Asterisk Development Team
2017-Oct-03 12:27 UTC
[asterisk-announce] Asterisk 15.0.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk
15.0.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
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:
-----------------------------------
* ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could
block PJSIP taskprocessor on startup
(Reported by Alexei
Gradinari)
* ASTERISK-27043 - Core/BuildSystem: Add defines to fix build
with LibreSSL
(Reported by Guido Falsi)
* ASTERISK-27042 - Unpatched asterisk sources fail to build on
FreeBSD due to missing crypt.h file
(Reported by Guido
Falsi)
* ASTERISK-26419 - audiohooks: Remove redundant codec
translations when using audiohooks
(Reported by Michael
Walton)
* ASTERISK-26976 - libsrtp-2.x.x support
(Reported by
Alex)
* ASTERISK-27014 - configurable busy_timeout in sqlite
backends
(Reported by Marek Cervenka)
* ASTERISK-26124 - res_agi: Set audio format for EAGI audio
stream
(Reported by John Fawcett)
* ASTERISK-26088 - Investigate heavy memory utilization by
res_pjsip_pubsub
(Reported by Richard Mudgett)
* ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report
channel name with res_hep_rtcp when using chan_sip
(Reported by Nir Simionovich (GreenfieldTech - Israel))
* ASTERISK-26932 - [patch] SIP/SDP: No rtpmap for static RTP
payload IDs
(Reported by Alexander Traud)
* ASTERISK-26864 - res_pjsip_session: Add support for overlap
dialling
(Reported by Richard Begg)
* ASTERISK-26846 - chan_sip: Add rtcp-mux support
(Reported by Sean Bright)
* ASTERISK-26568 - pbx_spool: OUTGOING_RETRY variable
(Reported by Roman Shubovich)
* ASTERISK-26292 - app_confbridge: 3D-Conferencing via Binaural
Synthesis
(Reported by Dennis Guse)
* ASTERISK-23828 - pjsip - Need a command to list active SIP
subscriptions
(Reported by Rusty Newton)
* ASTERISK-26559 - app_queue: New service level calculation
(Reported by Sebastian Gutierrez)
* ASTERISK-26658 - Add ability for dialplan show to display
filenames/line numbers of registered extensions
(Reported
by Jonathan R. Rose)
* ASTERISK-26527 - Testsuite: increase timeout to check "core
fullybooted wait" up to 30 sec
(Reported by Badalian
Vyacheslav)
* ASTERISK-22992 - [patch]Asterisk app_originate doesn't allow
setting Caller*ID on the originating channel
(Reported by
Anthony Messina)
* ASTERISK-26624 - res_calendar_caldav: Add support for gmail
(Reported by Eduardo Scudeller Libardi)
* ASTERISK-26562 - app_controlplayback: Transmit Silence on
ControlPlayback pause
(Reported by Mikheili Dautashvili)
* ASTERISK-24517 - TLS support for Solaris, Ming and non-glibc
Linux systems
(Reported by Timo Ter??s)
* ASTERISK-26540 - cdr_radius: use radcli instead of
freeradius-client
(Reported by Tzafrir Cohen)
* ASTERISK-26558 - app_queue: add variable to know if the call
is not answered after a queue
(Reported by Sebastian
Gutierrez)
* ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
(Reported by Sebastian Gutierrez)
* ASTERISK-26217 - [patch] Codec 2 Mode 2400
(Reported by
Alexander Traud)
* ASTERISK-26538 - codec_opus: Add sample to
configs/samples/codecs.conf.sample
(Reported by Kevin
Harwell)
* ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps',
and 'ari set debug' CLI commands
(Reported by Matt Jordan)
* ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by
blacklisting host subnets that are not involved in RTP
(Reported by Michael Walton)
* ASTERISK-26422 - [patch] Force calendars to do new fetch
after module reload
(Reported by Ludovic Gasc (Eyepea))
* ASTERISK-26398 - core: Remove ABI differences of LOW_MEMORY
(Reported by Corey Farrell)
* ASTERISK-26409 - codec_opus: Update Asterisk to support the
translation codec.
(Reported by Kevin Harwell)
* ASTERISK-26289 - Announcer channels in ConfBridges cause
inefficiencies
(Reported by Mark Michelson)
* ASTERISK-26321 - ARI : Add reason answered_elsewhere to
channel hangup
(Reported by Jean Aunis - Prescom)
* ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let
dialplan know what fax transport was used
(Reported by
Alexei Gradinari)
* ASTERISK-26218 - [patch] iLBC 20
(Reported by Alexander
Traud)
* ASTERISK-26190 - [patch] SRTP: Enable AES-256 and AES-GCM.
(Reported by Alexander Traud)
* ASTERISK-26220 - Add support for noreturn function
attributes.
(Reported by Corey Farrell)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27152 - Sending a "tel" uri in a From or To header
in an unauthenticated message causes asterisk to crash
(Reported by Ross Beer)
* ASTERISK-27103 - core: ast_safe_system command injection
possible.
(Reported by Corey Farrell)
* ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even
with strict RTP enabled
(Reported by Joshua Colp)
* ASTERISK-27231 - res_rtp_asterisk: Allow remote SSRC to
change due to renegotiation
(Reported by Joshua Colp)
* ASTERISK-27222 - core: Don't queue up multiple video update
frames.
(Reported by Joshua Colp)
* ASTERISK-27212 - bridge_softmix: Quickly joining/leaving may
cause video stream to remain in SFU
(Reported by Richard
Mudgett)
* ASTERISK-27202 - If wget is not installed and "or" is not
available, external components (excluding pjsip) are not
installed
(Reported by Se??n C. McCord)
* ASTERISK-27200 - manager: hook event is not being raised
(Reported by Kevin Harwell)
* ASTERISK-27179 - res_pjsip_session: Handling of 'msid' is
incorrect
(Reported by Kevin Harwell)
* ASTERISK-27182 - bridge: Crash when mapping streams
(Reported by Joshua Colp)
* ASTERISK-27189 - Make --with-pjproject-bundled the default
for Asterisk 15
(Reported by George Joseph)
* ASTERISK-27180 - channel: requester leaks joint_cap on
success.
(Reported by Corey Farrell)
* ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
(Reported by Ira Emus)
* ASTERISK-27119 - res_pjsip: parse/add msid attribute when
webrtc is enabled
(Reported by Kevin Harwell)
* ASTERISK-27143 - bridge_softmix / res_rtp_asterisk: Fix
packet loss and renegotiation issues.
(Reported by Joshua
Colp)
* ASTERISK-27142 - sounds: Conflict between files in
asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
(Reported by Corey Farrell)
* ASTERISK-25810 - 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)
* ASTERISK-27136 - bridge_softmix: Don't reorder SFU streams
(Reported by Joshua Colp)
* ASTERISK-27134 - bridge_softmix: Reuse any removed streams
for video
(Reported by Joshua Colp)
* ASTERISK-27133 - res_rtp_asterisk: RTCP does not use ICE when
RTCP-MUX in use
(Reported by Joshua Colp)
* ASTERISK-27123 - confbridge: Name recordings are left on
filesystem
(Reported by Sergej Kasumovic)
* ASTERISK-27122 - chan_iax2: On reload MWI taskprocessors keep
adding up
(Reported by Sergej Kasumovic)
* ASTERISK-26807 - sounds: New 3-D Binaural audio features
require new sound prompts
(Reported by Rusty Newton)
* ASTERISK-25816 - French conf-adminmenu, conf-usermenu prompts
differ in content from the English files
(Reported by
Benoit Duverger)
* ASTERISK-26274 - Resolve open sounds issues and then create a
new sounds release (1.5.1? or 1.6?)
(Reported by Rusty
Newton)
* ASTERISK-27118 - res_pjsip_session / res_rtp_asterisk: Add
support for BUNDLE
(Reported by Joshua Colp)
* ASTERISK-27036 - res_pjsip: Asterisk crashes when an
extension tries to use PJSIP trunk with from_user containing
'@'
(Reported by Maxim Vasilev)
* ASTERISK-27023 - res_rtp_asterisk: Deadlock when TURN session
in use
(Reported by Jatin Jain)
* ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add
only really different domain with TLS.
(Reported by
Alexander Traud)
* ASTERISK-27093 - ODBC deadlocks when app_directory tries to
play back non-existent voicemail greeting
(Reported by
James Terhune)
* ASTERISK-27100 - channel: ast_waitfordigit_full fails to
clear flag in an error branch.
(Reported by Corey Farrell)
* ASTERISK-27090 - PJSIP: Deadlock using TCP transport
(Reported by Richard Mudgett)
* ASTERISK-26997 - Create an StreamEcho dialplan application
(Reported by Kevin Harwell)
* ASTERISK-27076 - chan_pjsip: Add support for multiple
streams
(Reported by Joshua Colp)
* ASTERISK-27088 - res_rtp_asterisk: Better handle ICE
renegotiation and unidirectional negotiation
(Reported by
Joshua Colp)
* ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code()
(Reported by Ross Beer)
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
events
(Reported by Ove Aursand)
* ASTERISK-27065 - call hangup after leaving app_queue
(Reported by Marek Cervenka)
* ASTERISK-24052 - app_voicemail reloads result in leaked IMAP
sockets.
(Reported by Louis Jocelyn Paquet)
* ASTERISK-27074 - core_local: local channel data not being
properly unref'ed and unlocked
(Reported by Kevin Harwell)
* ASTERISK-27075 - bridge: stuck channel(s) after failed
attended transfer
(Reported by Kevin Harwell)
* ASTERISK-27060 - Comment typo format_g729.c
(Reported
by Matthew Fredrickson)
* ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan
execution and application unregistration
(Reported by
Frederic LE FOLL)
* ASTERISK-25370 - res_corosync segfaults at startup with
corosync version > 2.x
(Reported by mdu113)
* ASTERISK-27026 - res_ari: Crash when no ari.conf
configuration file exists
(Reported by Ronald Raikes)
* ASTERISK-27016 - Crash occurs when a channel in a
'mixing,dtmf_events' bridge is muted multiple times.
(Reported by Chris Howard)
* ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at
sorcery.c
(Reported by Ryan Smith)
* ASTERISK-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for
RTCP component
(Reported by Michael Walton)
* ASTERISK-26923 - bridging: T.38 request is lost when channels
are added to bridge
(Reported by Torrey Searle)
* ASTERISK-27053 - res_pjsip_refer/session: Calls dropped
during transfer
(Reported by Kevin Harwell)
* ASTERISK-27052 - Asterisk build process fails with flag
--with-pjproject-bundled with curl download command and slow
network
(Reported by alex)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
get_write_timeout
(Reported by J??rgen H)
* ASTERISK-27039 - chan_pjsip: Device state is idle when
channel from endpoint is in early media
(Reported by
Joshua Colp)
* ASTERISK-26996 - chan_pjsip: Flipping between codecs
(Reported by Michael Maier)
* ASTERISK-26281 - chan_pjsip would send INVITE to
'Unreachable' endpoints
(Reported by Jacek Konieczny)
* ASTERISK-26973 - bridge: Crash when freeing frame and
snooping
(Reported by Michel R. Vaillancourt)
* ASTERISK-19291 - Background in realtime
(Reported by
Andrew Nowrot)
* ASTERISK-27025 - channel / meetme: Fix missing parentheses
(Reported by Joshua Colp)
* ASTERISK-27021 - GET /recordings/stored returns 500 Internal
Server Error
(Reported by Tim Morgan)
* ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets
in wrong byte order on Intel platform when using slin codec
(Reported by Frankie Chin)
* ASTERISK-23951 - Asterisk attempts and fails to build
format_mp3 even if mp3lib was not downloaded
(Reported by
Tzafrir Cohen)
* ASTERISK-25294 - srtp's crypto_get_random deprecated
(Reported by Tzafrir Cohen)
* ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't
describe BEEP argument
(Reported by Rusty Newton)
* ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set
variable" command without args
(Reported by Antoine
Pitrou)
* ASTERISK-25662 - Malformed AGI 520 Usage response
(Reported by Tony Mountifield)
* ASTERISK-27008 - res_format_attr_h264: SDP parse fails if
fmtp optional parameters have a space
(Reported by John
Harris)
* ASTERISK-26399 - app_queue: Agent not called when caller is
parked
(Reported by wushumasters)
* ASTERISK-26400 - app_queue: Queue member stops being called
after AMI "Redirect" action for queues with wrapuptime
(Reported by Etienne Lessard)
* ASTERISK-26715 - 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)
* ASTERISK-26975 - app_queue: Non-zero wrapup time can cause
agents not to receive queue calls after transfer queue call
(Reported by Lorne Gaetz)
* ASTERISK-27012 - app_confbridge: ConfBridge sometimes does
not play user name recording while leaving
(Reported by
Robert Mordec)
* ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with
authentication failure 10 or 110
(Reported by Javier
Riveros )
* ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice
completion failure/delay if client offers rtcp-mux as
negotiable
(Reported by Stefan Engstr??m)
* ASTERISK-26939 - Out of bound memory access in PJSIP
multipart parser crashes Asterisk
(Reported by Sandro
Gauci)
* ASTERISK-26940 - Asterisk Skinny memory exhaustion
vulnerability leads to DoS
(Reported by Sandro Gauci)
* ASTERISK-26938 - Heap overflow in CSEQ header parsing affects
Asterisk chan_pjsip and PJSIP
(Reported by Sandro Gauci)
* ASTERISK-26789 - Audit manipulation of channel flags without
locks
(Reported by Joshua Colp)
* ASTERISK-26998 - res_pjsip_session: INVITE retransmissions
could still setup the same call again.
(Reported by
Richard Mudgett)
* ASTERISK-26143 - res_rtp_asterisk: One way audio when
transcoding
(Reported by Henning Holtschneider)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
6869i)
(Reported by Matthias Binder)
* ASTERISK-26606 - tcptls: Incorrect OpenSSL function call
leads to misleading error report
(Reported by Bob Ham)
* ASTERISK-26983 - Crash in Manager Reload when TLS Config
Changes
(Reported by Joshua Elson)
* ASTERISK-25032 - [patch]cel_odbc sometimes inserts CEL with
wrong eventtime
(Reported by Etienne Lessard)
* ASTERISK-26173 - func_cdr: CDR function does not permit empty
values to be assigned
(Reported by gkloepfer)
* ASTERISK-25506 - [patch]CONFBRIDGE failure after an
app_confbrige.so module reload results in segfault or
error/warning messages.
(Reported by Frederic LE FOLL)
* ASTERISK-24529 - Using AMI Action Bridge to on an already
bridged channel causes the incorrect return priority to be used
(Reported by Corey Farrell)
* ASTERISK-26966 - bridge_simple: Add support for streams
(Reported by Kevin Harwell)
* ASTERISK-26974 - res_pjsip: Deadlock in T.38 framehook
(Reported by Richard Mudgett)
* ASTERISK-26908 - res_pjsip: The ChanIsAvail causes a
res_pjsip session to be leaked.
(Reported by Richard
Mudgett)
* ASTERISK-26959 - dial: Allow topology of dialing channel to
influence dialed channel
(Reported by Joshua Colp)
* ASTERISK-25823 - SIGSEGV, Segmentation fault. -
../sysdeps/x86_64/strlen.S: No such file or directory.
(Reported by Andreas Kr??ger)
* ASTERISK-26926 - func_speex: Crash caused by frame with no
datalen
(Reported by Richard Kenner)
* ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
when request and To URI differ
(Reported by Yasin CANER)
* ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build
fails for non-SSE2 instrunction Linux
(Reported by
abelbeck)
* ASTERISK-26922 - chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)
* ASTERISK-26929 - pjsip: Add database tables for RLS
(Reported by Joshua Colp)
* ASTERISK-26949 - sdp: Implement T.38
(Reported by
Joshua Colp)
* ASTERISK-26953 - Asterisk crash if hep.conf have some missing
parameters
(Reported by Joel Vandal)
* ASTERISK-26890 - STUN server with non-default-route transport
causes INVITE delay
(Reported by George Joseph)
* ASTERISK-26951 - chan_sip: ACK with SDP does not update a
direct media bridge
(Reported by Jean Aunis - Prescom)
* ASTERISK-26692 - res_rtp_asterisk: Crash in
dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
(Reported by Sebastian Gutierrez)
* ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP
address string
(Reported by Niklas Larsson)
* ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when
receiving packet
(Reported by Adagio)
* ASTERISK-26613 - format_wav: wav16 format read file only by
320 - half of frame
(Reported by Vitaly K)
* ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in
MixMonitor
(Reported by Ivan Myalkin)
* ASTERISK-21856 - STUN never works when asterisk started
without internet access
(Reported by Jeremy Kister)
* ASTERISK-20984 - Audible clicks when playing sox encoded au
file with STREAM FILE AGI command
(Reported by Roman S.)
* ASTERISK-26528 - [UBSAN] strings.h:signed integer overflow in
ast_str_case_hash
(Reported by Badalian Vyacheslav)
* ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use
same IP as explicit transport
(Reported by Richard Begg)
* ASTERISK-26903 - Listening TCP/TLS sockets stop when
temporarily out of open files
(Reported by Walter Doekes)
* ASTERISK-26928 - pjsip: Add database tables for PUBLISH
support
(Reported by Joshua Colp)
* ASTERISK-26927 - pjproject_bundled: Crash on
pj_ssl_get_info() while ioqueue_on_read_complete().
(Reported by Alexander Traud)
* ASTERISK-26905 - pjproject_bundled: Merge 3 upstream
deadlock patches into bundled
(Reported by Ross Beer)
* ASTERISK-26920 - app_queue: PAUSEALL/UNPAUSEALL does not log
reason
(Reported by Troy Bowman)
* ASTERISK-26897 - chan_sip: Security vulnerability with client
code header
(Reported by Alex Villac??s Lasso)
* ASTERISK-25974 - Unused realtime MOH classes not purged on
'moh reload'
(Reported by S??bastien Couture)
* ASTERISK-26916 - res_pjsip: Excessive refcount reached on
transport ao2 object
(Reported by Ross Beer)
* ASTERISK-21721 - SIP Failed to parse multiple Supported:
headers
(Reported by Olle Johansson)
* ASTERISK-26915 - chan_sip: Session Timers required but
refused wrongly.
(Reported by Alexander Traud)
* ASTERISK-26363 - res_pjsip: Bye sent to sip trunk is not
authenticated even after receiving a 407 error code
(Reported by Yaacov Akiba Slama)
* ASTERISK-26896 - Overflow of buffer to PQEscapeStringConn
with large app_args causes ABRT
(Reported by twisted)
* ASTERISK-26705 - libasteriskssl.so not found when asterisk is
installed for the 1st time
(Reported by George Joseph)
* ASTERISK-26900 - sdp: Add support for connection address
management and topology updating
(Reported by Joshua Colp)
* ASTERISK-21009 - xmpp_pubsub_unsubscribe: Could not create IQ
when creating pubsub unsubscription on client
(Reported by
Marcello Ceschia)
* ASTERISK-25490 - [patch]SDP crypto tag is validated
incorrectly
(Reported by Joerg Sonnenberger)
* ASTERISK-26885 - channel: Support dynamic number of file
descriptors
(Reported by Joshua Colp)
* ASTERISK-26086 - res_musiconhold: format option is not
documented adequately
(Reported by Jens B??rger)
* ASTERISK-23996 - No core dumps because of res_musiconhold
chdir.
(Reported by Walter Doekes)
* ASTERISK-24712 - xmpp: starttls problem causes connection
spew
(Reported by Matthias Urlichs)
* ASTERISK-26814 - pjproject_bundled build fails to download
pjproject source when using cURL
(Reported by Gergely
D??ms??di)
* ASTERISK-23510 - JABBER_STATUS fails with improper code 7 for
unavailable clients
(Reported by Anthony Critelli)
* ASTERISK-21855 - Asterisk crashes when XMPP message is sent
(JabberSend) and no internet connection is available
(Reported by Jeremy Kister)
* ASTERISK-25622 - WARNING for "JABBER: socket read error"
should be more specific
(Reported by Sean Darcy)
* ASTERISK-26515 - rtp_engine: Allocate RTP payloads on a
per-session basis
(Reported by Joshua Colp)
* ASTERISK-26818 - cdr: Problem setting variables in h exten
(Reported by Sebastian Gutierrez)
* ASTERISK-26850 - res_hep_pjsip: Asterisk insert wrong
protocol name in "Protocol ID" field in HEP packets
(Reported by Max Norba)
* ASTERISK-26484 - res_pjsip_messaging: Crash when using
invalid URI in MessageSend 'from' argument.
(Reported by
Vinod Dharashive)
* ASTERISK-26776 - res_pjsip_pubsub: Crash when generating
xpidf content
(Reported by Andrew Green)
* ASTERISK-26880 - Asterisk crashes when multiple speex users
join confbridge with pp_vad and dtx enabled
(Reported by
Kirsty Tyerman)
* ASTERISK-26875 - app_mixmonitor: Recording out of sync when
183 but no RTP
(Reported by Aaron An)
* ASTERISK-26862 - app_queue: Queue stops calling members with
local interface after forwarding in previous call
(Reported by Robert Mordec)
* ASTERISK-26732 - res_rtp_asterisk: Implement RTCP
Multiplexing - breaking WebRTC in Chrome
(Reported by Dan
Jenkins)
* ASTERISK-26879 - PJSIP external_media_address ignored if no
local_net options are provided
(Reported by Matt Jordan)
* ASTERISK-26867 - autochan: Locking in a function
ast_autochan_destroy() on destroyed channel (after masquerade).
(Reported by Krzysztof Trempala)
* ASTERISK-26869 - res_pjsip_refer: blind call transfer w/o a
user name doesn't go to the s extension
(Reported by
Torrey Searle)
* ASTERISK-26668 - core: Malformed pattern matching extension
(various factors) results in crash
(Reported by xrobau)
* ASTERISK-26865 - chan_iax2: Reload of iax peer results in
loss of host address/port
(Reported by Richard Begg)
* ASTERISK-26872 - 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)
* ASTERISK-26717 - Document the fact that Asterisk HEP support
only works with the PJSIP channel driver
(Reported by
Olivier Krief)
* ASTERISK-26643 - Extra new line in Device field of
DeviceStateChange AMI Event after restart of Asterisk
(Reported by Roman Bedros)
* ASTERISK-25237 - stasis_cache.c:845 caching_topic_exec: -
misleading ERROR message
(Reported by Smirnov Aleksey)
* ASTERISK-26857 - chan_pjsip: Dialplan function race
condition
(Reported by Joshua Colp)
* ASTERISK-26822 - pjsip/cli_commands: pjsip show channelstats
shows wrong codec
(Reported by Kevin Harwell)
* ASTERISK-26353 - res_musiconhold: musiconhold seems to think
that the general section is a class and issues warning
(Reported by Jonathan Harris)
* ASTERISK-26685 - res_pjsip: Crash when using IPv6 and
Transport ws,wss
(Reported by Michael Balen)
* ASTERISK-24562 - app_voicemail: Cannot set fromstring on a
per-mailbox basis
(Reported by Mark Scholten)
* ASTERISK-26842 - Websocket becomes disconnected when trying
to place call from browser
(Reported by Mark Michelson)
* ASTERISK-26841 - chan_sip: Call not cancelled after receiving
a 422 response
(Reported by Jean Aunis - Prescom)
* ASTERISK-26839 - core: Implement stream topology changing in
channels
(Reported by Joshua Colp)
* ASTERISK-26598 - Saynumber is trying to get "and" from
"digits/" subfolder
(Reported by Jonathan Harris)
* ASTERISK-17067 - Long lines in call files cause spurious
syntax error
(Reported by Dave Olszewski)
* ASTERISK-26796 - res_pjsip_transport_websocket: Via header is
'WS' when it should be 'WSS'
(Reported by J??rgen H)
* ASTERISK-26816 - Implement ast_read_stream in channels
(Reported by Joshua Colp)
* ASTERISK-25628 - res_config_pgsql: should match the behavior
of other drivers so that queue_log can disable adaptive logging
(Reported by Dmitry Wagin)
* ASTERISK-26774 - core: Playback URL fails after some time
(Reported by Igor Gamayunov)
* ASTERISK-26825 - pjsip.conf.sample: user_agent: still refers
to branch 12
(Reported by Tzafrir Cohen)
* ASTERISK-26823 - PJSIP: Persistent subscriptions can cause
FRACKs if endpoint does not exist
(Reported by Mark
Michelson)
* ASTERISK-26623 - res_pjsip: Crash when calling
PJSIPShowEndpoint
(Reported by J??rgen H)
* ASTERISK-26808 - res_pjsip_outbound_registration doesn't know
about network change events
(Reported by George Joseph)
* ASTERISK-26781 - bridge: Passing the 'p' (play tone) flag to
Bridge() application results in garbled audio
(Reported by
Sean Bright)
* ASTERISK-26782 - res_pjsip: URI requirement for fields is not
consistently documented and error does not provide indication
(Reported by Peter Sokolov)
* ASTERISK-26793 - Implement ast_write_stream in channels
(Reported by George Joseph)
* ASTERISK-26812 - [patch] Fix download_externals To Allow The
Use Of curl Or wget
(Reported by Michael L. Young)
* ASTERISK-18271 - Pattern matching with res_config_mysql
extensions does not behave as expected
(Reported by
Charlie Smurthwaite)
* ASTERISK-26811 - stream: Add streams to "core show channel"
(Reported by Joshua Colp)
* ASTERISK-18731 - [patch] DUNDi weight parameter not processed
correctly
(Reported by Peter Racz)
* ASTERISK-26799 - res_pjsip: Using an auth object for inbound
and outbound authentication fails.
(Reported by Richard
Mudgett)
* ASTERISK-26669 - PJSIP Segfault 13.13.1 (Bundled PJSIP)
(Reported by Nic Colledge)
* ASTERISK-26738 - 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)
* ASTERISK-25893 - Function vmauthenticate accesses
uninitialized memory
(Reported by Filip Jenicek)
* ASTERISK-26580 - [patch] Error during LDAP modify action when
user unregisters
(Reported by Nicholas John Koch)
* ASTERISK-26802 - [patch] Integrity Check Of PJSIP Download
Fails
(Reported by Michael L. Young)
* ASTERISK-15858 - [patch] Fix query with double backslash in
string literals and stop log warnings
(Reported by
Humberto Figuera)
* ASTERISK-26057 - res_config_sqlite3 uses incorrect query -
unnecessary escape
(Reported by Stepan)
* ASTERISK-23457 - SQlite3: Realtime queue loading fails after
PRAGMA query result
(Reported by Scott Griepentrog)
* ASTERISK-26794 - http: Crash on Reload Only in
ast_tcptls_server_start
(Reported by Joshua Elson)
* ASTERISK-26714 - Phone default have not ringing on ARM
(Reported by Igor Goncharovsky)
* ASTERISK-26696 - pjsip_pubsub: PJSIP Subscription Persistence
in AstDB Does not update on subscription refresh
(Reported
by Zach R)
* ASTERISK-26756 - res_pjsip_mwi: Asterisk does not terminate
MWI subscription
(Reported by Carl Fortin)
* ASTERISK-26790 - Implement stream topology (non-change
request) API usage in channels
(Reported by George Joseph)
* ASTERISK-26723 - VoiceMailPlayMsg not playing messages via
realtime
(Reported by Ryan Rittgarn)
* ASTERISK-18286 - [patch] 'Silence' is truncated in Record()
(Reported by var)
* ASTERISK-26775 - app_queue: reset abandoned in service level
(Reported by Sebastian Gutierrez)
* ASTERISK-26786 - Implement ast_stream_topology API
(Reported by George Joseph)
* ASTERISK-26248 - chan_pjsip: Error when calling PJSIP client
with domain specified
(Reported by Norbert Varga)
* ASTERISK-26788 - core: Protect flags during ast_waitfor
(Reported by Joshua Colp)
* ASTERISK-26115 - pbx: AMI Originate ignore "failed" extension
on call failure
(Reported by Nasir Iqbal)
* ASTERISK-26773 - stream: Add basic API
(Reported by
Joshua Colp)
* ASTERISK-26785 - configs/samples: The 'identify' entry is in
the wrong section in sorcery.conf.sample
(Reported by
Torrey Searle)
* ASTERISK-26772 - Crash in srv.c on startup with pjsip
(Reported by nappsoft)
* ASTERISK-26770 - res_stasis_device_state: Duplicate
subscriptions when multiple received at same time
(Reported by Joshua Colp)
* ASTERISK-26767 - ARI channelvars cause memory leak
(Reported by S??bastien Duthil)
* ASTERISK-26716 - ari: Channels with pre-dial handlers cannot
be hung up via ARI
(Reported by Tom Pawelek)
* ASTERISK-26632 - core: Possibility of a frame "imbalance"
leading to stuck channels.
(Reported by Mark Michelson)
* ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't
(Reported by George Joseph)
* ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
manipulation through agi
(Reported by Morten Tryfoss)
* ASTERISK-26704 - res_odbc.conf contains deprecated
configuration: 'pooling', 'shared_connections',
'limit', and
'idlecheck' options were replaced by 'max_connections'.
(Reported by Anthony Messina)
* ASTERISK-26765 - res_resolver_unbound: FRACK! Excessive ref
count trap tripped.
(Reported by Richard Mudgett)
* ASTERISK-21094 - MixMonitorMute mutes through stream if
already slinear (e.g. Originate)
(Reported by David
Woolley)
* ASTERISK-26679 - Crash on invalid contact domain (pjsip aor)
(Reported by Dmitriy)
* ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS
request to endpoint
(Reported by Ross Beer)
* ASTERISK-26754 - build_tools: make_build_h does not handle \
in user name
(Reported by Kirill Katsnelson)
* ASTERISK-26755 - app_queue: Random queues disappear on "core
reload queue all"
(Reported by Kirill Katsnelson)
* ASTERISK-26735 - res_pjsip_endpoint_identifier_ip:
"srv_lookups" after match in .conf has no effect
(Reported
by Michael Maier)
* ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add
support for SRV
(Reported by Joshua Colp)
* ASTERISK-26743 - PJPROJECT: Detecting compiled max log level
does not work.
(Reported by Richard Mudgett)
* ASTERISK-26731 - res_sorcery_memory_cache: memory leak on
every sorcery memory cache populate
(Reported by Ustinov
Artem)
* ASTERISK-26739 - voicemail API test: confuses expected and
actual values
(Reported by Tzafrir Cohen)
* ASTERISK-26740 - voicemail API test: uses varlibdir instead
of datadir for a sound file
(Reported by Tzafrir Cohen)
* ASTERISK-26665 - app_queue: Agent ringing, Caller hangup
before timeout, no agent name logged - missing RINGNOANSWER?
(Reported by Marek Cervenka)
* ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments,
(rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return
0
(Reported by Aaron An)
* ASTERISK-26672 - Crash when setting remote address on RTP
instance
(Reported by Richard Mudgett)
* ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP
(Reported by Alexander Traud)
* ASTERISK-26691 - Remember SDP negotiation on
SIP_CODEC_INBOUND.
(Reported by Alexander Traud)
* ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL
dialplan function around masquerade
(Reported by Joshua
Colp)
* ASTERISK-26684 - res_pjsip: Various issues with compact SIP
headers
(Reported by Joshua Elson)
* ASTERISK-26683 - res_calendar: Calendars duplicated after
module reload
(Reported by Martin Tomec)
* ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact
Headers Enabled
(Reported by JoshE)
* ASTERISK-26621 - app_queue: Queue application does not ring
members with Local interface
(Reported by Jonas Kellens)
* ASTERISK-26586 - chan_sip: Segfaults upon reload if client
with MWI wasn't registered
(Reported by Michael Kuron)
* ASTERISK-25494 - build: GCC 5.1.x catches some new const,
array bounds and missing paren issues
(Reported by George
Joseph)
* ASTERISK-24499 - Need more explicit debug when PJSIP
dialstring is invalid
(Reported by Rusty Newton)
* ASTERISK-25083 - Message.c: Message channel becomes saturated
with frames leading to spammy log messages
(Reported by
Jonathan Rose)
* ASTERISK-26653 - pjproject_bundled doesn't verify already
downloaded tarballs
(Reported by George Joseph)
* ASTERISK-26433 - chan_sip: Allows To-tag checks to be
bypassed, setting up new calls
(Reported by Walter Doekes)
* ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp
line
(Reported by J??rgen H)
* ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all
aors
(Reported by George Joseph)
* ASTERISK-26647 - Support older DNS style for OpenBSD
(Reported by snuffy)
* ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does
Not Exist when transaction branch parameter contains "_"
(Reported by Juris Breicis)
* ASTERISK-26629 - tests/manager: 4 test failures as a result
of iostream change
(Reported by Joshua Colp)
* ASTERISK-26109 - Asterisk fails building with OpenSSL 1.1.0
(Reported by Tzafrir Cohen)
* ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems
without IPv6
(Reported by Guido Falsi)
* ASTERISK-26603 - [patch] chan_pjsip: not switching sending
codec to receiving codec when asymmetric_rtp_codec=no
(Reported by Alexei Gradinari)
* ASTERISK-24330 - Requirement for 'wss' value in Contact
header transport parameter on inbound traffic violates RFC7118
(Reported by Marek Cervenka)
* ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in
RTCP
(Reported by Hector Royo Concepcion)
* ASTERISK-26604 - chan_sip: sip reload doesn't apply changes
to tlscertfile, tlsciphers, etc.
(Reported by Michael
Kuron)
* ASTERISK-26608 - Compile and link failures on OpenBSD
(Reported by snuffy)
* ASTERISK-26520 - codec_opus: Generated fmtp line has no
content
(Reported by Sebastian Gutierrez)
* ASTERISK-26605 - codec_opus: Spammed warning when Opus
negotiated but codec_opus not loaded.
(Reported by Richard
Mudgett)
* ASTERISK-26516 - pjsip: Memory corruption with possible
memory leak.
(Reported by Richard Mudgett)
* ASTERISK-24515 - Unconditional use of fopencookie() /
funopen() is non-portable
(Reported by Timo Ter??s)
* ASTERISK-26556 - manager: AMI version report same in Ast 13 &
14, despite Ast 14 syntax changes
(Reported by Michelle
Dupuis)
* ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and
makes asterisk CLI read garbage
(Reported by George
Joseph)
* ASTERISK-26575 - testsuite: Need to check PJSIP functionality
when res_srtp is not loaded.
(Reported by Joshua Colp)
* ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on
hold temporarily locks up set
(Reported by Jason)
* ASTERISK-26573 - Some typos in documentation of chan_sip.c
(Reported by C.J. Collier)
* ASTERISK-26571 - res_pjsip: Resolution incorrect when
explicit IPv6 transport configured
(Reported by Joshua
Colp)
* ASTERISK-26468 - ari: Bridge events stop working after this
sequence of ARI calls
(Reported by Daniele Pallastrelli)
* ASTERISK-24400 - ooh323 sends wrong hangup code
(Reported by Dmitry Melekhov)
* ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11
regressions
(Reported by Matt Jordan)
* ASTERISK-26412 - build: Prepare for gcc 6.2
(Reported
by George Joseph)
* ASTERISK-26509 - A few non-critical deprecation warnings when
building on Ubuntu 16.10
(Reported by Jonathan Harris)
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects
incoming calls after 2 minutes - rtptimeout behaving badly -
regression
(Reported by Michael Keuter)
* ASTERISK-26549 - app_dial: When PickupChan() is used some
channels may have incorrect device state
(Reported by
Joshua Colp)
* ASTERISK-24274 - [patch]Codec Format Is Not Included in the
SDP Media Attributes When SLIN48 Codec Is Used
(Reported
by Frankie Chin)
* ASTERISK-26311 - [patch] rtp_engine: Allow more than 32
dynamic payload types.
(Reported by Alexander Traud)
* ASTERISK-26546 - mips64el and x32 - undefined reference to
symbol 'dlopen@@GLIBC_2.2'
(Reported by Tzafrir Cohen)
* ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of
formats to maximum
(Reported by Joshua Colp)
* ASTERISK-26476 - chan_sip: Incorrect display option "Outbound
reg. retry 403" in "sip show settings"
(Reported by Sergey
Grachev)
* ASTERISK-25070 - Fix FTBFS on Hurd
(Reported by
Gabriele Giacone)
* ASTERISK-26537 - AMI: NewConnectedLine event is not
documented
(Reported by Etienne Lessard)
* ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed
as argument 2 to memcpy
(Reported by Badalian Vyacheslav)
* ASTERISK-26524 - astobj2: data_size variable is wasted space
when AO2_DEBUG is not enabled.
(Reported by Corey Farrell)
* ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash
(Reported by Ian Gilmour)
* ASTERISK-26387 - Asterisk segfaults shortly after starting
even with no active calls.
(Reported by Harley Peters)
* ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash
when publishing, in publisher_client_send at
res_pjsip_outbound_publish.c
(Reported by Matt Krokosz)
* ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing
enough to be a nuisance
(Reported by Joshua Colp)
* ASTERISK-26514 - Super Awesome Company: Don't specify
transport in pjsip.conf
(Reported by Rusty Newton)
* ASTERISK-26510 - pjproject_bundled uses the
--strip-components option of tar which isn't supported in older
versions
(Reported by George Joseph)
* ASTERISK-22480 - Embedded pjproject: build.mak contains
hardcoded full path to version.mak
(Reported by Matt
Jordan)
* ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes
File not Module
(Reported by Alexander Traud)
* ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing
change
(Reported by Bill Brigden)
* ASTERISK-26503 - app_voicemail: Asterisk crashes when
MailboxExists is used
(Reported by Doug Lytle)
* ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can
cause audio loss and wonkiness
(Reported by Andreas
Wetzel)
* ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual
Stack) installations.
(Reported by Alexander Traud)
* ASTERISK-26482 - [patch] chan_pjsip: segfault on already
disconnected session
(Reported by Alexei Gradinari)
* ASTERISK-26455 - cdr_radius / cel_radius: try fix memory
leak
(Reported by Badalian Vyacheslav)
* ASTERISK-26421 - Segmentation Fault with ARI originate into
mixing bridge with 43 clients
(Reported by Andrew Nagy)
* ASTERISK-26444 - 'features show' command in CLI does not
return prompt.
(Reported by John Kiniston)
* ASTERISK-26356 - menuselect: invalid test for GTK2
(Reported by Tzafrir Cohen)
* ASTERISK-26477 - pjproject: SEGV during SSL operations
(Reported by George Joseph)
* ASTERISK-26462 - [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)
* ASTERISK-26439 - chan_rtp: Crash when originating
(Reported by Kayode)
* ASTERISK-17470 - [patch] - When videosupport=yes, asterisk
allows one end peer to send video, even though the other end
supports only audio.
(Reported by effie mouzeli)
* ASTERISK-26416 - pjproject-bundled: configure fails to check
for all required utilities
(Reported by Corey Farrell)
* ASTERISK-26466 - core: Be forgiving on external callerid that
may be flawed so we don't drop events
(Reported by Richard
Mudgett)
* ASTERISK-26362 - res_config_mysql: Broken after 13.10
(Reported by Carlos Chavez)
* ASTERISK-26446 - app_dial: There's no way to override the
hangupcause on unanswered channels
(Reported by George
Joseph)
* ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT
detection triggered.
(Reported by Alexander Traud)
* ASTERISK-26453 - res_pjsip_config_wizard: Memory leak in
module_unload
(Reported by Badalian Vyacheslav)
* ASTERISK-26410 - core: Asterisk 14 doesn't show the header in
the console or verbose when starting
(Reported by Dan
Jenkins)
* ASTERISK-24311 - Populating database via Alembic fails when
using same database for multiple schema sets
(Reported by
Dafi Ni)
* ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT No
Symmetric Response.
(Reported by Alexander Traud)
* ASTERISK-26330 - app_queue: Changing the "ringinuse"
parameter of a queue doesn't affect dynamic members
(Reported by Etienne Lessard)
* ASTERISK-26426 - format_ogg_opus: remove from source
(Reported by Kevin Harwell)
* ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
bindaddr=[::]
(Reported by Jacek)
* ASTERISK-25468 - Deadlock in chan_sip - core show locks shows
do_monitor lock
(Reported by Barry Flanagan)
* ASTERISK-26397 - manager: PresenceState action crashes
Asterisk 14
(Reported by Andrew Nagy)
* ASTERISK-26389 - res_odbc: Clean up pooling options
(Reported by Joshua Colp)
* ASTERISK-26273 - core: Won't compile when LOW_MEMORY is
enabled
(Reported by Anthony Messina)
* ASTERISK-26391 - Consoles do not display verbose logger
messages even when requested.
(Reported by Marcelo Terres)
* ASTERISK-26352 - Astcanary dies when doing "core restart"
(Reported by Walter Doekes)
* ASTERISK-19867 - asterisk fails to lower its priority when
astcanary dies
(Reported by Xavier Hienne)
* ASTERISK-26263 - SQL error when using realtime and
registering extension / inserting into ps_contacts
(Reported by Jeppe Ryskov Larsen)
* ASTERISK-26365 - rtp: Offer with multiple payloads for same
codec is incorrectly handled
(Reported by Joshua Colp)
* ASTERISK-26374 - res_pjsip_multihomed: Contact port is
rewritten for connectionful protocols
(Reported by Joshua
Colp)
* ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so
instructed
(Reported by Tzafrir Cohen)
* ASTERISK-26367 - rtp: Timestamps broken when video frame is
across multiple RTP packets
(Reported by Joshua Colp)
* ASTERISK-26375 - res_pjsip_transport_management: Log message
states seconds, but time value is milliseconds
(Reported
by Joshua Colp)
* ASTERISK-19968 - TCP Session-Timers not dropping call
(Reported by Aaron Hamstra)
* ASTERISK-26364 - res_pjsip: Don't assume a request will have
target addresses
(Reported by Joshua Colp)
* ASTERISK-26360 - app_queue: "queue show" output gets "failed
to extend from 240 to 327" msgs.
(Reported by Richard
Mudgett)
* ASTERISK-26358 - chan_sip: Contact is updated on re-200, but
not on re-INVITE
(Reported by Walter Doekes)
* ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes
unexpected callerid
(Reported by Kevin Harwell)
* ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c:
Request 'REGISTER' failed
(Reported by Dmitry Melekhov)
* ASTERISK-26317 - res_pjsip_session: Add ability to use
preferred codec only
(Reported by Aaron An)
* ASTERISK-26264 - res_pjsip: Crash when applying ACL from
non-existent endpoint
(Reported by nappsoft)
* ASTERISK-26272 - chan_sip: File descriptors leak (UDP
sockets)
(Reported by Etienne Lessard)
* ASTERISK-20234 - SRTP not working with some devices (Eg
snom320) - Message "We are requesting SRTP for audio, but they
responded without it!"
(Reported by tootai)
* ASTERISK-26341 - ARI: Stopping a media playlist only stops
the current media URI being played back, and not the whole list
(Reported by Matt Jordan)
* ASTERISK-26291 - res_pjsip_session: segfault on already
disconnected session
(Reported by Alexei Gradinari)
* ASTERISK-23989 - [patch]SDP offer/answer fails if crypto keys
added to non-crypto offer
(Reported by Olle Johansson)
* ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p'
argument) is enabled and callee rejects a call or hangs up.
(Reported by Etienne Lessard)
* ASTERISK-26331 - Crash on ???core show channeltype Surrogate???
in ast_format_cap_get_names
(Reported by CGI.NET)
* ASTERISK-26085 - app_mp3: results in timeout for streams
(Reported by Jens B??rger)
* ASTERISK-26269 - res_pjsip: Wrong state for aors without
registered contacts after startup
(Reported by nappsoft)
* ASTERISK-26226 - pbx: Asterisk crash on AMI action
"ShowDialplan" when there's a circular dependency between
contexts
(Reported by Etienne Lessard)
* ASTERISK-26299 - app_queue: Queue application sometimes stops
calling members with Local interface
(Reported by Etienne
Lessard)
* ASTERISK-26279 - pjproject-bundled: Fails to compile on
Debian 6
(Reported by George Joseph)
* ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not
cleaning up properly
(Reported by Alexander Traud)
* ASTERISK-26203 - res_fax: Deadlock when using
FAXOPT(gateway)=yes with Local channels
(Reported by
Etienne Lessard)
* ASTERISK-24822 - Deadlock: Fax Gateway framehook creates
locking inversion in T.38 query option with features bridging
code
(Reported by David Brillert)
* ASTERISK-22732 - Deadlock potential in res_fax and CCSS with
local channels.
(Reported by Richard Mudgett)
* ASTERISK-26282 - AEL: macro-call in Dial application, macro
"lacks 's' extension"
(Reported by chris de rock)
* ASTERISK-22820 - [patch] Plaintext auth is still supported in
IAX2
(Reported by Eugene)
* ASTERISK-22374 - Finish mapping the sip.conf parameters to
res_sip.conf parameters
(Reported by Matt Jordan)
* ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
SSLv3, security fix POODLE (CVE-2014-3566)
(Reported by
abelbeck)
* ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include
annexb=no attribute.
(Reported by Ali Ghavidel)
* ASTERISK-25472 - Swagger scripts are not replacing format
variable in file brief
(Reported by Corey Farrell)
* ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but
it's not mandatory to compile it
(Reported by J??zsef
Dud??s)
* ASTERISK-26305 - Asterisk 14: Two resolver unbound testsuite
tests fail
(Reported by Richard Mudgett)
* ASTERISK-26288 - followme: fails to reset config items to
default values on reload
(Reported by Tzafrir Cohen)
* ASTERISK-26303 - [patch] BuildSystem: ca_list_path
capabilities not detected in PJProject.
(Reported by
Alexander Traud)
* ASTERISK-25492 - ARI: Path parameters are case sensitive
(Reported by Joshua Colp)
* ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via
chan_pjsip
(Reported by Ross Beer)
* ASTERISK-26233 - pbx: Failure to remove inconsistent
extension names
(Reported by Corey Farrell)
* ASTERISK-26246 - Security: Privilege escalation by AMI adding
dialplan extensions.
(Reported by Richard Mudgett)
* ASTERISK-26267 - ast_register_atexit callbacks should be run
on failed startup.
(Reported by Corey Farrell)
* ASTERISK-26241 - res_pjsip: When using compact headers, rpid
and pai are incorrectly generated
(Reported by George
Joseph)
* ASTERISK-25797 - app_queue: Crash when calling a queue with a
member with a forward to an nonexistent extension
(Reported by Etienne Lessard)
* ASTERISK-26239 - res_pjsip_logger: An empty global/debug
option is treated as a "match all" hostname
(Reported by
George Joseph)
* ASTERISK-26238 - res_pjsip: Empty global default_from_user
causes crash
(Reported by Joshua Colp)
* ASTERISK-26268 - alembic: 'auth_username' not in PJSIP
'identify_by' enum
(Reported by Joshua Colp)
* ASTERISK-26253 - sdp_srtp: libsrtp now a required dependency,
shouldn't be
(Reported by Ben Merrills)
* ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade +
indicate
(Reported by Ross Beer)
* ASTERISK-26183 - alembic: error when using sqlalchemy version
1.1.0b2
(Reported by Kevin Harwell)
* ASTERISK-26283 - res_resolver_unbound: fails configure on
older Ubuntu and CentOS
(Reported by George Joseph)
* ASTERISK-26280 - DNS lookups can block channel media paths
(Reported by Mark Michelson)
* ASTERISK-26278 - asterisk.h should produce a reasonable error
for external modules that fail to define AST_MODULE_SELF_SYM.
(Reported by Corey Farrell)
* ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a
similar treatment for module unloading as
res_pjsip_outbound_registration.c
(Reported by Richard
Mudgett)
* ASTERISK-26265 - Errors ignored from some parts of system
initialization.
(Reported by Corey Farrell)
* ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex
for get all
(Reported by Dmitry Wagin)
* ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains
brackets with IP6
(Reported by Alexander Traud)
* ASTERISK-25996 - Remove "live_dangerously" requirement on
DB(read)
(Reported by Andrew Nagy)
* ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1:
"libasteriskpj.so: undefined reference to..."
(Reported by
Hans van Eijsden)
* ASTERISK-26237 - Fax is detected on regular calls.
(Reported by Richard Mudgett)
* ASTERISK-26227 - sqlalchemy error due to long identifier
name
(Reported by Mark Michelson)
* ASTERISK-14 - asterisk leaves zombie mpg123
(Reported
by dcarr)
* ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
command and attended transfer handling
(Reported by Ben
Smithurst)
* ASTERISK-26199 - PJSIP: tx_data_destroy called twice
(Reported by Scott Griepentrog)
* ASTERISK-26166 - res_pjsip_pubsub: Crash when decrementing
reference count of message
(Reported by Ross Beer)
* ASTERISK-26174 - res_pjsip: Crash when freeing cloned message
in distributor
(Reported by Ross Beer)
* ASTERISK-26216 - res_fax: Deadlock when detect fax while
channel executing Playback
(Reported by Richard Mudgett)
* ASTERISK-26214 - Allow arbitrary time for fax detection to
end on a channel
(Reported by Richard Mudgett)
New Features made in this release:
-----------------------------------
* ASTERISK-27063 - Add support for systemd socket activation
(Reported by Corey Farrell)
* ASTERISK-27117 - core: Add support for timelen parsing to
ast_parse_arg and ACO.
(Reported by Corey Farrell)
* ASTERISK-27129 - ast_waitfordigit_full: add support for
filtering DTMF keys which can break the wait.
(Reported by
Corey Farrell)
* ASTERISK-26995 - Add QUEUE_FLOAT_PENALTY to app_queue
(Reported by Steve Davies)
* ASTERISK-26878 - func_channel: Add ability to get the callid
so dialplan has access to it.
(Reported by Richard
Mudgett)
* ASTERISK-26863 - res_pjsip: Add endpoint identification
scheme based on a configured SIP header/value
(Reported by
Matt Jordan)
* ASTERISK-17428 - [patch] Allow "Comedian Mail" branding to be
removed
(Reported by John Covert)
* ASTERISK-26584 - [patch] RTCP feedback for codec modules
(Reported by Lorenzo Miniero)
* ASTERISK-19862 - app_queue: Update Data of Queues (use queues
as outbound calls container)
(Reported by Sebastian
Gutierrez)
* ASTERISK-26630 - Make logging PJPROJECT messages a bit
easier
(Reported by Richard Mudgett)
* ASTERISK-26587 - app_originate: Add option to execute gosub
prior to dial
(Reported by dkerr)
* ASTERISK-26595 - ARI: Add the ability to control the source
of video in a multi-party mixing bridge
(Reported by Matt
Jordan)
* ASTERISK-26492 - ARI: Add ability to specify channel
variables on websocket events
(Reported by Mark Michelson)
* ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing
events
(Reported by Matt Jordan)
* ASTERISK-26277 - 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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