Asterisk Development Team
2017-Aug-02 16:20 UTC
[asterisk-users] Asterisk 15.0.0-beta1 Now Available
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:
-----------------------------------
* 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 scgm11)
* 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 scgm11)
* ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
(Reported by scgm11)
* 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-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 scgm11)
* 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 scgm11)
* 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 scgm11)
* 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 scgm11)
* 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 scgm11)
* 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 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!
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<html><head><title>Re: [asterisk-users] Asterisk 15.0.0-beta1 Now Available</title> </head> <body> <span style=" font-family:'Arial'; font-size: 12pt;">Hello Asterisk,<br> <br> Wednesday, August 2, 2017, 9:20:19 AM, you wrote:<br> <br> <br>> The Asterisk Development Team would like to announce the first beta of Asterisk 15.0.0.<br> > This beta is available for immediate download at <br> > http://downloads.asterisk.org/pub/telephony/asterisk <br> > The release of Asterisk 15.0.0-beta1 resolves several issues reported by the<br> > community and would have not been possible without your participation.<br><br> In the interest of helping with the beta, maybe including a link to where to report bugs would be useful? I downloaded it but it fails to compile and ends with this error:<br> <br> <b> [CC] app_voicemail.c -> app_voicemail.o<br> cc1: error: unrecognized command line option "-Wno-format-truncation"<br> make[1]: *** [app_voicemail.o] Error 1<br> make: *** [apps] Error 2<br> <br> </b>A long time ago I knew where to report problems, but I've no idea where that location might be any more.<br> <br> 32 bit CentOS final version Don't recall if it's 5 or 6 but I know it's out of support as yum update stopped working.<br> <br> -- Ira</body></html>
On Wed, Aug 2, 2017, at 02:28 PM, Ira wrote:> Re: [asterisk-users] Asterisk 15.0.0-beta1 Now AvailableHello Asterisk, > > Wednesday, August 2, 2017, 9:20:19 AM, you wrote: > > > > 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. > > In the interest of helping with the beta, maybe including a link to > where to report bugs would be useful? I downloaded it but it fails to > compile and ends with this error: > > *? ?[CC] app_voicemail.c -> app_voicemail.o cc1: error: unrecognized > command line option "-Wno-format-truncation" make[1]: *** > [app_voicemail.o] Error 1 make: *** [apps] Error 2 > > *A long time ago I knew where to report problems, but I've no idea where > that location might be any more. > > 32 bit CentOS final version Don't recall if it's 5 or 6 but I know it's > out of support as yum update stopped working.The address to report issues is https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org