek) |
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Error during LDAP modify action when user unregisters (Reported by Nicholas John Koch) |
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Integrity Check Of PJSIP Download Fails (Reported by Michael L. Young) |
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Fix query with double backslash in string literals and stop log warnings (Reported by Humberto Figuera) |
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res_config_sqlite3 uses incorrect query – unnecessary escape (Reported by Stepan) |
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SQlite3: Realtime queue loading fails after PRAGMA query result (Reported by Scott Griepentrog) |
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http: Crash on Reload Only in ast_tcptls_server_start (Reported by Joshua Elson) |
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Phone default have not ringing on ARM (Reported by Igor Goncharovsky) |
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pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh (Reported by Zach R) |
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res_pjsip_mwi: Asterisk does not terminate MWI subscription (Reported by Carl Fortin) |
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Implement stream topology (non-change request) API usage in channels (Reported by George Joseph) |
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VoiceMailPlayMsg not playing messages via realtime (Reported by Ryan Rittgarn) |
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‘Silence’ is truncated in Record() (Reported by var) |
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app_queue: reset abandoned in service level (Reported by scgm11) |
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Implement ast_stream_topology API (Reported by George Joseph) |
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chan_pjsip: Error when calling PJSIP client with domain specified (Reported by Norbert Varga) |
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core: Protect flags during ast_waitfor (Reported by Joshua Colp) |
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pbx: AMI Originate ignore “failed” extension on call failure (Reported by Nasir Iqbal) |
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stream: Add basic API (Reported by Joshua Colp) |
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configs/samples: The ‘identify’ entry is in the wrong section in sorcery.conf.sample (Reported by Torrey Searle) |
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Crash in srv.c on startup with pjsip (Reported by nappsoft) |
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res_stasis_device_state: Duplicate subscriptions when multiple received at same time (Reported by Joshua Colp) |
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ARI channelvars cause memory leak (Reported by Sébastien Duthil) |
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ari: Channels with pre-dial handlers cannot be hung up via ARI (Reported by Tom Pawelek) |
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core: Possibility of a frame “imbalance” leading to stuck channels. (Reported by Mark Michelson) |
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res_agi: run_agi eats frames it shouldn’t (Reported by George Joseph) |
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ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) |
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res_odbc.conf contains deprecated configuration: ‘pooling’, ‘shared_connections’, ‘limit’, and ‘idlecheck’ options were replaced by ‘max_connections’. (Reported by Anthony Messina) |
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res_resolver_unbound: FRACK! Excessive ref count trap tripped. (Reported by Richard Mudgett) |
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MixMonitorMute mutes through stream if already slinear (e.g. Originate) (Reported by David Woolley) |
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Crash on invalid contact domain (pjsip aor) (Reported by Dmitriy) |
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res_pjsip: Assertion when sending OPTIONS request to endpoint (Reported by Ross Beer) |
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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) |
|
|
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) |
|
|
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) |
|
|
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) |
|
|
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 scgm11) |
|
|
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) |
|
|
Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used (Reported by Frankie Chin) |
|
|
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) |
|
|
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) |
|
|
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) |
|
|
res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations. (Reported by Alexander Traud) |
|
|
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) |
|
|
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) |
|
|
– 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) |
|
|
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) |
|
|
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) |
|
|
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) |
|
|
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 “core show channeltype Surrogate” 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) |
|
|
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) |
|
|
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) |
|
|
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) |
|
|
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) |
|
|
res_pjsip: Use more compatible regex for get all (Reported by Dmitry Wagin) |
|
|
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) |
|
|
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) |
|
|
Allow “Comedian Mail” branding to be removed (Reported by John Covert) |
|
|
RTCP feedback for codec modules (Reported by Lorenzo Miniero) |
|
|
app_queue: Update Data of Queues (use queues as outbound calls container) (Reported by scgm11) |
|
|
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 beta, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0-beta1
Thank you for your continued support of Asterisk!