Asterisk Development Team
2017-Dec-21 22:05 UTC
[asterisk-announce] Certified Asterisk 13.18-cert1 Now Available
The Asterisk Development Team would like to announce the release of Certified
Asterisk 13.18-cert1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/certified-asterisk
The release of Certified Asterisk 13.18-cert1 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-27355 - Upgrade bundled PJPROJECT to 2.7
(Reported by Richard Mudgett)
* ASTERISK-27278 - [patch] chan_sip: Provide access to read the
full SIP Request-URI from INVITE
(Reported by David J.
Pryke)
* ASTERISK-27255 - alembic: Add support for Microsoft SQL
server
(Reported by Florian Floimair)
* ASTERISK-27253 - [patch] libsrtp-2.1.x support
(Reported by Alexander Traud)
* ASTERISK-27220 - Enable CHANNEL function to get from and to
tag from SIP Headers
(Reported by Andre Nazario)
* ASTERISK-27169 - Google OAuth 2.0 support for XMPP / Motif
(Reported by Andrey)
* ASTERISK-27173 - Support for GMIME 3.0
(Reported by
Tzafrir Cohen)
* ASTERISK-27092 - [patch] app_queue: Add Priority to AMI
QueueStatus
(Reported by Niklas Larsson)
* ASTERISK-27085 - [patch] chan_pjsip: Port SIPDtmfMode to
chan_pjsip
(Reported by Torrey Searle)
* ASTERISK-27066 - res_pjsip: Add DTMF INFO Failback mode
(Reported by Torrey Searle)
* ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could
block PJSIP taskprocessor on startup
(Reported by Alexei
Gradinari)
* ASTERISK-27068 - app_voicemail: Add global option
"imap_poll_logout" to specify post-polling disconnect
(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-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-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-23828 - pjsip - Need a command to list active SIP
subscriptions
(Reported by Rusty Newton)
* ASTERISK-26527 - Testsuite: increase timeout to check "core
fullybooted wait" up to 30 sec
(Reported by Badalian
Vyacheslav)
* 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)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27430 - README refers to security documents that do
not exist.
(Reported by Corey Farrell)
* ASTERISK-27382 - crash after an invalid rtcp packet from GT48
FXS gateway
(Reported by Tzafrir Cohen)
* ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
RTCP packet will write past where it should
(Reported by
Vitezslav Novy)
* ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
not applied on reload
(Reported by John Bigelow)
* ASTERISK-27460 - CDR: Deadlock using AMI Originate with
Variable CDR(amaflags)=...
(Reported by Richard Mudgett)
* ASTERISK-27421 - RTP source learning not working with devices
that have some clock issues
(Reported by nappsoft)
* ASTERISK-27453 - RTP: Blind transfer direct media scenario
results in one way audio.
(Reported by Richard Mudgett)
* ASTERISK-27452 - Security: chan_skinny: Memory exhaustion if
flooded with unauthenticated requests
(Reported by George
Joseph)
* ASTERISK-27454 - res_http_post: Don't require
GMIME_MAJOR_VERSION
(Reported by Joshua Colp)
* ASTERISK-27411 - pjsip: TCP connections may not be destroyed
(Reported by Joshua Colp)
* ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
responses.
(Reported by Corey Farrell)
* ASTERISK-27337 - chan_sip: Security vulnerability with client
code header (revisited)
(Reported by Richard Mudgett)
* ASTERISK-27319 - (Security) Function in PJSIP 2.7
miscalculates the length of an unsigned long variable in 64bit
machines
(Reported by Kim youngsung)
* ASTERISK-27391 - Regression: Deadlock between AOR named lock
and pjproject grp lock
(Reported by shaurya jain)
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael
Maier)
* ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
curl is loaded
(Reported by Ronald Raikes)
* ASTERISK-27372 - ARI: Node ARI client broken in latest
versions of 13 and 14
(Reported by Benjamin Keith Ford)
* ASTERISK-27047 - res_pjsip: user=phone added to Anonymous
caller-id when it shouldn't be.
(Reported by dtryba)
* ASTERISK-26988 - res_pjsip_session: user_eq_phone adds double
user=phone parameters to URIs
(Reported by dtryba)
* ASTERISK-27270 - cdr_mysql: various crashes at second module
reload if cdr_mysql.conf is configured
(Reported by
Tzafrir Cohen)
* ASTERISK-27301 - [patch] app_queue: Music On Hold for
real-time queues is not reset to default
(Reported by
Nathan Bruning)
* ASTERISK-25266 - Application Originate returns SUCCESS to
ORIGINATE_STATUS upon failure to originate
(Reported by
Allen Ford)
* ASTERISK-27192 - res_pjsip: Loss of SIP registrations causing
unavailable endpoints
(Reported by Richard Mudgett)
* ASTERISK-27305 - res_ari: Memory leaks in ARI when using
Content-Type: application/json
(Reported by David Hajek)
* ASTERISK-26922 - chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)
* ASTERISK-27324 - [patch] Dual-Stack server cannot be used as
IPv4 client via TCP/TLS
(Reported by Alexander Traud)
* ASTERISK-27317 - vector: multiple evaluation of elem in
AST_VECTOR_ADD_SORTED.
(Reported by Corey Farrell)
* ASTERISK-27318 - res_pjsip_mwi: uninitialized value from
ast_strings_match
(Reported by Corey Farrell)
* ASTERISK-27296 - [patch] False positive busy checks when
icalendar's recurrence-id mechanism is involved
(Reported
by Beno??t Dereck-Tricot)
* ASTERISK-27284 - Status of RFC 3323 and PJSIP
(Reported
by dtryba)
* ASTERISK-27216 - app_queue: does its
check-makeannouncement-logic twice each head-caller-loop
(Reported by Stefan Engstr??m)
* ASTERISK-27295 - Contact is improperly translated after
d178f497
(Reported by Sean Bright)
* ASTERISK-27292 - Multiple RTP Stream Created Breaking RFC2833
(SSRC Changes)
(Reported by Ross Beer)
* ASTERISK-27289 - A codeblock that maintains a bug,but maybe
the codeblock will never run
(Reported by Huangyx)
* ASTERISK-27283 - Realtime config fail with PostgreSQL version
before 9.1
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-27257 - bridge_native_rtp: half-way direct media
when using early bridging
(Reported by Jean Aunis -
Prescom)
* ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer -
Possible PJSIP Vulnerability
(Reported by Ross Beer)
* ASTERISK-26606 - tcptls: Incorrect OpenSSL function call
leads to misleading error report
(Reported by Bob Ham)
* ASTERISK-16898 - SRTP unprotect: authentication failure when
RTP sequence number switches from 65535 -> 0
(Reported by
Marcello Ceschia)
* ASTERISK-27274 - RTCP needs better packet validation to
resist port scans.
(Reported by Richard Mudgett)
* ASTERISK-27252 - RTP: One way audio with direct media and
strictrtp=yes.
(Reported by Richard Mudgett)
* ASTERISK-25524 - module reload res_calendar.so does not
reload everything in calendar.conf
(Reported by Jesper)
* ASTERISK-24588 - res_calendar does not process CalDAV from
Owncloud [fix included]
(Reported by Stefan Gofferje)
* ASTERISK-25523 - res_calendar: Warning about invalid channel
value (for notification) occurs even when event has no
notification configured.
(Reported by Jesper)
* ASTERISK-21399 - RTP Multicast of L16 (type 10): Asterisk and
wireshark disagree
(Reported by Tzafrir Cohen)
* ASTERISK-27248 - [patch]external_media_address and
external_signaling_address don't always honor localnet
(Reported by Walter Doekes)
* ASTERISK-27165 - CDR: CDR(start,u) function won't work in
cdr_custom config
(Reported by Jacek Konieczny)
* ASTERISK-27217 - chan_sip: Asterisk crashing when
subscription doesn't get set
(Reported by Bryan Walters)
* ASTERISK-24066 - res_smdi: convert to astobj2
(Reported
by Corey Farrell)
* ASTERISK-17540 - SDP origin attribute modified when issuing
re-INVITE because of directmedia=yes
(Reported by saghul)
* ASTERISK-27254 - alembic: prune_on_boot fix erroneous
(Reported by Florian Floimair)
* ASTERISK-27232 - When in queue on g722 with interruptions,
music on hold can get stuck and no longer play
(Reported
by Jens T.)
* ASTERISK-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-26879 - PJSIP external_media_address ignored if no
local_net options are provided
(Reported by Matt Jordan)
* ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at
channel_internal_api.c:478 during T.38 Fax Receive
(Reported by Ross Beer)
* ASTERISK-27225 - Crash when freeing dtls_cfg->cafile
(Reported by Richard Kenner)
* ASTERISK-27177 - ooh323c: misleading indentation in
addons/ooh323c/src/ooSocket.c
(Reported by Tzafrir Cohen)
* ASTERISK-27241 - libc segfault upon entry into app_directory
(Reported by David Moore)
* 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-26994 - Confbridge: CBAnn channels intermittently
become stuck when caller hangs up before recording name
(Reported by James Terhune)
* ASTERISK-20858 - app_minivm fails to clean up mkstemp files
(Reported by Walter Doekes)
* ASTERISK-16777 - several filename bugs in Record()
application
(Reported by klaus3000)
* ASTERISK-27168 - alembic: PJSIP scripts are missing column
dtls_fingerprint in ps_endpoints table
(Reported by
Florian Floimair)
* ASTERISK-23608 - ControlPlayback fails to play files with
names containing certain non-alpha characters
(Reported by
Jonathan White)
* ASTERISK-19103 - When using realtime queues, function
QUEUE_MEMBER_LIST() will return an error if no other
app/function has loaded the queues first. This problem does not
exist if queues.conf is used.
(Reported by Jim Van
Meggelen)
* ASTERISK-21241 - When using voicemail as announce only
(maxmsg=0), the star dtmf to enter the voicemail is not honored
(Reported by Eelco Brolman)
* ASTERISK-27204 - [patch] app_queue: Wrong queue stat
calculation
(Reported by sungtae kim)
* ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE
is used
(Reported by Torrey Searle)
* ASTERISK-27207 - XMPP OAuth not working due to inverted
logic
(Reported by Michael Kuron)
* ASTERISK-27174 - res_calendar_icalendar: Recurring events not
being loaded from Google calendar using ical
(Reported by
Mark Thompson)
* 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-27147 - Either asterisk or pjproject isn't re-using
tcp connections (again)
(Reported by George Joseph)
* ASTERISK-27193 - IPv6 receive address in message doesn't
include brackets
(Reported by Scott Griepentrog)
* ASTERISK-27158 - [patch] res_rtp_asterisk: RTCP statistics
are not available when native bridge is used
(Reported by
Torrey Searle)
* ASTERISK-27110 - RTP session is not fully destroyed on
channel hangup
(Reported by Matt Jordan)
* ASTERISK-26745 - Asymmetric codecs when
asymmetric_rtp_codec=no
(Reported by Jesse Ross)
* ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
(Reported by Ira Emus)
* ASTERISK-26659 - res_pjsip: PJSIP presence - missing braces
around the status element in XML
(Reported by Abraham
Liebsch)
* ASTERISK-27156 - Asterisk won't compile on Fedora 26 with
devmode enabled.
(Reported by Corey Farrell)
* ASTERISK-27130 - Applications ARI: Unsubscribe action for
deviceStates does not remove old subscriptions properly
(Reported by Sergej Kasumovic)
* 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-27142 - sounds: Conflict between files in
asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
(Reported by Corey Farrell)
* ASTERISK-27124 - app_playback.c:say_date_generic use
timezonename parameter
(Reported by Holger Hans Peter
Freyther)
* ASTERISK-27133 - res_rtp_asterisk: RTCP does not use ICE when
RTCP-MUX in use
(Reported by Joshua Colp)
* ASTERISK-27128 - [patch]res_stasis_snoop: When recording a
snoop channel (using ARI) where no media is being received, no
recording happens when theres no media
(Reported by Dan
Jenkins)
* 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-27127 - configs: Erroneous load directive in sample
configuration results in "Error loading module
'res_pjsip_multihomed.so'"
(Reported by HZMI8gkCvPpom0tM)
* ASTERISK-27105 - [patch]core: when setting 'maxfiles' in
asterisk.conf, a message is printed, even in rasterisk -x
(Reported by Tzafrir Cohen)
* 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-27108 - Crash using 'data get' CLI command
(Reported by Sean Bright)
* 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-27097 - pjproject_bundled: We don't pass options
needed for cross-compile to pjproject configure
(Reported
by George Joseph)
* ASTERISK-27095 - chan_pjsip: When connected_line_method is
set to invite, we're not trying UPDATE
(Reported by George
Joseph)
* 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-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-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-24052 - app_voicemail reloads result in leaked IMAP
sockets.
(Reported by Louis Jocelyn Paquet)
* ASTERISK-27051 - res_pjsip_mwi: unsolicited MWI has to be
unsubscribed on deleting the endpoint's last contact
(Reported by Alexei Gradinari)
* ASTERISK-27059 - res_stasis: Stolen channel references are
leaking
(Reported by George Joseph)
* ASTERISK-27060 - Comment typo format_g729.c
(Reported
by Matthew Fredrickson)
* ASTERISK-26919 - res_pjsip_dialog_info_body_generator:
Ringing&&InUse behavior difference between chan_sip and
res_pjsip
(Reported by Zach R)
* 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-27041 - Core/PBX: [patch] Deadlock between dialplan
execution and application unregistration
(Reported by
Frederic LE FOLL)
* ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at
sorcery.c
(Reported by Ryan Smith)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
get_write_timeout
(Reported by J??rgen H)
* 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-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-25101 - DTLS configuration can not be specified in
the general section - documentation
(Reported by Ben
Langfeld)
* 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-26964 - res_pjsip_session: Wrong From on reinvite
when request and To URI differ
(Reported by Yasin CANER)
* ASTERISK-26938 - Heap overflow in CSEQ header parsing affects
Asterisk chan_pjsip and PJSIP
(Reported by Sandro Gauci)
* 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-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-26333 - Problems with Blind Transfer, PJSIP (Aastra
6869i)
(Reported by Matthias Binder)
* 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-26143 - res_rtp_asterisk: One way audio when
transcoding
(Reported by Henning Holtschneider)
* 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-26860 - Upon RTCP reception, netsock2.c:210
ast_sockaddr_split_hostport: Port missing in (null)
(Reported by Evers Lab)
* 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-25823 - SIGSEGV, Segmentation fault. -
../sysdeps/x86_64/strlen.S: No such file or directory.
(Reported by Andreas Kr??ger)
* ASTERISK-26951 - chan_sip: ACK with SDP does not update a
direct media bridge
(Reported by Jean Aunis - Prescom)
* ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build
fails for non-SSE2 instrunction Linux
(Reported by
abelbeck)
* ASTERISK-26926 - func_speex: Crash caused by frame with no
datalen
(Reported by Richard Kenner)
* ASTERISK-26929 - pjsip: Add database tables for RLS
(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-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-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-26528 - [UBSAN] strings.h:signed integer overflow in
ast_str_case_hash
(Reported by Badalian Vyacheslav)
* 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-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-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-24712 - xmpp: starttls problem causes connection
spew
(Reported by Matthias Urlichs)
* 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-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-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-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-26841 - chan_sip: Call not cancelled after receiving
a 422 response
(Reported by Jean Aunis - Prescom)
* ASTERISK-26822 - pjsip/cli_commands: pjsip show channelstats
shows wrong codec
(Reported by Kevin Harwell)
* 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-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-25628 - res_config_pgsql: should match the behavior
of other drivers so that queue_log can disable adaptive logging
(Reported by Dmitry Wagin)
* 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-26313 - chan_sip : Asterisk restart seems to be
required for changing encryption option
(Reported by
benasse)
* 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-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-26669 - PJSIP Segfault 13.13.1 (Bundled PJSIP)
(Reported by Nic Colledge)
* ASTERISK-18731 - [patch] DUNDi weight parameter not processed
correctly
(Reported by Peter Racz)
* ASTERISK-26580 - [patch] Error during LDAP modify action when
user unregisters
(Reported by Nicholas John Koch)
* ASTERISK-26799 - res_pjsip: Using an auth object for inbound
and outbound authentication fails.
(Reported by Richard
Mudgett)
* 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-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-26109 - Asterisk fails building with OpenSSL 1.1.0
(Reported by Tzafrir Cohen)
* ASTERISK-26723 - VoiceMailPlayMsg not playing messages via
realtime
(Reported by Ryan Rittgarn)
* ASTERISK-18286 - [patch] 'Silence' is truncated in Record()
(Reported by var)
* 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-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-26704 - res_odbc.conf contains deprecated
configuration: 'pooling', 'shared_connections',
'limit', and
'idlecheck' options were replaced by 'max_connections'.
(Reported by Anthony Messina)
* ASTERISK-21094 - MixMonitorMute mutes through stream if
already slinear (e.g. Originate)
(Reported by David
Woolley)
* 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-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-26753 - AMI disconnect causes "ast_careful_fwrite:
fwrite() returned error: Broken pipe"
(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-26740 - voicemail API test: uses varlibdir instead
of datadir for a sound file
(Reported by Tzafrir Cohen)
* ASTERISK-26739 - voicemail API test: confuses expected and
actual values
(Reported by Tzafrir Cohen)
* ASTERISK-26731 - res_sorcery_memory_cache: memory leak on
every sorcery memory cache populate
(Reported by Ustinov
Artem)
* 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-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-26490 - res_pjsip: sends 481 Call/Transaction Does
Not Exist when transaction branch parameter contains "_"
(Reported by Juris Breicis)
* ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems
without IPv6
(Reported by Guido Falsi)
* ASTERISK-24330 - Requirement for 'wss' value in Contact
header transport parameter on inbound traffic violates RFC7118
(Reported by Marek Cervenka)
* ASTERISK-26546 - mips64el and x32 - undefined reference to
symbol 'dlopen@@GLIBC_2.2'
(Reported by Tzafrir Cohen)
* 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-26603 - [patch] chan_pjsip: not switching sending
codec to receiving codec when asymmetric_rtp_codec=no
(Reported by Alexei Gradinari)
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects
incoming calls after 2 minutes - rtptimeout behaving badly -
regression
(Reported by Michael Keuter)
* ASTERISK-26503 - app_voicemail: Asterisk crashes when
MailboxExists is used
(Reported by Doug Lytle)
New Features made in this release:
-----------------------------------
* ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
incoming INVITE Request-URI.
(Reported by Richard Mudgett)
* ASTERISK-27215 - [patch]AMI : Add CancelAtxfer Action
(Reported by Thomas Sevestre)
* ASTERISK-27117 - core: Add support for timelen parsing to
ast_parse_arg and ACO.
(Reported by Corey Farrell)
* 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-26630 - Make logging PJPROJECT messages a bit
easier
(Reported by Richard Mudgett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.18-cert1
Thank you for your continued support of Asterisk!
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