update to Asterisk 13.19.0 -- this contains both security fixes
and general bug fixes, fixes AST-2017-005, AST-2017-006, AST-2017-007,
AST-2017-008, AST-2017-009, AST-2017-10, AST-2017-11, AST-2017-12,
AST-2017-13, and AST-2017-14 (note that a number of these only
pertain to PJSIP which isn't used in pkgsrc)
----- 13.19.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.19.0.
The release of Asterisk 13.19.0 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:
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-27413 - Add cache_media_frames debugging option.
(Reported by Richard Mudgett)
* ASTERISK-27206 - res_pjsip: No mechanism exists to limit
endpoint identification to IP only
(Reported by Ben Merrills)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27531 - Compiler optimizations can break module load
sequence.
(Reported by abelbeck)
* ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
Contact crashes asterisk
(Reported by Ross Beer)
* ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
read()
(Reported by Abhay Gupta)
* ASTERISK-25079 - AMI bridge of channels results in MOH not
destroyed and robotic audio on one channel
(Reported by Zane Conkle)
* ASTERISK-27490 - chan_console: 'set active' fails to work
(Reported by Tzafrir Cohen)
* ASTERISK-24756 - ConfBridge sound_muted does not work from
CLI or AMI
(Reported by Thomas Frederiksen)
* ASTERISK-25649 - Transfer application does not work with
Local channels - documentation misleading
(Reported by Ivan Ullmann)
* ASTERISK-25869 - chan_sip: "rejected because extension not
found" should be logged as a security event
(Reported by Brian J. Murrell)
* ASTERISK-27440 - Strictrtp has issues to qualify video rtp
streams
(Reported by Wim De Vlaminck)
* ASTERISK-24329 - Music On Hold announcement cuts intro of
music the first time it is played
(Reported by Thomas Frederiksen)
* ASTERISK-19657 - Coverity Report: Fix issues for error type
CHAR_IO
(Reported by Matt Jordan)
* ASTERISK-27175 - iax.conf demo peer is invalid
(Reported by Tzafrir Cohen)
* ASTERISK-27430 - README refers to security documents that do
not exist.
(Reported by Corey Farrell)
* ASTERISK-20281 - "core set verbose" behaves strangely, can't
alias it, cli.conf example broken
(Reported by Tim Ringenbach at Asteria Solutions Group)
* 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-27408 - Identify causes and fix
pjsip/resolver/srv/failover/in_dialog/transport_tcp
(Reported by Corey Farrell)
* ASTERISK-18411 - Queue members with hints for state_interface
get stuck in "In Use" state.
(Reported by Steven T. Wheeler)
* ASTERISK-26131 - chan_sip: Crash Asterisk (in
sip_request_call at chan_sip.c) by making a call to a single
character in a dot pattern match
(Reported by Dwayne Hubbard)
* ASTERISK-27475 - codec_opus requires libcurl
(Reported by Samuel For)
* ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
not applied on reload
(Reported by John Bigelow)
* ASTERISK-27465 - CLI Completion Not Working
(Reported by Ross Beer)
* ASTERISK-27460 - CDR: Deadlock using AMI Originate with
Variable CDR(amaflags)=...
(Reported by Richard Mudgett)
* ASTERISK-27453 - RTP: Blind transfer direct media scenario
results in one way audio.
(Reported by Richard Mudgett)
* ASTERISK-20643 - SIP ICE support - remove hardcoded
limitation on SDP size, make ICE support disabled by default in
SIP, maybe provide a better warning message
(Reported by Roy)
* ASTERISK-26980 - pjsip: Clean up WebRTC disables
(Reported by abelbeck)
* 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-23735 - Transcoding makes bad choice in high-rate
translations
(Reported by Richard Kenner)
* ASTERISK-27445 - ARI: Updating a bridge gives wrong error
message.
(Reported by Frank Durden)
* ASTERISK-24662 - [patch] column and row headers for Signed
Linear format variants in output of 'core show translation' are
ambiguous
(Reported by Rusty Newton)
* ASTERISK-27353 - H323 audio starts with a delay of 2
seconds.
(Reported by Marco Giordani)
* ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
media
(Reported by Kevin Harwell)
* ASTERISK-27437 - [patch] ICE: server-reflexive candidates
(srflx) with Dual-Stack.
(Reported by Alexander Traud)
* ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
IPv6 addresses.
(Reported by Alexander Traud)
* ASTERISK-27435 - [patch] configure:
pjsip_evsub_set_uas_timeout not found.
(Reported by Alexander Traud)
* ASTERISK-27431 - Asterisk fails to build when openssl headers
are not installed.
(Reported by Corey Farrell)
* ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
(Reported by Ivan Larionov)
* ASTERISK-27421 - RTP source learning not working with devices
that have some clock issues
(Reported by nappsoft)
* ASTERISK-27361 - Attended transfer crashes in Asterisk
13.17.2
(Reported by Alessandro Pimenta)
* ASTERISK-27238 - Bridging: Crash freeing a frame that's
already been freed
(Reported by Richard Kenner)
* ASTERISK-27412 - core: Audiohook freeing interpolated frame
when it shouldn't.
(Reported by Mikhail)
* ASTERISK-27423 - app_record: We set the RECORD_STATUS
channel variable before closing the file
(Reported by George Joseph)
* ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
insert same ip address in "source ip address" and "destination
ip address" fields in HEP packets
(Reported by Max Norba)
* ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
is equal to RemoteAddress)
(Reported by Vasilii Rogin)
* ASTERISK-27415 - asterisk.conf: Setting astctl without
setting astrundir is ineffective.
(Reported by Corey Farrell)
* 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-27393 - res_pjsip: Crash occurs when an empty
contact read from astdb or database
(Reported by Aaron An)
* ASTERISK-27290 - res_pjsip: PIDF contact field has
malformed/invalid XML
(Reported by basildane)
* ASTERISK-27032 - res_pjsip: TLS options do not handle empty
values
(Reported by seanchann.zhou)
* ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
enabled but not configured.
(Reported by Alexander Traud)
* ASTERISK-26426 - format_ogg_opus: remove from source
(Reported by Kevin Harwell)
* ASTERISK-27378 - Modules: Fix issues with CLI completion.
(Reported by Corey Farrell)
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27390 - Audit menuselect module dependencies
(Reported by Corey Farrell)
* ASTERISK-27389 - Optional API modules should not allow
unload.
(Reported by Corey Farrell)
* ASTERISK-27369 - Bridge() dialplan application fails without
setting BRIDGERESULT channel variable
(Reported by James Terhune)
* ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
documentation
(Reported by Igor Goncharovsky)
* ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
'imap_delete_old_greeting'
(Reported by Anthony Messina)
* ASTERISK-27194 - jitterbuffer: Does not handle case where
translator returns null frame.
(Reported by Joshua Elson)
* ASTERISK-26639 - core: Disabling xmldoc support does not
work. Also results in abort during Asterisk startup.
(Reported by Mr Dini)
* ASTERISK-27372 - ARI: Node ARI client broken in latest
versions of 13 and 14
(Reported by Benjamin Keith Ford)
* ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
absence of the Expires header field with an unsubscribe action.
(Reported by Jonathan Cloots)
* ASTERISK-25960 - The config_hook unit test causes Asterisk to
crash if run a second time
(Reported by George Joseph)
* ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
when rtp_ipv6 set to yes
(Reported by Martin Cisárik)
* ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
curl is loaded
(Reported by Ronald Raikes)
* ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
but first on SDP media level.
(Reported by Alexander Traud)
* ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
Assertion on un/re-load: mod.id == -1
(Reported by Tzafrir Cohen)
* ASTERISK-23462 - Cannot disable SIP debugging via CLI after
enabling with conf file option - also 'sip set debug off'
reports debugging disabled, when it really isn't
(Reported by Rusty Newton)
* ASTERISK-27328 - Missing openssl dependencies in
res_rtp_asterisk and tcptls
(Reported by Tzafrir Cohen)
* ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
(o=) contains local address.
(Reported by Alexander Traud)
* ASTERISK-27343 - Fails to build in FreeBSD due to
sys/sysmacros.h not existing there
(Reported by Guido Falsi)
* ASTERISK-27340 - backtrace.c: Crash due to double-free.
(Reported by Corey Farrell)
* ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
stopping.
(Reported by Alexander Traud)
* ASTERISK-27333 - sip_to_pjsip not correctly handling
disallow=all directive
(Reported by Torrey Searle)
Improvements made in this release:
-----------------------------------
* ASTERISK-24297 - cdr.c: Minor code optimizations.
(Reported by Richard Mudgett)
* ASTERISK-27449 - [PATCH] When failing to acquire target
during attended transfer, display wanted extension
(Reported by Niklas Larsson)
* ASTERISK-27456 - app_voicemail: Add new object for
VoicemailUserEntry
(Reported by sungtae kim)
* ASTERISK-27380 - ast_coredumper: allow pointing out the
asterisk binary explicitly
(Reported by Tzafrir Cohen)
* ASTERISK-23556 - Compilation warning for invert.c (array
subscript is above array bounds)
(Reported by Marcello Ceschia)
* ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
(Reported by Richard Mudgett)
* ASTERISK-27335 - CDR performance needs improvement.
(Reported by Richard Mudgett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.0
Thank you for your continued support of Asterisk!
----- 13.18.5 -----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18.
The available releases are released as versions 13.18.5, 14.7.5,
15.1.5 and 13.18-cert2.
The following security vulnerabilities were resolved in these versions:
* AST-2017-014: Crash in PJSIP resource when missing a contact header
A select set of SIP messages create a dialog in Asterisk. Those
SIP messages must contain a contact header. For those messages,
if the header was not present and using the PJSIP channel driver,
it would cause Asterisk to crash. The severity of this vulnerability
is somewhat mitigated if authentication is enabled. If authentication
is enabled a user would have to first be authorized before reaching
the crash point.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
The security advisory is available at:
https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
Thank you for your continued support of Asterisk!
----- 13.18.4 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert9, 13.18.4,
14.7.4 and 15.1.4.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-012: Remote Crash Vulnerability in RTCP Stack
If a compound RTCP packet is received containing more than
one report (for example a Receiver Report and a Sender
Report) the RTCP stack will incorrectly store report
information outside of allocated memory potentially causing
a crash.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-012.html
http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
Thank you for your continued support of Asterisk!
----- 13.18.3 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert8, 13.18.3,
14.7.3 and 15.1.3.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
If the chan_skinny (AKA SCCP protocol) channel driver is
flooded with certain requests it can cause the asterisk
process to use excessive amounts of virtual memory
eventually causing asterisk to stop processing requests of
any kind.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
Thank you for your continued support of Asterisk!
----- 13.18.2 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.18.2.
The release of Asterisk 13.18.2 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27391 - Regression: Deadlock between AOR named lock
and pjproject grp lock
(Reported by shaurya jain)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.2
Thank you for your continued support of Asterisk!
----- 13.18.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.18.0.
The release of Asterisk 13.18.0 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-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)
Bugs fixed in this release:
-----------------------------------
* 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 Benoît 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 Engström)
* 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 Seán 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-27093 - ODBC deadlocks when app_directory tries to
play back non-existent voicemail greeting
(Reported by James Terhune)
New Features made in this release:
-----------------------------------
* 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)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.0
Thank you for your continued support of Asterisk!
----- 13.17.0 ----
The Asterisk Development Team would like to announce the release
of Asterisk 13.17.0.
The release of Asterisk 13.17.0 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:
Bugs fixed in this release:
-----------------------------------
* 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-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-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-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-27060 - Comment typo format_g729.c
(Reported by Matthew Fredrickson)
* ASTERISK-27026 - res_ari: Crash when no ari.conf
configuration file exists
(Reported by Ronald Raikes)
* 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-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
get_write_timeout
(Reported by Jørgen 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 Engström)
* ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
when request and To URI differ
(Reported by Yasin CANER)
* ASTERISK-26789 - Audit manipulation of channel flags without
locks
(Reported by Joshua Colp)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
6869i)
(Reported by Matthias Binder)
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-26124 - res_agi: Set audio format for EAGI audio
stream
(Reported by John Fawcett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0
Thank you for your continued support of Asterisk!
2018-01-23 09:26:08 +01:00
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@comment $NetBSD: PLIST,v 1.11 2018/01/23 08:26:08 jnemeth Exp $
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2015-12-06 00:29:05 +01:00
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include/asterisk.h
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include/asterisk/_private.h
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include/asterisk/abstract_jb.h
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include/asterisk/acl.h
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include/asterisk/adsi.h
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include/asterisk/ael_structs.h
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include/asterisk/agi.h
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include/asterisk/alaw.h
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2017-06-04 09:51:27 +02:00
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include/asterisk/alertpipe.h
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2015-12-06 00:29:05 +01:00
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include/asterisk/aoc.h
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include/asterisk/app.h
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include/asterisk/ari.h
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include/asterisk/ast_expr.h
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include/asterisk/ast_version.h
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include/asterisk/astdb.h
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include/asterisk/astmm.h
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include/asterisk/astobj.h
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include/asterisk/astobj2.h
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include/asterisk/astosp.h
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include/asterisk/audiohook.h
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include/asterisk/autochan.h
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include/asterisk/autoconfig.h
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include/asterisk/backtrace.h
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include/asterisk/beep.h
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include/asterisk/bridge.h
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include/asterisk/bridge_after.h
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include/asterisk/bridge_basic.h
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include/asterisk/bridge_channel.h
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include/asterisk/bridge_channel_internal.h
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include/asterisk/bridge_features.h
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include/asterisk/bridge_internal.h
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include/asterisk/bridge_roles.h
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include/asterisk/bridge_technology.h
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include/asterisk/bucket.h
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include/asterisk/build.h
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include/asterisk/buildinfo.h
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include/asterisk/buildopts.h
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include/asterisk/calendar.h
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include/asterisk/callerid.h
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include/asterisk/causes.h
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include/asterisk/ccss.h
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include/asterisk/cdr.h
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include/asterisk/cel.h
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include/asterisk/celt.h
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include/asterisk/channel.h
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include/asterisk/channel_internal.h
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include/asterisk/channelstate.h
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include/asterisk/chanvars.h
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include/asterisk/cli.h
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include/asterisk/codec.h
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include/asterisk/compat.h
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include/asterisk/compiler.h
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include/asterisk/config.h
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include/asterisk/config_options.h
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include/asterisk/core_local.h
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include/asterisk/core_unreal.h
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include/asterisk/crypto.h
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include/asterisk/data.h
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include/asterisk/datastore.h
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include/asterisk/devicestate.h
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include/asterisk/dial.h
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include/asterisk/dlinkedlists.h
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include/asterisk/dns.h
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include/asterisk/dnsmgr.h
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include/asterisk/doxygen/architecture.h
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include/asterisk/doxygen/licensing.h
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include/asterisk/doxyref.h
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include/asterisk/dsp.h
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include/asterisk/dundi.h
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include/asterisk/endian.h
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include/asterisk/endpoints.h
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include/asterisk/enum.h
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include/asterisk/event.h
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include/asterisk/event_defs.h
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include/asterisk/extconf.h
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include/asterisk/features.h
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include/asterisk/features_config.h
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include/asterisk/file.h
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include/asterisk/format.h
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include/asterisk/format_cache.h
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include/asterisk/format_cap.h
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include/asterisk/format_compatibility.h
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include/asterisk/frame.h
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include/asterisk/framehook.h
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include/asterisk/fskmodem.h
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include/asterisk/fskmodem_float.h
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include/asterisk/fskmodem_int.h
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include/asterisk/global_datastores.h
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include/asterisk/hashtab.h
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include/asterisk/heap.h
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include/asterisk/http.h
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include/asterisk/http_websocket.h
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include/asterisk/image.h
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include/asterisk/indications.h
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include/asterisk/inline_api.h
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include/asterisk/io.h
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include/asterisk/json.h
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include/asterisk/linkedlists.h
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include/asterisk/localtime.h
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include/asterisk/lock.h
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include/asterisk/logger.h
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include/asterisk/manager.h
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include/asterisk/max_forwards.h
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include/asterisk/md5.h
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include/asterisk/media_index.h
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include/asterisk/message.h
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include/asterisk/mixmonitor.h
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include/asterisk/mod_format.h
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include/asterisk/module.h
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include/asterisk/monitor.h
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2016-07-24 08:35:50 +02:00
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include/asterisk/multicast_rtp.h
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2015-12-06 00:29:05 +01:00
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include/asterisk/musiconhold.h
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2016-06-09 06:41:48 +02:00
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include/asterisk/named_locks.h
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2015-12-06 00:29:05 +01:00
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include/asterisk/netsock.h
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include/asterisk/netsock2.h
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include/asterisk/network.h
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include/asterisk/optional_api.h
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include/asterisk/options.h
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include/asterisk/opus.h
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include/asterisk/parking.h
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include/asterisk/paths.h
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include/asterisk/pbx.h
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include/asterisk/phoneprov.h
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include/asterisk/pickup.h
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include/asterisk/pktccops.h
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include/asterisk/plc.h
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include/asterisk/poll-compat.h
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include/asterisk/presencestate.h
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include/asterisk/privacy.h
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include/asterisk/pval.h
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include/asterisk/res_fax.h
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include/asterisk/res_hep.h
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include/asterisk/res_mwi_external.h
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include/asterisk/res_odbc.h
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2016-05-06 09:41:06 +02:00
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include/asterisk/res_odbc_transaction.h
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include/asterisk/res_pjproject.h
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2015-12-06 00:29:05 +01:00
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include/asterisk/res_pjsip.h
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include/asterisk/res_pjsip_body_generator_types.h
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include/asterisk/res_pjsip_cli.h
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include/asterisk/res_pjsip_outbound_publish.h
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include/asterisk/res_pjsip_presence_xml.h
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include/asterisk/res_pjsip_pubsub.h
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include/asterisk/res_pjsip_session.h
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include/asterisk/res_srtp.h
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include/asterisk/rtp_engine.h
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include/asterisk/say.h
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include/asterisk/sched.h
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include/asterisk/sdp_srtp.h
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include/asterisk/security_events.h
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include/asterisk/security_events_defs.h
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include/asterisk/select.h
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include/asterisk/sem.h
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include/asterisk/sha1.h
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include/asterisk/silk.h
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include/asterisk/sip_api.h
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include/asterisk/slin.h
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include/asterisk/slinfactory.h
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include/asterisk/smdi.h
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include/asterisk/smoother.h
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include/asterisk/sorcery.h
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include/asterisk/sounds_index.h
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include/asterisk/speech.h
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include/asterisk/spinlock.h
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include/asterisk/srv.h
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include/asterisk/stasis.h
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include/asterisk/stasis_app.h
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include/asterisk/stasis_app_device_state.h
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include/asterisk/stasis_app_impl.h
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include/asterisk/stasis_app_mailbox.h
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include/asterisk/stasis_app_playback.h
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include/asterisk/stasis_app_recording.h
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include/asterisk/stasis_app_snoop.h
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include/asterisk/stasis_bridges.h
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include/asterisk/stasis_cache_pattern.h
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include/asterisk/stasis_channels.h
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include/asterisk/stasis_endpoints.h
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include/asterisk/stasis_internal.h
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include/asterisk/stasis_message_router.h
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include/asterisk/stasis_system.h
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include/asterisk/stasis_test.h
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include/asterisk/statsd.h
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include/asterisk/stringfields.h
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include/asterisk/strings.h
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include/asterisk/stun.h
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include/asterisk/syslog.h
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include/asterisk/taskprocessor.h
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include/asterisk/tcptls.h
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include/asterisk/tdd.h
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include/asterisk/term.h
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include/asterisk/test.h
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include/asterisk/threadpool.h
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include/asterisk/threadstorage.h
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include/asterisk/time.h
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include/asterisk/timing.h
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include/asterisk/transcap.h
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include/asterisk/translate.h
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include/asterisk/udptl.h
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include/asterisk/ulaw.h
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include/asterisk/unaligned.h
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include/asterisk/uri.h
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include/asterisk/utils.h
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include/asterisk/uuid.h
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include/asterisk/vector.h
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include/asterisk/version.h
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include/asterisk/xml.h
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include/asterisk/xmldoc.h
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include/asterisk/xmpp.h
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lib/asterisk/modules/app_adsiprog.so
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lib/asterisk/modules/app_agent_pool.so
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lib/asterisk/modules/app_alarmreceiver.so
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lib/asterisk/modules/app_amd.so
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lib/asterisk/modules/app_authenticate.so
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lib/asterisk/modules/app_bridgewait.so
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lib/asterisk/modules/app_cdr.so
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lib/asterisk/modules/app_celgenuserevent.so
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lib/asterisk/modules/app_chanisavail.so
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lib/asterisk/modules/app_channelredirect.so
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lib/asterisk/modules/app_chanspy.so
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lib/asterisk/modules/app_confbridge.so
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lib/asterisk/modules/app_controlplayback.so
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lib/asterisk/modules/app_db.so
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lib/asterisk/modules/app_dial.so
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lib/asterisk/modules/app_dictate.so
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lib/asterisk/modules/app_directed_pickup.so
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lib/asterisk/modules/app_directory.so
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lib/asterisk/modules/app_disa.so
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lib/asterisk/modules/app_dumpchan.so
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lib/asterisk/modules/app_echo.so
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lib/asterisk/modules/app_exec.so
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lib/asterisk/modules/app_externalivr.so
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lib/asterisk/modules/app_festival.so
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lib/asterisk/modules/app_followme.so
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lib/asterisk/modules/app_forkcdr.so
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lib/asterisk/modules/app_getcpeid.so
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lib/asterisk/modules/app_ices.so
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lib/asterisk/modules/app_image.so
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lib/asterisk/modules/app_macro.so
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lib/asterisk/modules/app_milliwatt.so
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lib/asterisk/modules/app_minivm.so
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lib/asterisk/modules/app_mixmonitor.so
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lib/asterisk/modules/app_morsecode.so
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lib/asterisk/modules/app_mp3.so
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lib/asterisk/modules/app_nbscat.so
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lib/asterisk/modules/app_originate.so
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lib/asterisk/modules/app_page.so
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lib/asterisk/modules/app_playback.so
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lib/asterisk/modules/app_playtones.so
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lib/asterisk/modules/app_privacy.so
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lib/asterisk/modules/app_queue.so
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lib/asterisk/modules/app_read.so
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lib/asterisk/modules/app_readexten.so
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lib/asterisk/modules/app_record.so
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lib/asterisk/modules/app_sayunixtime.so
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lib/asterisk/modules/app_senddtmf.so
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lib/asterisk/modules/app_sendtext.so
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lib/asterisk/modules/app_sms.so
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lib/asterisk/modules/app_softhangup.so
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lib/asterisk/modules/app_speech_utils.so
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lib/asterisk/modules/app_stack.so
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lib/asterisk/modules/app_stasis.so
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lib/asterisk/modules/app_system.so
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lib/asterisk/modules/app_talkdetect.so
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lib/asterisk/modules/app_test.so
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lib/asterisk/modules/app_transfer.so
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lib/asterisk/modules/app_url.so
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lib/asterisk/modules/app_userevent.so
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lib/asterisk/modules/app_verbose.so
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lib/asterisk/modules/app_voicemail.so
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lib/asterisk/modules/app_waitforring.so
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lib/asterisk/modules/app_waitforsilence.so
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lib/asterisk/modules/app_waituntil.so
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lib/asterisk/modules/app_while.so
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lib/asterisk/modules/app_zapateller.so
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lib/asterisk/modules/bridge_builtin_features.so
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lib/asterisk/modules/bridge_builtin_interval_features.so
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lib/asterisk/modules/bridge_holding.so
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lib/asterisk/modules/bridge_native_rtp.so
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lib/asterisk/modules/bridge_simple.so
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lib/asterisk/modules/bridge_softmix.so
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${PLIST.unixodbc}lib/asterisk/modules/cdr_adaptive_odbc.so
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lib/asterisk/modules/cdr_csv.so
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lib/asterisk/modules/cdr_custom.so
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lib/asterisk/modules/cdr_manager.so
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${PLIST.unixodbc}lib/asterisk/modules/cdr_odbc.so
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${PLIST.pgsql}lib/asterisk/modules/cdr_pgsql.so
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lib/asterisk/modules/cdr_sqlite3_custom.so
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lib/asterisk/modules/cdr_syslog.so
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lib/asterisk/modules/cel_custom.so
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lib/asterisk/modules/cel_manager.so
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${PLIST.unixodbc}lib/asterisk/modules/cel_odbc.so
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${PLIST.pgsql}lib/asterisk/modules/cel_pgsql.so
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lib/asterisk/modules/cel_sqlite3_custom.so
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lib/asterisk/modules/chan_bridge_media.so
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lib/asterisk/modules/chan_iax2.so
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${PLIST.mgcp}lib/asterisk/modules/chan_mgcp.so
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${PLIST.jabber}lib/asterisk/modules/chan_motif.so
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lib/asterisk/modules/chan_oss.so
|
2016-09-23 19:50:19 +02:00
|
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lib/asterisk/modules/chan_rtp.so
|
2015-12-06 00:29:05 +01:00
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lib/asterisk/modules/chan_sip.so
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lib/asterisk/modules/chan_skinny.so
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lib/asterisk/modules/chan_unistim.so
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lib/asterisk/modules/codec_a_mu.so
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lib/asterisk/modules/codec_adpcm.so
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lib/asterisk/modules/codec_alaw.so
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lib/asterisk/modules/codec_g722.so
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lib/asterisk/modules/codec_g726.so
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lib/asterisk/modules/codec_gsm.so
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lib/asterisk/modules/codec_ilbc.so
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lib/asterisk/modules/codec_lpc10.so
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lib/asterisk/modules/codec_resample.so
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${PLIST.speex}lib/asterisk/modules/codec_speex.so
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lib/asterisk/modules/codec_ulaw.so
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lib/asterisk/modules/format_g719.so
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lib/asterisk/modules/format_g723.so
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lib/asterisk/modules/format_g726.so
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lib/asterisk/modules/format_g729.so
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lib/asterisk/modules/format_gsm.so
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lib/asterisk/modules/format_h263.so
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lib/asterisk/modules/format_h264.so
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lib/asterisk/modules/format_ilbc.so
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lib/asterisk/modules/format_jpeg.so
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lib/asterisk/modules/format_pcm.so
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lib/asterisk/modules/format_siren14.so
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lib/asterisk/modules/format_siren7.so
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lib/asterisk/modules/format_sln.so
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lib/asterisk/modules/format_vox.so
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lib/asterisk/modules/format_wav.so
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lib/asterisk/modules/format_wav_gsm.so
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lib/asterisk/modules/func_aes.so
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lib/asterisk/modules/func_audiohookinherit.so
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lib/asterisk/modules/func_base64.so
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lib/asterisk/modules/func_blacklist.so
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lib/asterisk/modules/func_callcompletion.so
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lib/asterisk/modules/func_callerid.so
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lib/asterisk/modules/func_cdr.so
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lib/asterisk/modules/func_channel.so
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lib/asterisk/modules/func_config.so
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lib/asterisk/modules/func_curl.so
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lib/asterisk/modules/func_cut.so
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lib/asterisk/modules/func_db.so
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lib/asterisk/modules/func_devstate.so
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lib/asterisk/modules/func_dialgroup.so
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lib/asterisk/modules/func_dialplan.so
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lib/asterisk/modules/func_enum.so
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lib/asterisk/modules/func_env.so
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lib/asterisk/modules/func_extstate.so
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lib/asterisk/modules/func_frame_trace.so
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lib/asterisk/modules/func_global.so
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lib/asterisk/modules/func_groupcount.so
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lib/asterisk/modules/func_hangupcause.so
|
Update Asterisk to 13.7.2: this is mainly bug fixes with some minor
features and fixes for AST-2016-001, AST-2016-002, and AST-2016-003.
Also some pkglinting.
----- 13.7.2
The Asterisk Development Team has announced the release of Asterisk 13.7.2.
The release of Asterisk 13.7.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25702 - PjSip realtime DB and Cache Errors since
upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
Nic Colledge)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.2
Thank you for your continued support of Asterisk!
----- 13.7.1
The Asterisk Development Team has announced security releases for Certified
Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
The release of these versions resolves the following security vulnerabilities:
* AST-2016-001: BEAST vulnerability in HTTP server
The Asterisk HTTP server currently has a default configuration which allows
the BEAST vulnerability to be exploited if the TLS functionality is enabled.
This can allow a man-in-the-middle attack to decrypt data passing through it.
* AST-2016-002: File descriptor exhaustion in chan_sip
Setting the sip.conf timert1 value to a value higher than 1245 can cause an
integer overflow and result in large retransmit timeout times. These large
timeout values hold system file descriptors hostage and can cause the system
to run out of file descriptors.
* AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
If no UDPTL packets are lost there is no problem. However, a lost packet
causes Asterisk to use the available error correcting redundancy packets. If
those redundancy packets have zero length then Asterisk uses an uninitialized
buffer pointer and length value which can cause invalid memory accesses later
when the packet is copied.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
* http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
Thank you for your continued support of Asterisk!
----- 13.7.0
The Asterisk Development Team has announced the release of Asterisk 13.7.0.
The release of Asterisk 13.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-25419 - Dialplan Application for Integration of StatsD
(Reported by Ashley Sanders)
* ASTERISK-25549 - Confbridge: Add participant timeout option
(Reported by Mark Michelson)
* ASTERISK-24922 - ARI: Add the ability to intercept hold and
raise an event (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25689 - pjsip show contacts not working in Asterisk
13.7rc2 (Reported by Marcelo Terres)
* ASTERISK-25640 - pbx: Deadlock on features reload and state
change hint. (Reported by Krzysztof Trempala)
* ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
(Reported by Corey Farrell)
* ASTERISK-25601 - json: Audit reference usage and thread safety
(Reported by Joshua Colp)
* ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
caching (Reported by Joshua Colp)
* ASTERISK-25615 - res_pjsip: Setting transport async_operations >
1 causes segfault on tls transports (Reported by George Joseph)
* ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
thread of asterisk is not released (Reported by Hiroaki Komatsu)
* ASTERISK-25619 - res_chan_stats not sending the correct
information to StatsD (Reported by Tyler Cambron)
* ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
Corey Farrell)
* ASTERISK-25609 - [patch]Asterisk may crash when calling
ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
* ASTERISK-24146 - [patch]No audio on WebRtc caller side when
answer waiting time is more than ~7sec (Reported by Aleksei
Kulakov)
* ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
(Reported by Alexander Traud)
* ASTERISK-25616 - Warning with a Codec Module which supports PLC
with FEC (Reported by Alexander Traud)
* ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
Dudás József)
* ASTERISK-25608 - res_pjsip/contacts/statsd: Lifecycle events
aren't consistent (Reported by George Joseph)
* ASTERISK-25584 - [patch] format-attribute module: VP8 missing
(Reported by Alexander Traud)
* ASTERISK-25583 - [patch] format-attribute module: RFC 7587 (Opus
Codec) (Reported by Alexander Traud)
* ASTERISK-25498 - Asterisk crashes when negotiating g729 without
that module installed (Reported by Ben Langfeld)
* ASTERISK-25595 - Unescaped : in messge sent to statsd (Reported
by Niklas Larsson)
* ASTERISK-25476 - chan_sip loses registrations after a while
(Reported by Michael Keuter)
* ASTERISK-25598 - res_pjsip: Contact status messages are
printing a hash instead of the uri (Reported by George Joseph)
* ASTERISK-25600 - bridging: Inconsistency in BRIDGEPEER (Reported
by Jonathan Rose)
* ASTERISK-25582 - Testsuite: Reactor timeout error in
tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
Jordan)
* ASTERISK-25593 - fastagi: record file closed after sending
result (Reported by Kevin Harwell)
* ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
it's assumed to (Reported by Walter Doekes)
* ASTERISK-25590 - CLI Usage info for 'pjsip send notify'
references incorrect config (Reported by Corey Farrell)
* ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
(Reported by Corey Farrell)
* ASTERISK-25575 - res_pjsip: Dynamic outbound registrations
created via ARI are not loaded into memory on Asterisk
start/restart (Reported by Matt Jordan)
* ASTERISK-25545 - [patch] translation module gets cached not
joint format (Reported by Alexander Traud)
* ASTERISK-25573 - [patch] H.264 format attribute module: resets
whole SDP (Reported by Alexander Traud)
* ASTERISK-24958 - Forwarding loop detection inhibits certain
desirable scenarios (Reported by Mark Michelson)
* ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex
'qe->chan' freed more times than we've locked! (Reported by Alec
Davis)
* ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
Joshua Colp)
* ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missing
when called internally (Reported by Alexander Traud)
* ASTERISK-25535 - [patch] format creation on module load instead
of cache (Reported by Alexander Traud)
* ASTERISK-25449 - main/sched: Regression introduced by
5c713fdf18f causes erroneous duplicate RTCP messages; other
potential scheduling issues in chan_sip/chan_skinny (Reported by
Matt Jordan)
* ASTERISK-25546 - threadpool: Race condition between idle timeout
and activation (Reported by Joshua Colp)
* ASTERISK-25537 - [patch] format-attribute module: RFC or
internal defaults? (Reported by Alexander Traud)
* ASTERISK-25533 - [patch] buffer for ast_format_cap_get_names
only 64 bytes (Reported by Alexander Traud)
* ASTERISK-25373 - add documentation for CALLERID(pres) and also
the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
Doekes)
* ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
Walter Doekes)
* ASTERISK-24779 - Passthrough OPUS codec not working with
chan_pjsip (Reported by PowerPBX)
* ASTERISK-25522 - ARI: Crash when creating channel via ARI
originate with requesting channel (Reported by Matt Jordan)
* ASTERISK-25434 - Compiler flags not reported in 'core show
settings' despite usage during compilation (Reported by Rusty
Newton)
* ASTERISK-24106 - WebSockets Automatically decides what driver it
will use (Reported by Andrew Nagy)
* ASTERISK-25513 - Crash: malloc failed with high load of
subscriptions. (Reported by John Bigelow)
* ASTERISK-25505 - res_pjsip_pubsub: Crash on off-nominal when UAS
dialog can't be created (Reported by Joshua Colp)
* ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
possible codecs configured for peer as opposed to intersection
of configured codecs and offered codecs (Reported by Taylor
Hawkes)
* ASTERISK-25494 - build: GCC 5.1.x catches some new const, array
bounds and missing paren issues (Reported by George Joseph)
* ASTERISK-25485 - res_pjsip_outbound_registration: registration
stops due to 400 response (Reported by Kevin Harwell)
* ASTERISK-25486 - res_pjsip: Fix deadlock when validating URIs
(Reported by Joshua Colp)
* ASTERISK-7803 - [patch] Update the maximum packetization values
in frame.c (Reported by dea)
* ASTERISK-25484 - [patch] autoframing=yes has no effect (Reported
by Alexander Traud)
* ASTERISK-25461 - Nested dialplan #includes don't work as
expected. (Reported by Richard Mudgett)
* ASTERISK-25455 - Deadlock of PJSIP realtime over
res_config_pgsql (Reported by mdu113)
* ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
(Reported by Olle Johansson)
* ASTERISK-25435 - Asterisk periodically hangs. UDP Recv-Q greatly
exceeds zero. (Reported by Dmitriy Serov)
* ASTERISK-25451 - Broken video - erased rtp marker bit (Reported
by Stefan Engström)
* ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
exist in AstDB (Reported by Andrew Nagy)
* ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
parsing (Reported by ffs)
* ASTERISK-25404 - segfault/crash in chan_pjsip_hangup ... at
chan_pjsip.c (Reported by Chet Stevens)
* ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
(Reported by Bojan Nemčić)
* ASTERISK-25441 - Deadlock in res_sorcery_memory_cache. (Reported
by Richard Mudgett)
* ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
ICE is not enabled (Reported by Joshua Colp)
Improvements made in this release:
-----------------------------------
* ASTERISK-25618 - res_pjsip: Check for readability of TLS files
at startup (Reported by George Joseph)
* ASTERISK-25572 - Endpoints: Add StatsD stats for Asterisk
endpoints (Reported by Matt Jordan)
* ASTERISK-25571 - PJSIP: Add StatsD stats for some common PJSIP
objects (Reported by Matt Jordan)
* ASTERISK-25518 - taskprocessor: Add high water mark (Reported by
Jonathan Rose)
* ASTERISK-25477 - pjsip show "command" like [criteria] (Reported
by Bryant Zimmerman)
* ASTERISK-24718 - [patch]Add inital support of "sanitize" to
configure (Reported by Badalian Vyacheslav)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0
Thank you for your continued support of Asterisk!
2016-02-07 10:13:34 +01:00
|
|
|
lib/asterisk/modules/func_holdintercept.so
|
2015-12-06 00:29:05 +01:00
|
|
|
lib/asterisk/modules/func_iconv.so
|
|
|
|
|
lib/asterisk/modules/func_jitterbuffer.so
|
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|
lib/asterisk/modules/func_lock.so
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lib/asterisk/modules/func_logic.so
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lib/asterisk/modules/func_math.so
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lib/asterisk/modules/func_md5.so
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lib/asterisk/modules/func_module.so
|
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|
${PLIST.unixodbc}lib/asterisk/modules/func_odbc.so
|
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|
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|
lib/asterisk/modules/func_periodic_hook.so
|
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lib/asterisk/modules/func_pitchshift.so
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lib/asterisk/modules/func_presencestate.so
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lib/asterisk/modules/func_rand.so
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lib/asterisk/modules/func_realtime.so
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lib/asterisk/modules/func_sha1.so
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lib/asterisk/modules/func_shell.so
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lib/asterisk/modules/func_sorcery.so
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${PLIST.speex}lib/asterisk/modules/func_speex.so
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lib/asterisk/modules/func_sprintf.so
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lib/asterisk/modules/func_srv.so
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lib/asterisk/modules/func_strings.so
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lib/asterisk/modules/func_sysinfo.so
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lib/asterisk/modules/func_talkdetect.so
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lib/asterisk/modules/func_timeout.so
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lib/asterisk/modules/func_uri.so
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lib/asterisk/modules/func_version.so
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lib/asterisk/modules/func_vmcount.so
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lib/asterisk/modules/func_volume.so
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lib/asterisk/modules/pbx_ael.so
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lib/asterisk/modules/pbx_config.so
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lib/asterisk/modules/pbx_dundi.so
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lib/asterisk/modules/pbx_loopback.so
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lib/asterisk/modules/pbx_realtime.so
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lib/asterisk/modules/pbx_spool.so
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lib/asterisk/modules/res_adsi.so
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lib/asterisk/modules/res_ael_share.so
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lib/asterisk/modules/res_agi.so
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lib/asterisk/modules/res_ari.so
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lib/asterisk/modules/res_ari_applications.so
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lib/asterisk/modules/res_ari_asterisk.so
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lib/asterisk/modules/res_ari_bridges.so
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lib/asterisk/modules/res_ari_channels.so
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lib/asterisk/modules/res_ari_device_states.so
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lib/asterisk/modules/res_ari_endpoints.so
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lib/asterisk/modules/res_ari_events.so
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lib/asterisk/modules/res_ari_model.so
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lib/asterisk/modules/res_ari_playbacks.so
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lib/asterisk/modules/res_ari_recordings.so
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lib/asterisk/modules/res_ari_sounds.so
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lib/asterisk/modules/res_calendar.so
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lib/asterisk/modules/res_clialiases.so
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lib/asterisk/modules/res_clioriginate.so
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lib/asterisk/modules/res_config_curl.so
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|
${PLIST.ldap}lib/asterisk/modules/res_config_ldap.so
|
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|
${PLIST.unixodbc}lib/asterisk/modules/res_config_odbc.so
|
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|
${PLIST.pgsql}lib/asterisk/modules/res_config_pgsql.so
|
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lib/asterisk/modules/res_config_sqlite3.so
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lib/asterisk/modules/res_convert.so
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lib/asterisk/modules/res_crypto.so
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lib/asterisk/modules/res_curl.so
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lib/asterisk/modules/res_fax.so
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|
${PLIST.spandsp}lib/asterisk/modules/res_fax_spandsp.so
|
|
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|
lib/asterisk/modules/res_format_attr_celt.so
|
2016-10-27 03:08:17 +02:00
|
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|
lib/asterisk/modules/res_format_attr_g729.so
|
2015-12-06 00:29:05 +01:00
|
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|
lib/asterisk/modules/res_format_attr_h263.so
|
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lib/asterisk/modules/res_format_attr_h264.so
|
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lib/asterisk/modules/res_format_attr_opus.so
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|
lib/asterisk/modules/res_format_attr_silk.so
|
2016-09-23 19:50:19 +02:00
|
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lib/asterisk/modules/res_format_attr_siren14.so
|
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lib/asterisk/modules/res_format_attr_siren7.so
|
Update Asterisk to 13.7.2: this is mainly bug fixes with some minor
features and fixes for AST-2016-001, AST-2016-002, and AST-2016-003.
Also some pkglinting.
----- 13.7.2
The Asterisk Development Team has announced the release of Asterisk 13.7.2.
The release of Asterisk 13.7.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!
The following is the issue resolved in this release:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25702 - PjSip realtime DB and Cache Errors since
upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
Nic Colledge)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.2
Thank you for your continued support of Asterisk!
----- 13.7.1
The Asterisk Development Team has announced security releases for Certified
Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
The release of these versions resolves the following security vulnerabilities:
* AST-2016-001: BEAST vulnerability in HTTP server
The Asterisk HTTP server currently has a default configuration which allows
the BEAST vulnerability to be exploited if the TLS functionality is enabled.
This can allow a man-in-the-middle attack to decrypt data passing through it.
* AST-2016-002: File descriptor exhaustion in chan_sip
Setting the sip.conf timert1 value to a value higher than 1245 can cause an
integer overflow and result in large retransmit timeout times. These large
timeout values hold system file descriptors hostage and can cause the system
to run out of file descriptors.
* AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
If no UDPTL packets are lost there is no problem. However, a lost packet
causes Asterisk to use the available error correcting redundancy packets. If
those redundancy packets have zero length then Asterisk uses an uninitialized
buffer pointer and length value which can cause invalid memory accesses later
when the packet is copied.
For a full list of changes in the current releases, please see the ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
The security advisories are available at:
* http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
* http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
Thank you for your continued support of Asterisk!
----- 13.7.0
The Asterisk Development Team has announced the release of Asterisk 13.7.0.
The release of Asterisk 13.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-25419 - Dialplan Application for Integration of StatsD
(Reported by Ashley Sanders)
* ASTERISK-25549 - Confbridge: Add participant timeout option
(Reported by Mark Michelson)
* ASTERISK-24922 - ARI: Add the ability to intercept hold and
raise an event (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25689 - pjsip show contacts not working in Asterisk
13.7rc2 (Reported by Marcelo Terres)
* ASTERISK-25640 - pbx: Deadlock on features reload and state
change hint. (Reported by Krzysztof Trempala)
* ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
(Reported by Corey Farrell)
* ASTERISK-25601 - json: Audit reference usage and thread safety
(Reported by Joshua Colp)
* ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
caching (Reported by Joshua Colp)
* ASTERISK-25615 - res_pjsip: Setting transport async_operations >
1 causes segfault on tls transports (Reported by George Joseph)
* ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
thread of asterisk is not released (Reported by Hiroaki Komatsu)
* ASTERISK-25619 - res_chan_stats not sending the correct
information to StatsD (Reported by Tyler Cambron)
* ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
Corey Farrell)
* ASTERISK-25609 - [patch]Asterisk may crash when calling
ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
* ASTERISK-24146 - [patch]No audio on WebRtc caller side when
answer waiting time is more than ~7sec (Reported by Aleksei
Kulakov)
* ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
(Reported by Alexander Traud)
* ASTERISK-25616 - Warning with a Codec Module which supports PLC
with FEC (Reported by Alexander Traud)
* ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
Dudás József)
* ASTERISK-25608 - res_pjsip/contacts/statsd: Lifecycle events
aren't consistent (Reported by George Joseph)
* ASTERISK-25584 - [patch] format-attribute module: VP8 missing
(Reported by Alexander Traud)
* ASTERISK-25583 - [patch] format-attribute module: RFC 7587 (Opus
Codec) (Reported by Alexander Traud)
* ASTERISK-25498 - Asterisk crashes when negotiating g729 without
that module installed (Reported by Ben Langfeld)
* ASTERISK-25595 - Unescaped : in messge sent to statsd (Reported
by Niklas Larsson)
* ASTERISK-25476 - chan_sip loses registrations after a while
(Reported by Michael Keuter)
* ASTERISK-25598 - res_pjsip: Contact status messages are
printing a hash instead of the uri (Reported by George Joseph)
* ASTERISK-25600 - bridging: Inconsistency in BRIDGEPEER (Reported
by Jonathan Rose)
* ASTERISK-25582 - Testsuite: Reactor timeout error in
tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
Jordan)
* ASTERISK-25593 - fastagi: record file closed after sending
result (Reported by Kevin Harwell)
* ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
it's assumed to (Reported by Walter Doekes)
* ASTERISK-25590 - CLI Usage info for 'pjsip send notify'
references incorrect config (Reported by Corey Farrell)
* ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
(Reported by Corey Farrell)
* ASTERISK-25575 - res_pjsip: Dynamic outbound registrations
created via ARI are not loaded into memory on Asterisk
start/restart (Reported by Matt Jordan)
* ASTERISK-25545 - [patch] translation module gets cached not
joint format (Reported by Alexander Traud)
* ASTERISK-25573 - [patch] H.264 format attribute module: resets
whole SDP (Reported by Alexander Traud)
* ASTERISK-24958 - Forwarding loop detection inhibits certain
desirable scenarios (Reported by Mark Michelson)
* ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex
'qe->chan' freed more times than we've locked! (Reported by Alec
Davis)
* ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
Joshua Colp)
* ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missing
when called internally (Reported by Alexander Traud)
* ASTERISK-25535 - [patch] format creation on module load instead
of cache (Reported by Alexander Traud)
* ASTERISK-25449 - main/sched: Regression introduced by
5c713fdf18f causes erroneous duplicate RTCP messages; other
potential scheduling issues in chan_sip/chan_skinny (Reported by
Matt Jordan)
* ASTERISK-25546 - threadpool: Race condition between idle timeout
and activation (Reported by Joshua Colp)
* ASTERISK-25537 - [patch] format-attribute module: RFC or
internal defaults? (Reported by Alexander Traud)
* ASTERISK-25533 - [patch] buffer for ast_format_cap_get_names
only 64 bytes (Reported by Alexander Traud)
* ASTERISK-25373 - add documentation for CALLERID(pres) and also
the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
Doekes)
* ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
Walter Doekes)
* ASTERISK-24779 - Passthrough OPUS codec not working with
chan_pjsip (Reported by PowerPBX)
* ASTERISK-25522 - ARI: Crash when creating channel via ARI
originate with requesting channel (Reported by Matt Jordan)
* ASTERISK-25434 - Compiler flags not reported in 'core show
settings' despite usage during compilation (Reported by Rusty
Newton)
* ASTERISK-24106 - WebSockets Automatically decides what driver it
will use (Reported by Andrew Nagy)
* ASTERISK-25513 - Crash: malloc failed with high load of
subscriptions. (Reported by John Bigelow)
* ASTERISK-25505 - res_pjsip_pubsub: Crash on off-nominal when UAS
dialog can't be created (Reported by Joshua Colp)
* ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
possible codecs configured for peer as opposed to intersection
of configured codecs and offered codecs (Reported by Taylor
Hawkes)
* ASTERISK-25494 - build: GCC 5.1.x catches some new const, array
bounds and missing paren issues (Reported by George Joseph)
* ASTERISK-25485 - res_pjsip_outbound_registration: registration
stops due to 400 response (Reported by Kevin Harwell)
* ASTERISK-25486 - res_pjsip: Fix deadlock when validating URIs
(Reported by Joshua Colp)
* ASTERISK-7803 - [patch] Update the maximum packetization values
in frame.c (Reported by dea)
* ASTERISK-25484 - [patch] autoframing=yes has no effect (Reported
by Alexander Traud)
* ASTERISK-25461 - Nested dialplan #includes don't work as
expected. (Reported by Richard Mudgett)
* ASTERISK-25455 - Deadlock of PJSIP realtime over
res_config_pgsql (Reported by mdu113)
* ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
(Reported by Olle Johansson)
* ASTERISK-25435 - Asterisk periodically hangs. UDP Recv-Q greatly
exceeds zero. (Reported by Dmitriy Serov)
* ASTERISK-25451 - Broken video - erased rtp marker bit (Reported
by Stefan Engström)
* ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
exist in AstDB (Reported by Andrew Nagy)
* ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
parsing (Reported by ffs)
* ASTERISK-25404 - segfault/crash in chan_pjsip_hangup ... at
chan_pjsip.c (Reported by Chet Stevens)
* ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
(Reported by Bojan Nemčić)
* ASTERISK-25441 - Deadlock in res_sorcery_memory_cache. (Reported
by Richard Mudgett)
* ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
ICE is not enabled (Reported by Joshua Colp)
Improvements made in this release:
-----------------------------------
* ASTERISK-25618 - res_pjsip: Check for readability of TLS files
at startup (Reported by George Joseph)
* ASTERISK-25572 - Endpoints: Add StatsD stats for Asterisk
endpoints (Reported by Matt Jordan)
* ASTERISK-25571 - PJSIP: Add StatsD stats for some common PJSIP
objects (Reported by Matt Jordan)
* ASTERISK-25518 - taskprocessor: Add high water mark (Reported by
Jonathan Rose)
* ASTERISK-25477 - pjsip show "command" like [criteria] (Reported
by Bryant Zimmerman)
* ASTERISK-24718 - [patch]Add inital support of "sanitize" to
configure (Reported by Badalian Vyacheslav)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0
Thank you for your continued support of Asterisk!
2016-02-07 10:13:34 +01:00
|
|
|
lib/asterisk/modules/res_format_attr_vp8.so
|
2015-12-06 00:29:05 +01:00
|
|
|
lib/asterisk/modules/res_hep.so
|
|
|
|
|
lib/asterisk/modules/res_hep_rtcp.so
|
|
|
|
|
lib/asterisk/modules/res_http_websocket.so
|
|
|
|
|
lib/asterisk/modules/res_limit.so
|
|
|
|
|
lib/asterisk/modules/res_manager_devicestate.so
|
|
|
|
|
lib/asterisk/modules/res_manager_presencestate.so
|
|
|
|
|
lib/asterisk/modules/res_monitor.so
|
|
|
|
|
lib/asterisk/modules/res_musiconhold.so
|
|
|
|
|
lib/asterisk/modules/res_mutestream.so
|
|
|
|
|
${PLIST.unixodbc}lib/asterisk/modules/res_odbc.so
|
2016-05-06 09:41:06 +02:00
|
|
|
${PLIST.unixodbc}lib/asterisk/modules/res_odbc_transaction.so
|
2015-12-06 00:29:05 +01:00
|
|
|
lib/asterisk/modules/res_parking.so
|
|
|
|
|
lib/asterisk/modules/res_phoneprov.so
|
|
|
|
|
${PLIST.mgcp}lib/asterisk/modules/res_pktccops.so
|
|
|
|
|
lib/asterisk/modules/res_realtime.so
|
|
|
|
|
lib/asterisk/modules/res_rtp_asterisk.so
|
|
|
|
|
lib/asterisk/modules/res_rtp_multicast.so
|
|
|
|
|
lib/asterisk/modules/res_security_log.so
|
|
|
|
|
lib/asterisk/modules/res_smdi.so
|
|
|
|
|
${PLIST.snmp}lib/asterisk/modules/res_snmp.so
|
|
|
|
|
lib/asterisk/modules/res_sorcery_astdb.so
|
|
|
|
|
lib/asterisk/modules/res_sorcery_config.so
|
|
|
|
|
lib/asterisk/modules/res_sorcery_memory.so
|
|
|
|
|
lib/asterisk/modules/res_sorcery_memory_cache.so
|
|
|
|
|
lib/asterisk/modules/res_sorcery_realtime.so
|
|
|
|
|
lib/asterisk/modules/res_speech.so
|
|
|
|
|
${PLIST.srtp}lib/asterisk/modules/res_srtp.so
|
|
|
|
|
lib/asterisk/modules/res_stasis.so
|
|
|
|
|
lib/asterisk/modules/res_stasis_answer.so
|
|
|
|
|
lib/asterisk/modules/res_stasis_device_state.so
|
|
|
|
|
lib/asterisk/modules/res_stasis_playback.so
|
|
|
|
|
lib/asterisk/modules/res_stasis_recording.so
|
|
|
|
|
lib/asterisk/modules/res_stasis_snoop.so
|
|
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|
|
lib/asterisk/modules/res_statsd.so
|
|
|
|
|
lib/asterisk/modules/res_stun_monitor.so
|
|
|
|
|
${PLIST.kqueue}lib/asterisk/modules/res_timing_kqueue.so
|
|
|
|
|
lib/asterisk/modules/res_timing_pthread.so
|
|
|
|
|
${PLIST.jabber}lib/asterisk/modules/res_xmpp.so
|
|
|
|
|
lib/libasteriskssl.so
|
|
|
|
|
lib/libasteriskssl.so.1
|
|
|
|
|
lib/pkgconfig/asterisk.pc
|
|
|
|
|
libdata/asterisk/documentation/appdocsxml.dtd
|
|
|
|
|
libdata/asterisk/documentation/appdocsxml.xslt
|
|
|
|
|
libdata/asterisk/documentation/core-en_US.xml
|
|
|
|
|
libdata/asterisk/images/asterisk-intro.jpg
|
|
|
|
|
libdata/asterisk/images/kpad2.jpg
|
|
|
|
|
libdata/asterisk/moh/.asterisk-moh-opsound-wav-2.03
|
|
|
|
|
libdata/asterisk/moh/CHANGES-asterisk-moh-opsound-wav
|
|
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|
|
libdata/asterisk/moh/CREDITS-asterisk-moh-opsound-wav
|
|
|
|
|
libdata/asterisk/moh/LICENSE-asterisk-moh-opsound-wav
|
|
|
|
|
libdata/asterisk/moh/macroform-cold_day.wav
|
|
|
|
|
libdata/asterisk/moh/macroform-robot_dity.wav
|
|
|
|
|
libdata/asterisk/moh/macroform-the_simplicity.wav
|
|
|
|
|
libdata/asterisk/moh/manolo_camp-morning_coffee.wav
|
|
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|
|
libdata/asterisk/moh/reno_project-system.wav
|
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|
libdata/asterisk/phoneprov/000000000000-directory.xml
|
|
|
|
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libdata/asterisk/phoneprov/000000000000-phone.cfg
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libdata/asterisk/phoneprov/000000000000.cfg
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libdata/asterisk/phoneprov/polycom.xml
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libdata/asterisk/phoneprov/polycom_line.xml
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libdata/asterisk/phoneprov/snom-mac.xml
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libdata/asterisk/rest-api/applications.json
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libdata/asterisk/rest-api/asterisk.json
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libdata/asterisk/rest-api/bridges.json
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libdata/asterisk/rest-api/channels.json
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libdata/asterisk/rest-api/deviceStates.json
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libdata/asterisk/rest-api/endpoints.json
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libdata/asterisk/rest-api/events.json
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libdata/asterisk/rest-api/mailboxes.json
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libdata/asterisk/rest-api/playbacks.json
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libdata/asterisk/rest-api/recordings.json
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libdata/asterisk/rest-api/resources.json
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libdata/asterisk/rest-api/sounds.json
|
Update to Asterisk 13.15.0. This is mostly a bug fix release with a few
minor enhancements. 13.14.1 was released to fix AST-2017-001.
----- 13.15.0
The Asterisk Development Team would like to announce the release of
Asterisk 13.15.0.
The release of Asterisk 13.15.0 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:
*New Features made in this release:*
-----------------------------------
- [ASTERISK-26878 <https://issues.asterisk.org/jira/browse/ASTERISK-26878>]
- func_channel: Add ability to get the callid so dialplan has access to it.
(Reported by Richard Mudgett)
- [ASTERISK-26863 <https://issues.asterisk.org/jira/browse/ASTERISK-26863>]
- res_pjsip: Add endpoint identification scheme based on a configured SIP
header/value
(Reported by Matt Jordan)
- [ASTERISK-17428 <https://issues.asterisk.org/jira/browse/ASTERISK-17428>]
- [patch] Allow "Comedian Mail" branding to be removed
(Reported by John Covert)
*Bugs fixed in this release:*
-----------------------------------
- [ASTERISK-26851 <https://issues.asterisk.org/jira/browse/ASTERISK-26851>]
- res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
(Reported by Richard Begg)
- [ASTERISK-26897 <https://issues.asterisk.org/jira/browse/ASTERISK-26897>]
- chan_sip: Security vulnerability with client code header
(Reported by Alex Villacís Lasso)
- [ASTERISK-26916 <https://issues.asterisk.org/jira/browse/ASTERISK-26916>]
- res_pjsip: Excessive refcount reached on transport ao2 object
(Reported by Ross Beer)
- [ASTERISK-26705 <https://issues.asterisk.org/jira/browse/ASTERISK-26705>]
- libasteriskssl.so not found when asterisk is installed for the 1st time
(Reported by George Joseph)
- [ASTERISK-26850 <https://issues.asterisk.org/jira/browse/ASTERISK-26850>]
- res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field
in HEP packets
(Reported by Max Norba)
- [ASTERISK-26484 <https://issues.asterisk.org/jira/browse/ASTERISK-26484>]
- res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from'
argument.
(Reported by Vinod Dharashive)
- [ASTERISK-26776 <https://issues.asterisk.org/jira/browse/ASTERISK-26776>]
- res_pjsip_pubsub: Crash when generating xpidf content
(Reported by Andrew Green)
- [ASTERISK-26880 <https://issues.asterisk.org/jira/browse/ASTERISK-26880>]
- Asterisk crashes when multiple speex users join confbridge with pp_vad
and dtx enabled
(Reported by Kirsty Tyerman)
- [ASTERISK-26862 <https://issues.asterisk.org/jira/browse/ASTERISK-26862>]
- app_queue: Queue stops calling members with local interface after
forwarding in previous call
(Reported by Robert Mordec)
- [ASTERISK-26732 <https://issues.asterisk.org/jira/browse/ASTERISK-26732>]
- res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
(Reported by Dan Jenkins)
- [ASTERISK-26879 <https://issues.asterisk.org/jira/browse/ASTERISK-26879>]
- PJSIP external_media_address ignored if no local_net options are provided
(Reported by Matt Jordan)
- [ASTERISK-26867 <https://issues.asterisk.org/jira/browse/ASTERISK-26867>]
- autochan: Locking in a function ast_autochan_destroy() on destroyed
channel (after masquerade).
(Reported by Krzysztof Trempala)
- [ASTERISK-26869 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-26668>]
- core: Malformed pattern matching extension (various factors) results in
crash
(Reported by xrobau)
- [ASTERISK-26865 <https://issues.asterisk.org/jira/browse/ASTERISK-26865>]
- chan_iax2: Reload of iax peer results in loss of host address/port
(Reported by Richard Begg)
- [ASTERISK-26872 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-26717>]
- Document the fact that Asterisk HEP support only works with the PJSIP
channel driver
(Reported by Olivier Krief)
- [ASTERISK-26643 <https://issues.asterisk.org/jira/browse/ASTERISK-26643>]
- Extra new line in Device field of DeviceStateChange AMI Event after
restart of Asterisk
(Reported by Roman Bedros)
- [ASTERISK-25237 <https://issues.asterisk.org/jira/browse/ASTERISK-25237>]
- stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
(Reported by Smirnov Aleksey)
- [ASTERISK-26857 <https://issues.asterisk.org/jira/browse/ASTERISK-26857>]
- chan_pjsip: Dialplan function race condition
(Reported by Joshua Colp)
- [ASTERISK-26841 <https://issues.asterisk.org/jira/browse/ASTERISK-26841>]
- chan_sip: Call not cancelled after receiving a 422 response
(Reported by Jean Aunis - Prescom)
- [ASTERISK-26822 <https://issues.asterisk.org/jira/browse/ASTERISK-26822>]
- pjsip/cli_commands: pjsip show channelstats shows wrong codec
(Reported by Kevin Harwell)
- [ASTERISK-26685 <https://issues.asterisk.org/jira/browse/ASTERISK-26685>]
- res_pjsip: Crash when using IPv6 and Transport ws,wss
(Reported by Michael Balen)
- [ASTERISK-24562 <https://issues.asterisk.org/jira/browse/ASTERISK-24562>]
- app_voicemail: Cannot set fromstring on a per-mailbox basis
(Reported by Mark Scholten)
- [ASTERISK-26598 <https://issues.asterisk.org/jira/browse/ASTERISK-26598>]
- Saynumber is trying to get "and" from "digits/" subfolder
(Reported by Jonathan Harris)
- [ASTERISK-17067 <https://issues.asterisk.org/jira/browse/ASTERISK-17067>]
- Long lines in call files cause spurious syntax error
(Reported by Dave Olszewski)
- [ASTERISK-26796 <https://issues.asterisk.org/jira/browse/ASTERISK-26796>]
- res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
(Reported by Jørgen H)
- [ASTERISK-25628 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-26825>]
- pjsip.conf.sample: user_agent: still refers to branch 12
(Reported by Tzafrir Cohen)
- [ASTERISK-26823 <https://issues.asterisk.org/jira/browse/ASTERISK-26823>]
- PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not
exist
(Reported by Mark Michelson)
- [ASTERISK-26623 <https://issues.asterisk.org/jira/browse/ASTERISK-26623>]
- res_pjsip: Crash when calling PJSIPShowEndpoint
(Reported by Jørgen H)
- [ASTERISK-26808 <https://issues.asterisk.org/jira/browse/ASTERISK-26808>]
- res_pjsip_outbound_registration doesn't know about network change events
(Reported by George Joseph)
- [ASTERISK-26313 <https://issues.asterisk.org/jira/browse/ASTERISK-26313>]
- chan_sip : Asterisk restart seems to be required for changing encryption
option
(Reported by benasse)
- [ASTERISK-26781 <https://issues.asterisk.org/jira/browse/ASTERISK-26781>]
- bridge: Passing the 'p' (play tone) flag to Bridge() application results
in garbled audio
(Reported by Sean Bright)
- [ASTERISK-26782 <https://issues.asterisk.org/jira/browse/ASTERISK-26782>]
- res_pjsip: URI requirement for fields is not consistently documented and
error does not provide indication
(Reported by Peter Sokolov)
- [ASTERISK-26812 <https://issues.asterisk.org/jira/browse/ASTERISK-26812>]
- [patch] Fix download_externals To Allow The Use Of curl Or wget
(Reported by Michael L. Young)
- [ASTERISK-18271 <https://issues.asterisk.org/jira/browse/ASTERISK-18271>]
- Pattern matching with res_config_mysql extensions does not behave as
expected
(Reported by Charlie Smurthwaite)
- [ASTERISK-26669 <https://issues.asterisk.org/jira/browse/ASTERISK-26669>]
- PJSIP Segfault 13.13.1 (Bundled PJSIP)
(Reported by Nic Colledge)
- [ASTERISK-18731 <https://issues.asterisk.org/jira/browse/ASTERISK-18731>]
- [patch] DUNDi weight parameter not processed correctly
(Reported by Peter Racz)
- [ASTERISK-26580 <https://issues.asterisk.org/jira/browse/ASTERISK-26580>]
- [patch] Error during LDAP modify action when user unregisters
(Reported by Nicholas John Koch)
- [ASTERISK-26799 <https://issues.asterisk.org/jira/browse/ASTERISK-26799>]
- res_pjsip: Using an auth object for inbound and outbound authentication
fails.
(Reported by Richard Mudgett)
- [ASTERISK-26738 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-25893>]
- Function vmauthenticate accesses uninitialized memory
(Reported by Filip Jenicek)
- [ASTERISK-26802 <https://issues.asterisk.org/jira/browse/ASTERISK-26802>]
- [patch] Integrity Check Of PJSIP Download Fails
(Reported by Michael L. Young)
- [ASTERISK-15858 <https://issues.asterisk.org/jira/browse/ASTERISK-15858>]
- [patch] Fix query with double backslash in string literals and stop log
warnings
(Reported by Humberto Figuera)
- [ASTERISK-26057 <https://issues.asterisk.org/jira/browse/ASTERISK-26057>]
- res_config_sqlite3 uses incorrect query - unnecessary escape
(Reported by Stepan)
- [ASTERISK-23457 <https://issues.asterisk.org/jira/browse/ASTERISK-23457>]
- SQlite3: Realtime queue loading fails after PRAGMA query result
(Reported by Scott Griepentrog)
- [ASTERISK-26794 <https://issues.asterisk.org/jira/browse/ASTERISK-26794>]
- http: Crash on Reload Only in ast_tcptls_server_start
(Reported by Joshua Elson)
- [ASTERISK-26714 <https://issues.asterisk.org/jira/browse/ASTERISK-26714>]
- Phone default have not ringing on ARM
(Reported by Igor Goncharovsky)
- [ASTERISK-26696 <https://issues.asterisk.org/jira/browse/ASTERISK-26696>]
- pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on
subscription refresh
(Reported by Zach R)
- [ASTERISK-26756 <https://issues.asterisk.org/jira/browse/ASTERISK-26756>]
- res_pjsip_mwi: Asterisk does not terminate MWI subscription
(Reported by Carl Fortin)
- [ASTERISK-26109 <https://issues.asterisk.org/jira/browse/ASTERISK-26109>]
- Asterisk fails building with OpenSSL 1.1.0
(Reported by Tzafrir Cohen)
- [ASTERISK-26723 <https://issues.asterisk.org/jira/browse/ASTERISK-26723>]
- VoiceMailPlayMsg not playing messages via realtime
(Reported by Ryan Rittgarn)
- [ASTERISK-18286 <https://issues.asterisk.org/jira/browse/ASTERISK-18286>]
- [patch] 'Silence' is truncated in Record()
(Reported by var)
- [ASTERISK-26248 <https://issues.asterisk.org/jira/browse/ASTERISK-26248>]
- chan_pjsip: Error when calling PJSIP client with domain specified
(Reported by Norbert Varga)
- [ASTERISK-26788 <https://issues.asterisk.org/jira/browse/ASTERISK-26788>]
- core: Protect flags during ast_waitfor
(Reported by Joshua Colp)
- [ASTERISK-26115 <https://issues.asterisk.org/jira/browse/ASTERISK-26115>]
- pbx: AMI Originate ignore "failed" extension on call failure
(Reported by Nasir Iqbal)
- [ASTERISK-26785 <https://issues.asterisk.org/jira/browse/ASTERISK-26785>]
- configs/samples: The 'identify' entry is in the wrong section in
sorcery.conf.sample
(Reported by Torrey Searle)
- [ASTERISK-26772 <https://issues.asterisk.org/jira/browse/ASTERISK-26772>]
- Crash in srv.c on startup with pjsip
(Reported by nappsoft)
- [ASTERISK-26770 <https://issues.asterisk.org/jira/browse/ASTERISK-26770>]
- res_stasis_device_state: Duplicate subscriptions when multiple received
at same time
(Reported by Joshua Colp)
*Improvements made in this release:*
-----------------------------------
- [ASTERISK-26864 <https://issues.asterisk.org/jira/browse/ASTERISK-26864>]
- res_pjsip_session: Add support for overlap dialling
(Reported by Richard Begg)
- [ASTERISK-26846 <https://issues.asterisk.org/jira/browse/ASTERISK-26846>]
- chan_sip: Add rtcp-mux support
(Reported by Sean Bright)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.15.0
*Thank you for your continued support of Asterisk!*
----- 13.14.0
The Asterisk Development Team has announced the release of Asterisk 13.14.0.
The release of Asterisk 13.14.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-26630 - Make logging PJPROJECT messages a bit easier
(Reported by Richard Mudgett)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported
by nappsoft)
* 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-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in
wrong byte order on Intel platform when using slin codec
(Reported by Frankie Chin)
* 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 Jørgen 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)
Improvements made in this release:
-----------------------------------
* 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)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.14.0
Thank you for your continued support of Asterisk!
-----
2017-05-14 00:39:13 +02:00
|
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libdata/asterisk/scripts/ast_coredumper
|
|
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libdata/asterisk/scripts/ast_logescalator
|
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libdata/asterisk/scripts/ast_loggrabber
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libdata/asterisk/scripts/refcounter.py
|
update to Asterisk 13.19.0 -- this contains both security fixes
and general bug fixes, fixes AST-2017-005, AST-2017-006, AST-2017-007,
AST-2017-008, AST-2017-009, AST-2017-10, AST-2017-11, AST-2017-12,
AST-2017-13, and AST-2017-14 (note that a number of these only
pertain to PJSIP which isn't used in pkgsrc)
----- 13.19.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.19.0.
The release of Asterisk 13.19.0 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:
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-27413 - Add cache_media_frames debugging option.
(Reported by Richard Mudgett)
* ASTERISK-27206 - res_pjsip: No mechanism exists to limit
endpoint identification to IP only
(Reported by Ben Merrills)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27531 - Compiler optimizations can break module load
sequence.
(Reported by abelbeck)
* ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
Contact crashes asterisk
(Reported by Ross Beer)
* ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
read()
(Reported by Abhay Gupta)
* ASTERISK-25079 - AMI bridge of channels results in MOH not
destroyed and robotic audio on one channel
(Reported by Zane Conkle)
* ASTERISK-27490 - chan_console: 'set active' fails to work
(Reported by Tzafrir Cohen)
* ASTERISK-24756 - ConfBridge sound_muted does not work from
CLI or AMI
(Reported by Thomas Frederiksen)
* ASTERISK-25649 - Transfer application does not work with
Local channels - documentation misleading
(Reported by Ivan Ullmann)
* ASTERISK-25869 - chan_sip: "rejected because extension not
found" should be logged as a security event
(Reported by Brian J. Murrell)
* ASTERISK-27440 - Strictrtp has issues to qualify video rtp
streams
(Reported by Wim De Vlaminck)
* ASTERISK-24329 - Music On Hold announcement cuts intro of
music the first time it is played
(Reported by Thomas Frederiksen)
* ASTERISK-19657 - Coverity Report: Fix issues for error type
CHAR_IO
(Reported by Matt Jordan)
* ASTERISK-27175 - iax.conf demo peer is invalid
(Reported by Tzafrir Cohen)
* ASTERISK-27430 - README refers to security documents that do
not exist.
(Reported by Corey Farrell)
* ASTERISK-20281 - "core set verbose" behaves strangely, can't
alias it, cli.conf example broken
(Reported by Tim Ringenbach at Asteria Solutions Group)
* 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-27408 - Identify causes and fix
pjsip/resolver/srv/failover/in_dialog/transport_tcp
(Reported by Corey Farrell)
* ASTERISK-18411 - Queue members with hints for state_interface
get stuck in "In Use" state.
(Reported by Steven T. Wheeler)
* ASTERISK-26131 - chan_sip: Crash Asterisk (in
sip_request_call at chan_sip.c) by making a call to a single
character in a dot pattern match
(Reported by Dwayne Hubbard)
* ASTERISK-27475 - codec_opus requires libcurl
(Reported by Samuel For)
* ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
not applied on reload
(Reported by John Bigelow)
* ASTERISK-27465 - CLI Completion Not Working
(Reported by Ross Beer)
* ASTERISK-27460 - CDR: Deadlock using AMI Originate with
Variable CDR(amaflags)=...
(Reported by Richard Mudgett)
* ASTERISK-27453 - RTP: Blind transfer direct media scenario
results in one way audio.
(Reported by Richard Mudgett)
* ASTERISK-20643 - SIP ICE support - remove hardcoded
limitation on SDP size, make ICE support disabled by default in
SIP, maybe provide a better warning message
(Reported by Roy)
* ASTERISK-26980 - pjsip: Clean up WebRTC disables
(Reported by abelbeck)
* 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-23735 - Transcoding makes bad choice in high-rate
translations
(Reported by Richard Kenner)
* ASTERISK-27445 - ARI: Updating a bridge gives wrong error
message.
(Reported by Frank Durden)
* ASTERISK-24662 - [patch] column and row headers for Signed
Linear format variants in output of 'core show translation' are
ambiguous
(Reported by Rusty Newton)
* ASTERISK-27353 - H323 audio starts with a delay of 2
seconds.
(Reported by Marco Giordani)
* ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
media
(Reported by Kevin Harwell)
* ASTERISK-27437 - [patch] ICE: server-reflexive candidates
(srflx) with Dual-Stack.
(Reported by Alexander Traud)
* ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
IPv6 addresses.
(Reported by Alexander Traud)
* ASTERISK-27435 - [patch] configure:
pjsip_evsub_set_uas_timeout not found.
(Reported by Alexander Traud)
* ASTERISK-27431 - Asterisk fails to build when openssl headers
are not installed.
(Reported by Corey Farrell)
* ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
(Reported by Ivan Larionov)
* ASTERISK-27421 - RTP source learning not working with devices
that have some clock issues
(Reported by nappsoft)
* ASTERISK-27361 - Attended transfer crashes in Asterisk
13.17.2
(Reported by Alessandro Pimenta)
* ASTERISK-27238 - Bridging: Crash freeing a frame that's
already been freed
(Reported by Richard Kenner)
* ASTERISK-27412 - core: Audiohook freeing interpolated frame
when it shouldn't.
(Reported by Mikhail)
* ASTERISK-27423 - app_record: We set the RECORD_STATUS
channel variable before closing the file
(Reported by George Joseph)
* ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
insert same ip address in "source ip address" and "destination
ip address" fields in HEP packets
(Reported by Max Norba)
* ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
is equal to RemoteAddress)
(Reported by Vasilii Rogin)
* ASTERISK-27415 - asterisk.conf: Setting astctl without
setting astrundir is ineffective.
(Reported by Corey Farrell)
* 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-27393 - res_pjsip: Crash occurs when an empty
contact read from astdb or database
(Reported by Aaron An)
* ASTERISK-27290 - res_pjsip: PIDF contact field has
malformed/invalid XML
(Reported by basildane)
* ASTERISK-27032 - res_pjsip: TLS options do not handle empty
values
(Reported by seanchann.zhou)
* ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
enabled but not configured.
(Reported by Alexander Traud)
* ASTERISK-26426 - format_ogg_opus: remove from source
(Reported by Kevin Harwell)
* ASTERISK-27378 - Modules: Fix issues with CLI completion.
(Reported by Corey Farrell)
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27390 - Audit menuselect module dependencies
(Reported by Corey Farrell)
* ASTERISK-27389 - Optional API modules should not allow
unload.
(Reported by Corey Farrell)
* ASTERISK-27369 - Bridge() dialplan application fails without
setting BRIDGERESULT channel variable
(Reported by James Terhune)
* ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
documentation
(Reported by Igor Goncharovsky)
* ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
'imap_delete_old_greeting'
(Reported by Anthony Messina)
* ASTERISK-27194 - jitterbuffer: Does not handle case where
translator returns null frame.
(Reported by Joshua Elson)
* ASTERISK-26639 - core: Disabling xmldoc support does not
work. Also results in abort during Asterisk startup.
(Reported by Mr Dini)
* ASTERISK-27372 - ARI: Node ARI client broken in latest
versions of 13 and 14
(Reported by Benjamin Keith Ford)
* ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
absence of the Expires header field with an unsubscribe action.
(Reported by Jonathan Cloots)
* ASTERISK-25960 - The config_hook unit test causes Asterisk to
crash if run a second time
(Reported by George Joseph)
* ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
when rtp_ipv6 set to yes
(Reported by Martin Cisárik)
* ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
curl is loaded
(Reported by Ronald Raikes)
* ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
but first on SDP media level.
(Reported by Alexander Traud)
* ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
Assertion on un/re-load: mod.id == -1
(Reported by Tzafrir Cohen)
* ASTERISK-23462 - Cannot disable SIP debugging via CLI after
enabling with conf file option - also 'sip set debug off'
reports debugging disabled, when it really isn't
(Reported by Rusty Newton)
* ASTERISK-27328 - Missing openssl dependencies in
res_rtp_asterisk and tcptls
(Reported by Tzafrir Cohen)
* ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
(o=) contains local address.
(Reported by Alexander Traud)
* ASTERISK-27343 - Fails to build in FreeBSD due to
sys/sysmacros.h not existing there
(Reported by Guido Falsi)
* ASTERISK-27340 - backtrace.c: Crash due to double-free.
(Reported by Corey Farrell)
* ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
stopping.
(Reported by Alexander Traud)
* ASTERISK-27333 - sip_to_pjsip not correctly handling
disallow=all directive
(Reported by Torrey Searle)
Improvements made in this release:
-----------------------------------
* ASTERISK-24297 - cdr.c: Minor code optimizations.
(Reported by Richard Mudgett)
* ASTERISK-27449 - [PATCH] When failing to acquire target
during attended transfer, display wanted extension
(Reported by Niklas Larsson)
* ASTERISK-27456 - app_voicemail: Add new object for
VoicemailUserEntry
(Reported by sungtae kim)
* ASTERISK-27380 - ast_coredumper: allow pointing out the
asterisk binary explicitly
(Reported by Tzafrir Cohen)
* ASTERISK-23556 - Compilation warning for invert.c (array
subscript is above array bounds)
(Reported by Marcello Ceschia)
* ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
(Reported by Richard Mudgett)
* ASTERISK-27335 - CDR performance needs improvement.
(Reported by Richard Mudgett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.0
Thank you for your continued support of Asterisk!
----- 13.18.5 -----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18.
The available releases are released as versions 13.18.5, 14.7.5,
15.1.5 and 13.18-cert2.
The following security vulnerabilities were resolved in these versions:
* AST-2017-014: Crash in PJSIP resource when missing a contact header
A select set of SIP messages create a dialog in Asterisk. Those
SIP messages must contain a contact header. For those messages,
if the header was not present and using the PJSIP channel driver,
it would cause Asterisk to crash. The severity of this vulnerability
is somewhat mitigated if authentication is enabled. If authentication
is enabled a user would have to first be authorized before reaching
the crash point.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
The security advisory is available at:
https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
Thank you for your continued support of Asterisk!
----- 13.18.4 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert9, 13.18.4,
14.7.4 and 15.1.4.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-012: Remote Crash Vulnerability in RTCP Stack
If a compound RTCP packet is received containing more than
one report (for example a Receiver Report and a Sender
Report) the RTCP stack will incorrectly store report
information outside of allocated memory potentially causing
a crash.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-012.html
http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
Thank you for your continued support of Asterisk!
----- 13.18.3 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert8, 13.18.3,
14.7.3 and 15.1.3.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
If the chan_skinny (AKA SCCP protocol) channel driver is
flooded with certain requests it can cause the asterisk
process to use excessive amounts of virtual memory
eventually causing asterisk to stop processing requests of
any kind.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
Thank you for your continued support of Asterisk!
----- 13.18.2 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.18.2.
The release of Asterisk 13.18.2 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27391 - Regression: Deadlock between AOR named lock
and pjproject grp lock
(Reported by shaurya jain)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.2
Thank you for your continued support of Asterisk!
----- 13.18.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.18.0.
The release of Asterisk 13.18.0 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-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)
Bugs fixed in this release:
-----------------------------------
* 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 Benoît 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 Engström)
* 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 Seán 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-27093 - ODBC deadlocks when app_directory tries to
play back non-existent voicemail greeting
(Reported by James Terhune)
New Features made in this release:
-----------------------------------
* 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)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.0
Thank you for your continued support of Asterisk!
----- 13.17.0 ----
The Asterisk Development Team would like to announce the release
of Asterisk 13.17.0.
The release of Asterisk 13.17.0 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:
Bugs fixed in this release:
-----------------------------------
* 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-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-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-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-27060 - Comment typo format_g729.c
(Reported by Matthew Fredrickson)
* ASTERISK-27026 - res_ari: Crash when no ari.conf
configuration file exists
(Reported by Ronald Raikes)
* 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-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
get_write_timeout
(Reported by Jørgen 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 Engström)
* ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
when request and To URI differ
(Reported by Yasin CANER)
* ASTERISK-26789 - Audit manipulation of channel flags without
locks
(Reported by Joshua Colp)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
6869i)
(Reported by Matthias Binder)
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-26124 - res_agi: Set audio format for EAGI audio
stream
(Reported by John Fawcett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0
Thank you for your continued support of Asterisk!
2018-01-23 09:26:08 +01:00
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libdata/asterisk/sounds/en/.asterisk-core-sounds-en-gsm-1.6
|
2015-12-06 00:29:05 +01:00
|
|
|
libdata/asterisk/sounds/en/1-for-am-2-for-pm.gsm
|
|
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libdata/asterisk/sounds/en/1-yes-2-no.gsm
|
update to Asterisk 13.19.0 -- this contains both security fixes
and general bug fixes, fixes AST-2017-005, AST-2017-006, AST-2017-007,
AST-2017-008, AST-2017-009, AST-2017-10, AST-2017-11, AST-2017-12,
AST-2017-13, and AST-2017-14 (note that a number of these only
pertain to PJSIP which isn't used in pkgsrc)
----- 13.19.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.19.0.
The release of Asterisk 13.19.0 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:
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-27413 - Add cache_media_frames debugging option.
(Reported by Richard Mudgett)
* ASTERISK-27206 - res_pjsip: No mechanism exists to limit
endpoint identification to IP only
(Reported by Ben Merrills)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27531 - Compiler optimizations can break module load
sequence.
(Reported by abelbeck)
* ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
Contact crashes asterisk
(Reported by Ross Beer)
* ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
read()
(Reported by Abhay Gupta)
* ASTERISK-25079 - AMI bridge of channels results in MOH not
destroyed and robotic audio on one channel
(Reported by Zane Conkle)
* ASTERISK-27490 - chan_console: 'set active' fails to work
(Reported by Tzafrir Cohen)
* ASTERISK-24756 - ConfBridge sound_muted does not work from
CLI or AMI
(Reported by Thomas Frederiksen)
* ASTERISK-25649 - Transfer application does not work with
Local channels - documentation misleading
(Reported by Ivan Ullmann)
* ASTERISK-25869 - chan_sip: "rejected because extension not
found" should be logged as a security event
(Reported by Brian J. Murrell)
* ASTERISK-27440 - Strictrtp has issues to qualify video rtp
streams
(Reported by Wim De Vlaminck)
* ASTERISK-24329 - Music On Hold announcement cuts intro of
music the first time it is played
(Reported by Thomas Frederiksen)
* ASTERISK-19657 - Coverity Report: Fix issues for error type
CHAR_IO
(Reported by Matt Jordan)
* ASTERISK-27175 - iax.conf demo peer is invalid
(Reported by Tzafrir Cohen)
* ASTERISK-27430 - README refers to security documents that do
not exist.
(Reported by Corey Farrell)
* ASTERISK-20281 - "core set verbose" behaves strangely, can't
alias it, cli.conf example broken
(Reported by Tim Ringenbach at Asteria Solutions Group)
* 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-27408 - Identify causes and fix
pjsip/resolver/srv/failover/in_dialog/transport_tcp
(Reported by Corey Farrell)
* ASTERISK-18411 - Queue members with hints for state_interface
get stuck in "In Use" state.
(Reported by Steven T. Wheeler)
* ASTERISK-26131 - chan_sip: Crash Asterisk (in
sip_request_call at chan_sip.c) by making a call to a single
character in a dot pattern match
(Reported by Dwayne Hubbard)
* ASTERISK-27475 - codec_opus requires libcurl
(Reported by Samuel For)
* ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
not applied on reload
(Reported by John Bigelow)
* ASTERISK-27465 - CLI Completion Not Working
(Reported by Ross Beer)
* ASTERISK-27460 - CDR: Deadlock using AMI Originate with
Variable CDR(amaflags)=...
(Reported by Richard Mudgett)
* ASTERISK-27453 - RTP: Blind transfer direct media scenario
results in one way audio.
(Reported by Richard Mudgett)
* ASTERISK-20643 - SIP ICE support - remove hardcoded
limitation on SDP size, make ICE support disabled by default in
SIP, maybe provide a better warning message
(Reported by Roy)
* ASTERISK-26980 - pjsip: Clean up WebRTC disables
(Reported by abelbeck)
* 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-23735 - Transcoding makes bad choice in high-rate
translations
(Reported by Richard Kenner)
* ASTERISK-27445 - ARI: Updating a bridge gives wrong error
message.
(Reported by Frank Durden)
* ASTERISK-24662 - [patch] column and row headers for Signed
Linear format variants in output of 'core show translation' are
ambiguous
(Reported by Rusty Newton)
* ASTERISK-27353 - H323 audio starts with a delay of 2
seconds.
(Reported by Marco Giordani)
* ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
media
(Reported by Kevin Harwell)
* ASTERISK-27437 - [patch] ICE: server-reflexive candidates
(srflx) with Dual-Stack.
(Reported by Alexander Traud)
* ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
IPv6 addresses.
(Reported by Alexander Traud)
* ASTERISK-27435 - [patch] configure:
pjsip_evsub_set_uas_timeout not found.
(Reported by Alexander Traud)
* ASTERISK-27431 - Asterisk fails to build when openssl headers
are not installed.
(Reported by Corey Farrell)
* ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
(Reported by Ivan Larionov)
* ASTERISK-27421 - RTP source learning not working with devices
that have some clock issues
(Reported by nappsoft)
* ASTERISK-27361 - Attended transfer crashes in Asterisk
13.17.2
(Reported by Alessandro Pimenta)
* ASTERISK-27238 - Bridging: Crash freeing a frame that's
already been freed
(Reported by Richard Kenner)
* ASTERISK-27412 - core: Audiohook freeing interpolated frame
when it shouldn't.
(Reported by Mikhail)
* ASTERISK-27423 - app_record: We set the RECORD_STATUS
channel variable before closing the file
(Reported by George Joseph)
* ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
insert same ip address in "source ip address" and "destination
ip address" fields in HEP packets
(Reported by Max Norba)
* ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
is equal to RemoteAddress)
(Reported by Vasilii Rogin)
* ASTERISK-27415 - asterisk.conf: Setting astctl without
setting astrundir is ineffective.
(Reported by Corey Farrell)
* 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-27393 - res_pjsip: Crash occurs when an empty
contact read from astdb or database
(Reported by Aaron An)
* ASTERISK-27290 - res_pjsip: PIDF contact field has
malformed/invalid XML
(Reported by basildane)
* ASTERISK-27032 - res_pjsip: TLS options do not handle empty
values
(Reported by seanchann.zhou)
* ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
enabled but not configured.
(Reported by Alexander Traud)
* ASTERISK-26426 - format_ogg_opus: remove from source
(Reported by Kevin Harwell)
* ASTERISK-27378 - Modules: Fix issues with CLI completion.
(Reported by Corey Farrell)
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27390 - Audit menuselect module dependencies
(Reported by Corey Farrell)
* ASTERISK-27389 - Optional API modules should not allow
unload.
(Reported by Corey Farrell)
* ASTERISK-27369 - Bridge() dialplan application fails without
setting BRIDGERESULT channel variable
(Reported by James Terhune)
* ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
documentation
(Reported by Igor Goncharovsky)
* ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
'imap_delete_old_greeting'
(Reported by Anthony Messina)
* ASTERISK-27194 - jitterbuffer: Does not handle case where
translator returns null frame.
(Reported by Joshua Elson)
* ASTERISK-26639 - core: Disabling xmldoc support does not
work. Also results in abort during Asterisk startup.
(Reported by Mr Dini)
* ASTERISK-27372 - ARI: Node ARI client broken in latest
versions of 13 and 14
(Reported by Benjamin Keith Ford)
* ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
absence of the Expires header field with an unsubscribe action.
(Reported by Jonathan Cloots)
* ASTERISK-25960 - The config_hook unit test causes Asterisk to
crash if run a second time
(Reported by George Joseph)
* ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
when rtp_ipv6 set to yes
(Reported by Martin Cisárik)
* ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
curl is loaded
(Reported by Ronald Raikes)
* ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
but first on SDP media level.
(Reported by Alexander Traud)
* ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
Assertion on un/re-load: mod.id == -1
(Reported by Tzafrir Cohen)
* ASTERISK-23462 - Cannot disable SIP debugging via CLI after
enabling with conf file option - also 'sip set debug off'
reports debugging disabled, when it really isn't
(Reported by Rusty Newton)
* ASTERISK-27328 - Missing openssl dependencies in
res_rtp_asterisk and tcptls
(Reported by Tzafrir Cohen)
* ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
(o=) contains local address.
(Reported by Alexander Traud)
* ASTERISK-27343 - Fails to build in FreeBSD due to
sys/sysmacros.h not existing there
(Reported by Guido Falsi)
* ASTERISK-27340 - backtrace.c: Crash due to double-free.
(Reported by Corey Farrell)
* ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
stopping.
(Reported by Alexander Traud)
* ASTERISK-27333 - sip_to_pjsip not correctly handling
disallow=all directive
(Reported by Torrey Searle)
Improvements made in this release:
-----------------------------------
* ASTERISK-24297 - cdr.c: Minor code optimizations.
(Reported by Richard Mudgett)
* ASTERISK-27449 - [PATCH] When failing to acquire target
during attended transfer, display wanted extension
(Reported by Niklas Larsson)
* ASTERISK-27456 - app_voicemail: Add new object for
VoicemailUserEntry
(Reported by sungtae kim)
* ASTERISK-27380 - ast_coredumper: allow pointing out the
asterisk binary explicitly
(Reported by Tzafrir Cohen)
* ASTERISK-23556 - Compilation warning for invert.c (array
subscript is above array bounds)
(Reported by Marcello Ceschia)
* ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
(Reported by Richard Mudgett)
* ASTERISK-27335 - CDR performance needs improvement.
(Reported by Richard Mudgett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.0
Thank you for your continued support of Asterisk!
----- 13.18.5 -----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18.
The available releases are released as versions 13.18.5, 14.7.5,
15.1.5 and 13.18-cert2.
The following security vulnerabilities were resolved in these versions:
* AST-2017-014: Crash in PJSIP resource when missing a contact header
A select set of SIP messages create a dialog in Asterisk. Those
SIP messages must contain a contact header. For those messages,
if the header was not present and using the PJSIP channel driver,
it would cause Asterisk to crash. The severity of this vulnerability
is somewhat mitigated if authentication is enabled. If authentication
is enabled a user would have to first be authorized before reaching
the crash point.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
The security advisory is available at:
https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
Thank you for your continued support of Asterisk!
----- 13.18.4 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert9, 13.18.4,
14.7.4 and 15.1.4.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-012: Remote Crash Vulnerability in RTCP Stack
If a compound RTCP packet is received containing more than
one report (for example a Receiver Report and a Sender
Report) the RTCP stack will incorrectly store report
information outside of allocated memory potentially causing
a crash.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-012.html
http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
Thank you for your continued support of Asterisk!
----- 13.18.3 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert8, 13.18.3,
14.7.3 and 15.1.3.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
If the chan_skinny (AKA SCCP protocol) channel driver is
flooded with certain requests it can cause the asterisk
process to use excessive amounts of virtual memory
eventually causing asterisk to stop processing requests of
any kind.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
Thank you for your continued support of Asterisk!
----- 13.18.2 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.18.2.
The release of Asterisk 13.18.2 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27391 - Regression: Deadlock between AOR named lock
and pjproject grp lock
(Reported by shaurya jain)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.2
Thank you for your continued support of Asterisk!
----- 13.18.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.18.0.
The release of Asterisk 13.18.0 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-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)
Bugs fixed in this release:
-----------------------------------
* 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 Benoît 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 Engström)
* 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 Seán 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-27093 - ODBC deadlocks when app_directory tries to
play back non-existent voicemail greeting
(Reported by James Terhune)
New Features made in this release:
-----------------------------------
* 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)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.0
Thank you for your continued support of Asterisk!
----- 13.17.0 ----
The Asterisk Development Team would like to announce the release
of Asterisk 13.17.0.
The release of Asterisk 13.17.0 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:
Bugs fixed in this release:
-----------------------------------
* 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-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-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-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-27060 - Comment typo format_g729.c
(Reported by Matthew Fredrickson)
* ASTERISK-27026 - res_ari: Crash when no ari.conf
configuration file exists
(Reported by Ronald Raikes)
* 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-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
get_write_timeout
(Reported by Jørgen 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 Engström)
* ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
when request and To URI differ
(Reported by Yasin CANER)
* ASTERISK-26789 - Audit manipulation of channel flags without
locks
(Reported by Joshua Colp)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
6869i)
(Reported by Matthias Binder)
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-26124 - res_agi: Set audio format for EAGI audio
stream
(Reported by John Fawcett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0
Thank you for your continued support of Asterisk!
2018-01-23 09:26:08 +01:00
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libdata/asterisk/sounds/en/CHANGES-asterisk-core-en-1.6
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libdata/asterisk/sounds/en/CREDITS-asterisk-core-en-1.6
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libdata/asterisk/sounds/en/CREDITS-asterisk-extra-en-1.5.1
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libdata/asterisk/sounds/en/LICENSE-asterisk-core-en-1.6
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libdata/asterisk/sounds/en/LICENSE-asterisk-extra-en-1.5.1
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libdata/asterisk/sounds/en/OfficeSpace.gsm
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libdata/asterisk/sounds/en/Randulo-allison.gsm
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libdata/asterisk/sounds/en/SIP_Test_Failure.gsm
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libdata/asterisk/sounds/en/SIP_Test_Success.gsm
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libdata/asterisk/sounds/en/T-changed-to.gsm
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libdata/asterisk/sounds/en/T-is-not-available.gsm
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libdata/asterisk/sounds/en/T-to-disable-ancmnt.gsm
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libdata/asterisk/sounds/en/T-to-enable-ancmnt.gsm
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libdata/asterisk/sounds/en/T-to-hear-cur-ancmnt.gsm
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libdata/asterisk/sounds/en/T-to-leave-msg.gsm
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libdata/asterisk/sounds/en/T-to-reach-main-office.gsm
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libdata/asterisk/sounds/en/T-to-rec-ancmnt.gsm
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libdata/asterisk/sounds/en/T-to-rtrn-to-main-menu.gsm
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libdata/asterisk/sounds/en/a-charge-for-this-svc.gsm
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libdata/asterisk/sounds/en/a-collect-charge-of.gsm
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libdata/asterisk/sounds/en/a-collect-charge.gsm
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libdata/asterisk/sounds/en/a-connect-charge-of.gsm
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libdata/asterisk/sounds/en/a-connect-charge.gsm
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libdata/asterisk/sounds/en/abandon-all-hope.gsm
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libdata/asterisk/sounds/en/abandons.gsm
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libdata/asterisk/sounds/en/academic-support.gsm
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libdata/asterisk/sounds/en/access-code.gsm
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libdata/asterisk/sounds/en/access-denied.gsm
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libdata/asterisk/sounds/en/access-granted.gsm
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libdata/asterisk/sounds/en/accessible-through-system.gsm
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libdata/asterisk/sounds/en/account-balance-is.gsm
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libdata/asterisk/sounds/en/account_number.gsm
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libdata/asterisk/sounds/en/accounting.gsm
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libdata/asterisk/sounds/en/accounts-payable.gsm
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libdata/asterisk/sounds/en/accounts-receivable.gsm
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libdata/asterisk/sounds/en/activated.gsm
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libdata/asterisk/sounds/en/added-to.gsm
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libdata/asterisk/sounds/en/added.gsm
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libdata/asterisk/sounds/en/address.gsm
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libdata/asterisk/sounds/en/administration.gsm
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libdata/asterisk/sounds/en/advised-to-seek-shelter.gsm
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libdata/asterisk/sounds/en/after-the-tone.gsm
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libdata/asterisk/sounds/en/after_tone.gsm
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libdata/asterisk/sounds/en/afternoon.gsm
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libdata/asterisk/sounds/en/agent-alreadyon.gsm
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libdata/asterisk/sounds/en/agent-incorrect.gsm
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libdata/asterisk/sounds/en/agent-loggedoff.gsm
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libdata/asterisk/sounds/en/agent-loginok.gsm
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libdata/asterisk/sounds/en/agent-newlocation.gsm
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libdata/asterisk/sounds/en/agent-pass.gsm
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libdata/asterisk/sounds/en/agent-user.gsm
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libdata/asterisk/sounds/en/alabama.gsm
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libdata/asterisk/sounds/en/albuquerque.gsm
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libdata/asterisk/sounds/en/alert.gsm
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libdata/asterisk/sounds/en/all-circuits-busy-now.gsm
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libdata/asterisk/sounds/en/all-outgoing-lines-unavailable.gsm
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libdata/asterisk/sounds/en/all-reps-busy.gsm
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libdata/asterisk/sounds/en/all-your-base.gsm
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libdata/asterisk/sounds/en/altitude.gsm
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libdata/asterisk/sounds/en/ampersand.gsm
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libdata/asterisk/sounds/en/an-error-has-occurred.gsm
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libdata/asterisk/sounds/en/and-area-code.gsm
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libdata/asterisk/sounds/en/and-or.gsm
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libdata/asterisk/sounds/en/and-prs-pound-whn-finished.gsm
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libdata/asterisk/sounds/en/andnowstandby.gsm
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libdata/asterisk/sounds/en/are-you-still-there.gsm
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libdata/asterisk/sounds/en/arkansas.gsm
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libdata/asterisk/sounds/en/arlington.gsm
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libdata/asterisk/sounds/en/ascending-2tone.gsm
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libdata/asterisk/sounds/en/astcc-account-balance-is.gsm
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libdata/asterisk/sounds/en/astcc-account-number-invalid.gsm
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libdata/asterisk/sounds/en/astcc-balance-of-account-is.gsm
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libdata/asterisk/sounds/en/astcc-card-number-invalid.gsm
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libdata/asterisk/sounds/en/astcc-digit-account-number.gsm
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libdata/asterisk/sounds/en/astcc-followed-by-the-hash-key.gsm
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libdata/asterisk/sounds/en/astcc-followed-by-the-pound-key.gsm
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libdata/asterisk/sounds/en/astcc-login12pound.gsm
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libdata/asterisk/sounds/en/astcc-please-enter-your.gsm
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libdata/asterisk/sounds/en/astcc-skipping-any-punctuation.gsm
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libdata/asterisk/sounds/en/asterisk-friend.gsm
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libdata/asterisk/sounds/en/at-any-time.gsm
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libdata/asterisk/sounds/en/at-customers-request.gsm
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libdata/asterisk/sounds/en/at-following-number.gsm
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libdata/asterisk/sounds/en/at-sign.gsm
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libdata/asterisk/sounds/en/at-tone-time-exactly.gsm
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libdata/asterisk/sounds/en/at_tone.gsm
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libdata/asterisk/sounds/en/athletics.gsm
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libdata/asterisk/sounds/en/atlanta.gsm
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libdata/asterisk/sounds/en/atlantic.gsm
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libdata/asterisk/sounds/en/attention-required.gsm
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libdata/asterisk/sounds/en/auditing.gsm
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libdata/asterisk/sounds/en/austin.gsm
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libdata/asterisk/sounds/en/auth-incorrect.gsm
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libdata/asterisk/sounds/en/auth-thankyou.gsm
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libdata/asterisk/sounds/en/available-options.gsm
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libdata/asterisk/sounds/en/available.gsm
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libdata/asterisk/sounds/en/avg-speed-answer.gsm
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libdata/asterisk/sounds/en/away-naughty-boy.gsm
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libdata/asterisk/sounds/en/away-naughty-girl.gsm
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libdata/asterisk/sounds/en/awkward.gsm
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libdata/asterisk/sounds/en/backslash.gsm
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libdata/asterisk/sounds/en/bad.gsm
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libdata/asterisk/sounds/en/baltimore.gsm
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libdata/asterisk/sounds/en/bar.gsm
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libdata/asterisk/sounds/en/barn.gsm
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libdata/asterisk/sounds/en/barns.gsm
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libdata/asterisk/sounds/en/barometric.gsm
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libdata/asterisk/sounds/en/basic-pbx-ivr-main.gsm
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libdata/asterisk/sounds/en/bearing.gsm
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libdata/asterisk/sounds/en/beaufort.gsm
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libdata/asterisk/sounds/en/because-paranoid.gsm
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libdata/asterisk/sounds/en/beep.gsm
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libdata/asterisk/sounds/en/beeperr.gsm
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libdata/asterisk/sounds/en/before-the-number.gsm
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libdata/asterisk/sounds/en/believe-its-free.gsm
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libdata/asterisk/sounds/en/billing-and-collections.gsm
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libdata/asterisk/sounds/en/billing.gsm
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libdata/asterisk/sounds/en/billionth.gsm
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libdata/asterisk/sounds/en/binary.gsm
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libdata/asterisk/sounds/en/bits.gsm
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libdata/asterisk/sounds/en/blue-eyed-polar-bear.gsm
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libdata/asterisk/sounds/en/bombsquad.gsm
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libdata/asterisk/sounds/en/bookstore.gsm
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libdata/asterisk/sounds/en/boston.gsm
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libdata/asterisk/sounds/en/box.gsm
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libdata/asterisk/sounds/en/brian.gsm
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libdata/asterisk/sounds/en/business-development.gsm
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libdata/asterisk/sounds/en/busy-hangovers.gsm
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libdata/asterisk/sounds/en/busy-pls-hold.gsm
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libdata/asterisk/sounds/en/but.gsm
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libdata/asterisk/sounds/en/by.gsm
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libdata/asterisk/sounds/en/bytes.gsm
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libdata/asterisk/sounds/en/cafeteria.gsm
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libdata/asterisk/sounds/en/california.gsm
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libdata/asterisk/sounds/en/call-forward.gsm
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libdata/asterisk/sounds/en/call-forwarding.gsm
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libdata/asterisk/sounds/en/call-fwd-cancelled.gsm
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libdata/asterisk/sounds/en/call-fwd-no-ans.gsm
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libdata/asterisk/sounds/en/call-fwd-on-busy.gsm
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libdata/asterisk/sounds/en/call-fwd-parallel.gsm
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libdata/asterisk/sounds/en/call-fwd-unconditional.gsm
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libdata/asterisk/sounds/en/call-preempted.gsm
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libdata/asterisk/sounds/en/call-quality-menu.gsm
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libdata/asterisk/sounds/en/call-requres.gsm
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libdata/asterisk/sounds/en/call-terminated.gsm
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libdata/asterisk/sounds/en/call-waiting.gsm
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libdata/asterisk/sounds/en/call.gsm
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libdata/asterisk/sounds/en/calling.gsm
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libdata/asterisk/sounds/en/calls-taken-by.gsm
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libdata/asterisk/sounds/en/calls-waiting-for-rep.gsm
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libdata/asterisk/sounds/en/calls.gsm
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libdata/asterisk/sounds/en/campground-office.gsm
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libdata/asterisk/sounds/en/cancelled.gsm
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libdata/asterisk/sounds/en/cannot-complete-as-dialed.gsm
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libdata/asterisk/sounds/en/cannot-complete-network-error.gsm
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libdata/asterisk/sounds/en/cannot-complete-otherend-error.gsm
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libdata/asterisk/sounds/en/cannot-complete-temp-error.gsm
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libdata/asterisk/sounds/en/card-balance-is.gsm
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libdata/asterisk/sounds/en/card-is-invalid.gsm
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libdata/asterisk/sounds/en/card-number.gsm
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libdata/asterisk/sounds/en/carried-away-by-monkeys.gsm
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libdata/asterisk/sounds/en/cause-code.gsm
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libdata/asterisk/sounds/en/cc-amex.gsm
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libdata/asterisk/sounds/en/cc-discover.gsm
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libdata/asterisk/sounds/en/cc-mastercard.gsm
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libdata/asterisk/sounds/en/cc-visa.gsm
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libdata/asterisk/sounds/en/celsius.gsm
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libdata/asterisk/sounds/en/cent.gsm
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libdata/asterisk/sounds/en/central.gsm
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libdata/asterisk/sounds/en/cents-per-minute.gsm
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libdata/asterisk/sounds/en/cents.gsm
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libdata/asterisk/sounds/en/ceo-office.gsm
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libdata/asterisk/sounds/en/challenge_try_again.gsm
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libdata/asterisk/sounds/en/chance-of.gsm
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libdata/asterisk/sounds/en/changing.gsm
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libdata/asterisk/sounds/en/channel-insecure-warn.gsm
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libdata/asterisk/sounds/en/channel-secure.gsm
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libdata/asterisk/sounds/en/channel.gsm
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libdata/asterisk/sounds/en/charlotte.gsm
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libdata/asterisk/sounds/en/chat-room.gsm
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libdata/asterisk/sounds/en/check-number-dial-again.gsm
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libdata/asterisk/sounds/en/chemistry.gsm
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libdata/asterisk/sounds/en/chicago.gsm
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libdata/asterisk/sounds/en/chris.gsm
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libdata/asterisk/sounds/en/claims.gsm
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libdata/asterisk/sounds/en/clear.gsm
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libdata/asterisk/sounds/en/clearing.gsm
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libdata/asterisk/sounds/en/cleveland.gsm
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libdata/asterisk/sounds/en/clli.gsm
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libdata/asterisk/sounds/en/close-parenthesis.gsm
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libdata/asterisk/sounds/en/closed.gsm
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libdata/asterisk/sounds/en/clouds.gsm
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libdata/asterisk/sounds/en/cloudy.gsm
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libdata/asterisk/sounds/en/collections.gsm
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libdata/asterisk/sounds/en/colorado-springs.gsm
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libdata/asterisk/sounds/en/colorado.gsm
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libdata/asterisk/sounds/en/columbus.gsm
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libdata/asterisk/sounds/en/comedyclub.gsm
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libdata/asterisk/sounds/en/comma.gsm
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libdata/asterisk/sounds/en/communications.gsm
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libdata/asterisk/sounds/en/company-dir-411.gsm
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libdata/asterisk/sounds/en/complaint.gsm
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libdata/asterisk/sounds/en/compliance.gsm
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libdata/asterisk/sounds/en/computer-friend1.gsm
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libdata/asterisk/sounds/en/computer-friend2.gsm
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libdata/asterisk/sounds/en/conditions.gsm
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libdata/asterisk/sounds/en/conf-1-to-list-users.gsm
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libdata/asterisk/sounds/en/conf-2-to-kick-nonadmin.gsm
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libdata/asterisk/sounds/en/conf-3-mute-or-unmute-nonadmin.gsm
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libdata/asterisk/sounds/en/conf-4-to-record-conf.gsm
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libdata/asterisk/sounds/en/conf-8-for-more-options.gsm
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libdata/asterisk/sounds/en/conf-8-to-exit-return-to-conf.gsm
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libdata/asterisk/sounds/en/conf-adminmenu-162.gsm
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libdata/asterisk/sounds/en/conf-adminmenu-18.gsm
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libdata/asterisk/sounds/en/conf-adminmenu-menu8.gsm
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libdata/asterisk/sounds/en/conf-adminmenu.gsm
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libdata/asterisk/sounds/en/conf-banned.gsm
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libdata/asterisk/sounds/en/conf-enteringno.gsm
|
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libdata/asterisk/sounds/en/conf-errormenu.gsm
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libdata/asterisk/sounds/en/conf-extended.gsm
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libdata/asterisk/sounds/en/conf-full.gsm
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libdata/asterisk/sounds/en/conf-getchannel.gsm
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libdata/asterisk/sounds/en/conf-getconfno.gsm
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libdata/asterisk/sounds/en/conf-getpin.gsm
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libdata/asterisk/sounds/en/conf-hasentered.gsm
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libdata/asterisk/sounds/en/conf-hasjoin.gsm
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libdata/asterisk/sounds/en/conf-hasleft.gsm
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libdata/asterisk/sounds/en/conf-invalid.gsm
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libdata/asterisk/sounds/en/conf-invalidpin.gsm
|
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libdata/asterisk/sounds/en/conf-kicked.gsm
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libdata/asterisk/sounds/en/conf-leaderhasleft.gsm
|
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libdata/asterisk/sounds/en/conf-locked.gsm
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libdata/asterisk/sounds/en/conf-lockednow.gsm
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libdata/asterisk/sounds/en/conf-muted.gsm
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libdata/asterisk/sounds/en/conf-noempty.gsm
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libdata/asterisk/sounds/en/conf-nonextended.gsm
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libdata/asterisk/sounds/en/conf-now-muted.gsm
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libdata/asterisk/sounds/en/conf-now-recording.gsm
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libdata/asterisk/sounds/en/conf-now-unmuted.gsm
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libdata/asterisk/sounds/en/conf-onlyone.gsm
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libdata/asterisk/sounds/en/conf-onlyperson.gsm
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libdata/asterisk/sounds/en/conf-onlypersonleft.gsm
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libdata/asterisk/sounds/en/conf-otherinparty.gsm
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libdata/asterisk/sounds/en/conf-peopleinconf.gsm
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libdata/asterisk/sounds/en/conf-placeintoconf.gsm
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libdata/asterisk/sounds/en/conf-roll-callcomplete.gsm
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libdata/asterisk/sounds/en/conf-sysop.gsm
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libdata/asterisk/sounds/en/conf-sysopreq.gsm
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libdata/asterisk/sounds/en/conf-sysopreqcancelled.gsm
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libdata/asterisk/sounds/en/conf-thereare.gsm
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libdata/asterisk/sounds/en/conf-unlockednow.gsm
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libdata/asterisk/sounds/en/conf-unmuted.gsm
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libdata/asterisk/sounds/en/conf-usermenu-162.gsm
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libdata/asterisk/sounds/en/conf-usermenu.gsm
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libdata/asterisk/sounds/en/conf-userswilljoin.gsm
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libdata/asterisk/sounds/en/conf-userwilljoin.gsm
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libdata/asterisk/sounds/en/conf-waitforleader.gsm
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libdata/asterisk/sounds/en/conf-youareinconfnum.gsm
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libdata/asterisk/sounds/en/confbridge-begin-glorious-a.gsm
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libdata/asterisk/sounds/en/confbridge-begin-glorious-b.gsm
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libdata/asterisk/sounds/en/confbridge-begin-glorious-c.gsm
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libdata/asterisk/sounds/en/confbridge-begin-leader.gsm
|
update to Asterisk 13.19.0 -- this contains both security fixes
and general bug fixes, fixes AST-2017-005, AST-2017-006, AST-2017-007,
AST-2017-008, AST-2017-009, AST-2017-10, AST-2017-11, AST-2017-12,
AST-2017-13, and AST-2017-14 (note that a number of these only
pertain to PJSIP which isn't used in pkgsrc)
----- 13.19.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.19.0.
The release of Asterisk 13.19.0 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:
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-27413 - Add cache_media_frames debugging option.
(Reported by Richard Mudgett)
* ASTERISK-27206 - res_pjsip: No mechanism exists to limit
endpoint identification to IP only
(Reported by Ben Merrills)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27531 - Compiler optimizations can break module load
sequence.
(Reported by abelbeck)
* ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
Contact crashes asterisk
(Reported by Ross Beer)
* ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
read()
(Reported by Abhay Gupta)
* ASTERISK-25079 - AMI bridge of channels results in MOH not
destroyed and robotic audio on one channel
(Reported by Zane Conkle)
* ASTERISK-27490 - chan_console: 'set active' fails to work
(Reported by Tzafrir Cohen)
* ASTERISK-24756 - ConfBridge sound_muted does not work from
CLI or AMI
(Reported by Thomas Frederiksen)
* ASTERISK-25649 - Transfer application does not work with
Local channels - documentation misleading
(Reported by Ivan Ullmann)
* ASTERISK-25869 - chan_sip: "rejected because extension not
found" should be logged as a security event
(Reported by Brian J. Murrell)
* ASTERISK-27440 - Strictrtp has issues to qualify video rtp
streams
(Reported by Wim De Vlaminck)
* ASTERISK-24329 - Music On Hold announcement cuts intro of
music the first time it is played
(Reported by Thomas Frederiksen)
* ASTERISK-19657 - Coverity Report: Fix issues for error type
CHAR_IO
(Reported by Matt Jordan)
* ASTERISK-27175 - iax.conf demo peer is invalid
(Reported by Tzafrir Cohen)
* ASTERISK-27430 - README refers to security documents that do
not exist.
(Reported by Corey Farrell)
* ASTERISK-20281 - "core set verbose" behaves strangely, can't
alias it, cli.conf example broken
(Reported by Tim Ringenbach at Asteria Solutions Group)
* 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-27408 - Identify causes and fix
pjsip/resolver/srv/failover/in_dialog/transport_tcp
(Reported by Corey Farrell)
* ASTERISK-18411 - Queue members with hints for state_interface
get stuck in "In Use" state.
(Reported by Steven T. Wheeler)
* ASTERISK-26131 - chan_sip: Crash Asterisk (in
sip_request_call at chan_sip.c) by making a call to a single
character in a dot pattern match
(Reported by Dwayne Hubbard)
* ASTERISK-27475 - codec_opus requires libcurl
(Reported by Samuel For)
* ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
not applied on reload
(Reported by John Bigelow)
* ASTERISK-27465 - CLI Completion Not Working
(Reported by Ross Beer)
* ASTERISK-27460 - CDR: Deadlock using AMI Originate with
Variable CDR(amaflags)=...
(Reported by Richard Mudgett)
* ASTERISK-27453 - RTP: Blind transfer direct media scenario
results in one way audio.
(Reported by Richard Mudgett)
* ASTERISK-20643 - SIP ICE support - remove hardcoded
limitation on SDP size, make ICE support disabled by default in
SIP, maybe provide a better warning message
(Reported by Roy)
* ASTERISK-26980 - pjsip: Clean up WebRTC disables
(Reported by abelbeck)
* 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-23735 - Transcoding makes bad choice in high-rate
translations
(Reported by Richard Kenner)
* ASTERISK-27445 - ARI: Updating a bridge gives wrong error
message.
(Reported by Frank Durden)
* ASTERISK-24662 - [patch] column and row headers for Signed
Linear format variants in output of 'core show translation' are
ambiguous
(Reported by Rusty Newton)
* ASTERISK-27353 - H323 audio starts with a delay of 2
seconds.
(Reported by Marco Giordani)
* ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
media
(Reported by Kevin Harwell)
* ASTERISK-27437 - [patch] ICE: server-reflexive candidates
(srflx) with Dual-Stack.
(Reported by Alexander Traud)
* ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
IPv6 addresses.
(Reported by Alexander Traud)
* ASTERISK-27435 - [patch] configure:
pjsip_evsub_set_uas_timeout not found.
(Reported by Alexander Traud)
* ASTERISK-27431 - Asterisk fails to build when openssl headers
are not installed.
(Reported by Corey Farrell)
* ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
(Reported by Ivan Larionov)
* ASTERISK-27421 - RTP source learning not working with devices
that have some clock issues
(Reported by nappsoft)
* ASTERISK-27361 - Attended transfer crashes in Asterisk
13.17.2
(Reported by Alessandro Pimenta)
* ASTERISK-27238 - Bridging: Crash freeing a frame that's
already been freed
(Reported by Richard Kenner)
* ASTERISK-27412 - core: Audiohook freeing interpolated frame
when it shouldn't.
(Reported by Mikhail)
* ASTERISK-27423 - app_record: We set the RECORD_STATUS
channel variable before closing the file
(Reported by George Joseph)
* ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
insert same ip address in "source ip address" and "destination
ip address" fields in HEP packets
(Reported by Max Norba)
* ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
is equal to RemoteAddress)
(Reported by Vasilii Rogin)
* ASTERISK-27415 - asterisk.conf: Setting astctl without
setting astrundir is ineffective.
(Reported by Corey Farrell)
* 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-27393 - res_pjsip: Crash occurs when an empty
contact read from astdb or database
(Reported by Aaron An)
* ASTERISK-27290 - res_pjsip: PIDF contact field has
malformed/invalid XML
(Reported by basildane)
* ASTERISK-27032 - res_pjsip: TLS options do not handle empty
values
(Reported by seanchann.zhou)
* ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
enabled but not configured.
(Reported by Alexander Traud)
* ASTERISK-26426 - format_ogg_opus: remove from source
(Reported by Kevin Harwell)
* ASTERISK-27378 - Modules: Fix issues with CLI completion.
(Reported by Corey Farrell)
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27390 - Audit menuselect module dependencies
(Reported by Corey Farrell)
* ASTERISK-27389 - Optional API modules should not allow
unload.
(Reported by Corey Farrell)
* ASTERISK-27369 - Bridge() dialplan application fails without
setting BRIDGERESULT channel variable
(Reported by James Terhune)
* ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
documentation
(Reported by Igor Goncharovsky)
* ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
'imap_delete_old_greeting'
(Reported by Anthony Messina)
* ASTERISK-27194 - jitterbuffer: Does not handle case where
translator returns null frame.
(Reported by Joshua Elson)
* ASTERISK-26639 - core: Disabling xmldoc support does not
work. Also results in abort during Asterisk startup.
(Reported by Mr Dini)
* ASTERISK-27372 - ARI: Node ARI client broken in latest
versions of 13 and 14
(Reported by Benjamin Keith Ford)
* ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
absence of the Expires header field with an unsubscribe action.
(Reported by Jonathan Cloots)
* ASTERISK-25960 - The config_hook unit test causes Asterisk to
crash if run a second time
(Reported by George Joseph)
* ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
when rtp_ipv6 set to yes
(Reported by Martin Cisárik)
* ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
curl is loaded
(Reported by Ronald Raikes)
* ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
but first on SDP media level.
(Reported by Alexander Traud)
* ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
Assertion on un/re-load: mod.id == -1
(Reported by Tzafrir Cohen)
* ASTERISK-23462 - Cannot disable SIP debugging via CLI after
enabling with conf file option - also 'sip set debug off'
reports debugging disabled, when it really isn't
(Reported by Rusty Newton)
* ASTERISK-27328 - Missing openssl dependencies in
res_rtp_asterisk and tcptls
(Reported by Tzafrir Cohen)
* ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
(o=) contains local address.
(Reported by Alexander Traud)
* ASTERISK-27343 - Fails to build in FreeBSD due to
sys/sysmacros.h not existing there
(Reported by Guido Falsi)
* ASTERISK-27340 - backtrace.c: Crash due to double-free.
(Reported by Corey Farrell)
* ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
stopping.
(Reported by Alexander Traud)
* ASTERISK-27333 - sip_to_pjsip not correctly handling
disallow=all directive
(Reported by Torrey Searle)
Improvements made in this release:
-----------------------------------
* ASTERISK-24297 - cdr.c: Minor code optimizations.
(Reported by Richard Mudgett)
* ASTERISK-27449 - [PATCH] When failing to acquire target
during attended transfer, display wanted extension
(Reported by Niklas Larsson)
* ASTERISK-27456 - app_voicemail: Add new object for
VoicemailUserEntry
(Reported by sungtae kim)
* ASTERISK-27380 - ast_coredumper: allow pointing out the
asterisk binary explicitly
(Reported by Tzafrir Cohen)
* ASTERISK-23556 - Compilation warning for invert.c (array
subscript is above array bounds)
(Reported by Marcello Ceschia)
* ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
(Reported by Richard Mudgett)
* ASTERISK-27335 - CDR performance needs improvement.
(Reported by Richard Mudgett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.0
Thank you for your continued support of Asterisk!
----- 13.18.5 -----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18.
The available releases are released as versions 13.18.5, 14.7.5,
15.1.5 and 13.18-cert2.
The following security vulnerabilities were resolved in these versions:
* AST-2017-014: Crash in PJSIP resource when missing a contact header
A select set of SIP messages create a dialog in Asterisk. Those
SIP messages must contain a contact header. For those messages,
if the header was not present and using the PJSIP channel driver,
it would cause Asterisk to crash. The severity of this vulnerability
is somewhat mitigated if authentication is enabled. If authentication
is enabled a user would have to first be authorized before reaching
the crash point.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
The security advisory is available at:
https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
Thank you for your continued support of Asterisk!
----- 13.18.4 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert9, 13.18.4,
14.7.4 and 15.1.4.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-012: Remote Crash Vulnerability in RTCP Stack
If a compound RTCP packet is received containing more than
one report (for example a Receiver Report and a Sender
Report) the RTCP stack will incorrectly store report
information outside of allocated memory potentially causing
a crash.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-012.html
http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
Thank you for your continued support of Asterisk!
----- 13.18.3 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert8, 13.18.3,
14.7.3 and 15.1.3.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
If the chan_skinny (AKA SCCP protocol) channel driver is
flooded with certain requests it can cause the asterisk
process to use excessive amounts of virtual memory
eventually causing asterisk to stop processing requests of
any kind.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
Thank you for your continued support of Asterisk!
----- 13.18.2 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.18.2.
The release of Asterisk 13.18.2 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27391 - Regression: Deadlock between AOR named lock
and pjproject grp lock
(Reported by shaurya jain)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.2
Thank you for your continued support of Asterisk!
----- 13.18.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.18.0.
The release of Asterisk 13.18.0 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-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)
Bugs fixed in this release:
-----------------------------------
* 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 Benoît 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 Engström)
* 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 Seán 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-27093 - ODBC deadlocks when app_directory tries to
play back non-existent voicemail greeting
(Reported by James Terhune)
New Features made in this release:
-----------------------------------
* 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)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.0
Thank you for your continued support of Asterisk!
----- 13.17.0 ----
The Asterisk Development Team would like to announce the release
of Asterisk 13.17.0.
The release of Asterisk 13.17.0 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:
Bugs fixed in this release:
-----------------------------------
* 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-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-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-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-27060 - Comment typo format_g729.c
(Reported by Matthew Fredrickson)
* ASTERISK-27026 - res_ari: Crash when no ari.conf
configuration file exists
(Reported by Ronald Raikes)
* 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-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
get_write_timeout
(Reported by Jørgen 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 Engström)
* ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
when request and To URI differ
(Reported by Yasin CANER)
* ASTERISK-26789 - Audit manipulation of channel flags without
locks
(Reported by Joshua Colp)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
6869i)
(Reported by Matthias Binder)
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-26124 - res_agi: Set audio format for EAGI audio
stream
(Reported by John Fawcett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0
Thank you for your continued support of Asterisk!
2018-01-23 09:26:08 +01:00
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libdata/asterisk/sounds/en/confbridge-binaural-off.gsm
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libdata/asterisk/sounds/en/confbridge-binaural-on.gsm
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2015-12-06 00:29:05 +01:00
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libdata/asterisk/sounds/en/confbridge-conf-begin.gsm
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libdata/asterisk/sounds/en/confbridge-conf-end.gsm
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libdata/asterisk/sounds/en/confbridge-has-joined.gsm
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libdata/asterisk/sounds/en/confbridge-inc-list-vol-in.gsm
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libdata/asterisk/sounds/en/confbridge-inc-talk-vol-in.gsm
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libdata/asterisk/sounds/en/confbridge-lock-extended.gsm
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libdata/asterisk/sounds/en/confbridge-menu-exit-in.gsm
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libdata/asterisk/sounds/en/confbridge-rest-list-vol-in.gsm
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libdata/asterisk/sounds/en/confbridge-rest-list-vol-out.gsm
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libdata/asterisk/sounds/en/confbridge-rest-talk-vol-in.gsm
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libdata/asterisk/sounds/en/confbridge-there-are.gsm
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libdata/asterisk/sounds/en/confbridge-unlocked.gsm
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libdata/asterisk/sounds/en/conference-call.gsm
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libdata/asterisk/sounds/en/confirm-number-is.gsm
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libdata/asterisk/sounds/en/connected.gsm
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libdata/asterisk/sounds/en/connecticut.gsm
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libdata/asterisk/sounds/en/connecting.gsm
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libdata/asterisk/sounds/en/connection-failed.gsm
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libdata/asterisk/sounds/en/connection-timed-out.gsm
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libdata/asterisk/sounds/en/continue-english-press.gsm
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libdata/asterisk/sounds/en/core-sounds-en.txt
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libdata/asterisk/sounds/en/could-lose-a-few-pounds.gsm
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libdata/asterisk/sounds/en/counseling-services.gsm
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libdata/asterisk/sounds/en/crashing_conf.gsm
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libdata/asterisk/sounds/en/current-time-is.gsm
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libdata/asterisk/sounds/en/current_account_balance.gsm
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libdata/asterisk/sounds/en/currently.gsm
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libdata/asterisk/sounds/en/customer-accounts.gsm
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libdata/asterisk/sounds/en/customer-relations.gsm
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libdata/asterisk/sounds/en/cyclone.gsm
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libdata/asterisk/sounds/en/demo-abouttotry.gsm
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libdata/asterisk/sounds/en/demo-enterkeywords.gsm
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libdata/asterisk/sounds/en/demo-instruct.gsm
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libdata/asterisk/sounds/en/denial-of-service.gsm
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libdata/asterisk/sounds/en/denver.gsm
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libdata/asterisk/sounds/en/digits/dollars.gsm
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libdata/asterisk/sounds/en/digits/h-50.gsm
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libdata/asterisk/sounds/en/digits/h-6.gsm
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libdata/asterisk/sounds/en/digits/h-60.gsm
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libdata/asterisk/sounds/en/digits/h-70.gsm
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libdata/asterisk/sounds/en/digits/h-8.gsm
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libdata/asterisk/sounds/en/digits/h-80.gsm
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libdata/asterisk/sounds/en/digits/h-9.gsm
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libdata/asterisk/sounds/en/digits/h-90.gsm
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libdata/asterisk/sounds/en/digits/h-billion.gsm
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libdata/asterisk/sounds/en/digits/h-hundred.gsm
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libdata/asterisk/sounds/en/digits/h-million.gsm
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libdata/asterisk/sounds/en/digits/h-thousand.gsm
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libdata/asterisk/sounds/en/digits/hundred.gsm
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libdata/asterisk/sounds/en/digits/million.gsm
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libdata/asterisk/sounds/en/digits/minus.gsm
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libdata/asterisk/sounds/en/digits/mon-0.gsm
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libdata/asterisk/sounds/en/digits/mon-1.gsm
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libdata/asterisk/sounds/en/digits/mon-10.gsm
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libdata/asterisk/sounds/en/digits/mon-11.gsm
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libdata/asterisk/sounds/en/digits/mon-2.gsm
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libdata/asterisk/sounds/en/digits/mon-3.gsm
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libdata/asterisk/sounds/en/digits/mon-4.gsm
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libdata/asterisk/sounds/en/digits/mon-5.gsm
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libdata/asterisk/sounds/en/digits/mon-6.gsm
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libdata/asterisk/sounds/en/digits/mon-7.gsm
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libdata/asterisk/sounds/en/digits/mon-8.gsm
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libdata/asterisk/sounds/en/digits/mon-9.gsm
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libdata/asterisk/sounds/en/digits/oclock.gsm
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libdata/asterisk/sounds/en/digits/oh.gsm
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libdata/asterisk/sounds/en/digits/p-m.gsm
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libdata/asterisk/sounds/en/digits/pound.gsm
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libdata/asterisk/sounds/en/digits/star.gsm
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libdata/asterisk/sounds/en/digits/thousand.gsm
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libdata/asterisk/sounds/en/digits/today.gsm
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libdata/asterisk/sounds/en/digits/tomorrow.gsm
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libdata/asterisk/sounds/en/digits/yesterday.gsm
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libdata/asterisk/sounds/en/dir-first.gsm
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libdata/asterisk/sounds/en/dir-firstlast.gsm
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libdata/asterisk/sounds/en/dir-instr.gsm
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libdata/asterisk/sounds/en/dir-intro-fn.gsm
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libdata/asterisk/sounds/en/dir-intro.gsm
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libdata/asterisk/sounds/en/dir-last.gsm
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libdata/asterisk/sounds/en/dir-multi1.gsm
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libdata/asterisk/sounds/en/dir-multi2.gsm
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libdata/asterisk/sounds/en/dir-multi3.gsm
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libdata/asterisk/sounds/en/dir-multi9.gsm
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libdata/asterisk/sounds/en/dir-nomatch.gsm
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libdata/asterisk/sounds/en/dir-nomore.gsm
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libdata/asterisk/sounds/en/dir-pls-enter.gsm
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libdata/asterisk/sounds/en/dir-usingkeypad.gsm
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libdata/asterisk/sounds/en/dir-welcome.gsm
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libdata/asterisk/sounds/en/directory-assistance.gsm
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libdata/asterisk/sounds/en/directory.gsm
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libdata/asterisk/sounds/en/disabled.gsm
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libdata/asterisk/sounds/en/discon-or-out-of-service.gsm
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libdata/asterisk/sounds/en/disconnected.gsm
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libdata/asterisk/sounds/en/disk.gsm
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libdata/asterisk/sounds/en/distribution.gsm
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libdata/asterisk/sounds/en/divided-by.gsm
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libdata/asterisk/sounds/en/dns.gsm
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libdata/asterisk/sounds/en/do-not-disturb.gsm
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libdata/asterisk/sounds/en/doing-enum-lookup.gsm
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libdata/asterisk/sounds/en/donotcall1.gsm
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libdata/asterisk/sounds/en/donotcall2.gsm
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libdata/asterisk/sounds/en/dont-know-who-sent.gsm
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libdata/asterisk/sounds/en/doppler-radar.gsm
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libdata/asterisk/sounds/en/down.gsm
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libdata/asterisk/sounds/en/driving-directions.gsm
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libdata/asterisk/sounds/en/duplex.gsm
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libdata/asterisk/sounds/en/duplication.gsm
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libdata/asterisk/sounds/en/early.gsm
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libdata/asterisk/sounds/en/east.gsm
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libdata/asterisk/sounds/en/easterly.gsm
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libdata/asterisk/sounds/en/eastern.gsm
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libdata/asterisk/sounds/en/echo-test.gsm
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libdata/asterisk/sounds/en/ed.gsm
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libdata/asterisk/sounds/en/eighteenth.gsm
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libdata/asterisk/sounds/en/eighth.gsm
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libdata/asterisk/sounds/en/eightieth.gsm
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libdata/asterisk/sounds/en/el-paso.gsm
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libdata/asterisk/sounds/en/eletelephony.gsm
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libdata/asterisk/sounds/en/eleventh.gsm
|
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libdata/asterisk/sounds/en/email.gsm
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libdata/asterisk/sounds/en/emergency.gsm
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libdata/asterisk/sounds/en/enabled.gsm
|
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libdata/asterisk/sounds/en/encode.gsm
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libdata/asterisk/sounds/en/engineering.gsm
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libdata/asterisk/sounds/en/ent-target-attendant.gsm
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libdata/asterisk/sounds/en/enter-a-time.gsm
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libdata/asterisk/sounds/en/enter-conf-call-number.gsm
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libdata/asterisk/sounds/en/enter-conf-pin-number.gsm
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libdata/asterisk/sounds/en/enter-ext-of-person.gsm
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libdata/asterisk/sounds/en/enter-num-blacklist.gsm
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libdata/asterisk/sounds/en/enter-password.gsm
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libdata/asterisk/sounds/en/enter-phone-number10.gsm
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libdata/asterisk/sounds/en/enter_account.gsm
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libdata/asterisk/sounds/en/entering-conf-number.gsm
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libdata/asterisk/sounds/en/entr-num-rmv-blklist.gsm
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libdata/asterisk/sounds/en/enum-lookup-failed.gsm
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libdata/asterisk/sounds/en/enum-lookup-successful.gsm
|
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libdata/asterisk/sounds/en/error-number.gsm
|
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libdata/asterisk/sounds/en/error.gsm
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libdata/asterisk/sounds/en/est-hold-time-is.gsm
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libdata/asterisk/sounds/en/euro.gsm
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libdata/asterisk/sounds/en/european.gsm
|
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libdata/asterisk/sounds/en/euros.gsm
|
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libdata/asterisk/sounds/en/evening.gsm
|
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libdata/asterisk/sounds/en/explanation.gsm
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libdata/asterisk/sounds/en/ext-or-zero.gsm
|
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libdata/asterisk/sounds/en/extension.gsm
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libdata/asterisk/sounds/en/extensions.gsm
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libdata/asterisk/sounds/en/extra-sounds-en.txt
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libdata/asterisk/sounds/en/facilities.gsm
|
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libdata/asterisk/sounds/en/fahrenheit.gsm
|
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libdata/asterisk/sounds/en/falling.gsm
|
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libdata/asterisk/sounds/en/fast.gsm
|
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libdata/asterisk/sounds/en/feature-not-avail-line.gsm
|
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libdata/asterisk/sounds/en/feeling_lucky_punk.gsm
|
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libdata/asterisk/sounds/en/feet.gsm
|
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libdata/asterisk/sounds/en/female.gsm
|
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libdata/asterisk/sounds/en/fifteenth.gsm
|
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libdata/asterisk/sounds/en/fifth.gsm
|
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libdata/asterisk/sounds/en/fiftieth.gsm
|
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libdata/asterisk/sounds/en/finals.gsm
|
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libdata/asterisk/sounds/en/finance.gsm
|
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libdata/asterisk/sounds/en/first-in-line.gsm
|
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libdata/asterisk/sounds/en/first.gsm
|
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libdata/asterisk/sounds/en/flagged-for-lea.gsm
|
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libdata/asterisk/sounds/en/flooding.gsm
|
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libdata/asterisk/sounds/en/florida.gsm
|
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libdata/asterisk/sounds/en/fog.gsm
|
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libdata/asterisk/sounds/en/foggy.gsm
|
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libdata/asterisk/sounds/en/followed-by.gsm
|
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libdata/asterisk/sounds/en/followed_hash.gsm
|
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libdata/asterisk/sounds/en/followed_pound.gsm
|
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libdata/asterisk/sounds/en/followme/call-from.gsm
|
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libdata/asterisk/sounds/en/followme/no-recording.gsm
|
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libdata/asterisk/sounds/en/followme/options.gsm
|
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libdata/asterisk/sounds/en/followme/pls-hold-while-try.gsm
|
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libdata/asterisk/sounds/en/followme/sorry.gsm
|
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libdata/asterisk/sounds/en/followme/status.gsm
|
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libdata/asterisk/sounds/en/food-service.gsm
|
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libdata/asterisk/sounds/en/food-services.gsm
|
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libdata/asterisk/sounds/en/for-a-daily-wakeup-call.gsm
|
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libdata/asterisk/sounds/en/for-a-list-of.gsm
|
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|
libdata/asterisk/sounds/en/for-accounting.gsm
|
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libdata/asterisk/sounds/en/for-billing.gsm
|
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libdata/asterisk/sounds/en/for-english-press.gsm
|
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libdata/asterisk/sounds/en/for-investor-relations.gsm
|
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|
libdata/asterisk/sounds/en/for-louie-louie.gsm
|
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|
libdata/asterisk/sounds/en/for-no-press.gsm
|
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|
libdata/asterisk/sounds/en/for-qc-and-training-purposes.gsm
|
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|
libdata/asterisk/sounds/en/for-quality-purposes.gsm
|
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|
libdata/asterisk/sounds/en/for-sales.gsm
|
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libdata/asterisk/sounds/en/for-service.gsm
|
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|
libdata/asterisk/sounds/en/for-tech-support.gsm
|
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libdata/asterisk/sounds/en/for-the-first.gsm
|
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|
libdata/asterisk/sounds/en/for-the-weather.gsm
|
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|
libdata/asterisk/sounds/en/for-wakeup-call.gsm
|
|
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|
libdata/asterisk/sounds/en/for-yes-press.gsm
|
|
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|
libdata/asterisk/sounds/en/for.gsm
|
|
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|
libdata/asterisk/sounds/en/forget_about_it.gsm
|
|
|
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|
libdata/asterisk/sounds/en/fort-worth.gsm
|
|
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|
libdata/asterisk/sounds/en/fortieth.gsm
|
|
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|
libdata/asterisk/sounds/en/fourteenth.gsm
|
|
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|
libdata/asterisk/sounds/en/fourth.gsm
|
|
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|
libdata/asterisk/sounds/en/freeze.gsm
|
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|
libdata/asterisk/sounds/en/freezing.gsm
|
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|
libdata/asterisk/sounds/en/frequency.gsm
|
|
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|
libdata/asterisk/sounds/en/fresno.gsm
|
|
|
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|
libdata/asterisk/sounds/en/from-unknown-caller.gsm
|
|
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|
libdata/asterisk/sounds/en/from.gsm
|
|
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|
libdata/asterisk/sounds/en/ftp.gsm
|
|
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|
libdata/asterisk/sounds/en/gale.gsm
|
|
|
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|
libdata/asterisk/sounds/en/gambling-drunk.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/georgia.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/get-in-line-sales-guy.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/get_bleep_outta.gsm
|
|
|
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|
libdata/asterisk/sounds/en/get_information.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/get_information_first.gsm
|
|
|
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|
libdata/asterisk/sounds/en/gigabits.gsm
|
|
|
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|
libdata/asterisk/sounds/en/gigabytes.gsm
|
|
|
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|
libdata/asterisk/sounds/en/gigahertz.gsm
|
|
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|
libdata/asterisk/sounds/en/giggle1.gsm
|
|
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|
libdata/asterisk/sounds/en/gmt.gsm
|
|
|
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|
libdata/asterisk/sounds/en/go-away1.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/go-away2.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/good-afternoon.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/good-evening.gsm
|
|
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|
libdata/asterisk/sounds/en/good-morning.gsm
|
|
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|
libdata/asterisk/sounds/en/good.gsm
|
|
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|
libdata/asterisk/sounds/en/goodbye.gsm
|
|
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|
libdata/asterisk/sounds/en/goodbye_for_the_best.gsm
|
|
|
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|
libdata/asterisk/sounds/en/goodbye_love.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/got_kidding.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/grammar.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/greater-than.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/groovy.gsm
|
|
|
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|
libdata/asterisk/sounds/en/ha/ac.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/ha/air-conditioner.gsm
|
|
|
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|
libdata/asterisk/sounds/en/ha/alarm.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/ha/amp.gsm
|
|
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|
libdata/asterisk/sounds/en/ha/amps.gsm
|
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|
libdata/asterisk/sounds/en/ha/attic.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/ha/baby-sleeping-mode.gsm
|
|
|
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|
libdata/asterisk/sounds/en/ha/back.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/ha/basement.gsm
|
|
|
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|
libdata/asterisk/sounds/en/ha/bathroom.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/ha/bedroom.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/ha/bright.gsm
|
|
|
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|
libdata/asterisk/sounds/en/ha/callerid.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/ha/carport.gsm
|
|
|
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|
libdata/asterisk/sounds/en/ha/closet.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/ha/coffee-pot.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/ha/cool.gsm
|
|
|
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|
libdata/asterisk/sounds/en/ha/cooling.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/ha/dc.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/ha/decibel.gsm
|
|
|
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|
libdata/asterisk/sounds/en/ha/decibels.gsm
|
|
|
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|
libdata/asterisk/sounds/en/ha/deck.gsm
|
|
|
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|
libdata/asterisk/sounds/en/ha/degree.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/ha/degrees.gsm
|
|
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|
libdata/asterisk/sounds/en/ha/den.gsm
|
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|
libdata/asterisk/sounds/en/ha/dim.gsm
|
|
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|
libdata/asterisk/sounds/en/ha/dining-room.gsm
|
|
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|
libdata/asterisk/sounds/en/ha/door.gsm
|
|
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|
libdata/asterisk/sounds/en/ha/doors.gsm
|
|
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|
libdata/asterisk/sounds/en/ha/down.gsm
|
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|
libdata/asterisk/sounds/en/ha/driveway.gsm
|
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|
libdata/asterisk/sounds/en/ha/dryer.gsm
|
|
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|
libdata/asterisk/sounds/en/ha/fan.gsm
|
|
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|
libdata/asterisk/sounds/en/ha/farad.gsm
|
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|
libdata/asterisk/sounds/en/ha/farads.gsm
|
|
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|
libdata/asterisk/sounds/en/ha/first-floor.gsm
|
|
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|
libdata/asterisk/sounds/en/ha/floor.gsm
|
|
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|
libdata/asterisk/sounds/en/ha/floors.gsm
|
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libdata/asterisk/sounds/en/ha/for-extended-status-report.gsm
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libdata/asterisk/sounds/en/ha/for-quick-status-report.gsm
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libdata/asterisk/sounds/en/ha/for-wx-report.gsm
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libdata/asterisk/sounds/en/ha/fountain.gsm
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libdata/asterisk/sounds/en/ha/foyer.gsm
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libdata/asterisk/sounds/en/ha/front.gsm
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libdata/asterisk/sounds/en/ha/furnace.gsm
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libdata/asterisk/sounds/en/ha/game-room.gsm
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libdata/asterisk/sounds/en/ha/garage.gsm
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libdata/asterisk/sounds/en/ha/great-room.gsm
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libdata/asterisk/sounds/en/ha/guest-room.gsm
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libdata/asterisk/sounds/en/ha/hall.gsm
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libdata/asterisk/sounds/en/ha/has-been-left.gsm
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libdata/asterisk/sounds/en/ha/heat-pump.gsm
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libdata/asterisk/sounds/en/ha/heat.gsm
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libdata/asterisk/sounds/en/ha/heating.gsm
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libdata/asterisk/sounds/en/ha/hot-tub.gsm
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libdata/asterisk/sounds/en/ha/house.gsm
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libdata/asterisk/sounds/en/ha/intruder.gsm
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libdata/asterisk/sounds/en/ha/is.gsm
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libdata/asterisk/sounds/en/ha/kelvin.gsm
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libdata/asterisk/sounds/en/ha/kitchen.gsm
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libdata/asterisk/sounds/en/ha/lamp.gsm
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libdata/asterisk/sounds/en/ha/lamps.gsm
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libdata/asterisk/sounds/en/ha/landscape.gsm
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libdata/asterisk/sounds/en/ha/laundry.gsm
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libdata/asterisk/sounds/en/ha/library.gsm
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libdata/asterisk/sounds/en/ha/light.gsm
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libdata/asterisk/sounds/en/ha/lights.gsm
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libdata/asterisk/sounds/en/ha/living-room.gsm
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libdata/asterisk/sounds/en/ha/locked.gsm
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libdata/asterisk/sounds/en/ha/locking.gsm
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libdata/asterisk/sounds/en/ha/mailbox.gsm
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libdata/asterisk/sounds/en/ha/master.gsm
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libdata/asterisk/sounds/en/ha/off.gsm
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libdata/asterisk/sounds/en/ha/office.gsm
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libdata/asterisk/sounds/en/ha/ohm.gsm
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libdata/asterisk/sounds/en/ha/ohms.gsm
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libdata/asterisk/sounds/en/ha/on.gsm
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libdata/asterisk/sounds/en/ha/open.gsm
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libdata/asterisk/sounds/en/ha/patio.gsm
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libdata/asterisk/sounds/en/ha/phone.gsm
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libdata/asterisk/sounds/en/ha/play-room.gsm
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libdata/asterisk/sounds/en/ha/play.gsm
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libdata/asterisk/sounds/en/ha/pool.gsm
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libdata/asterisk/sounds/en/ha/porch.gsm
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libdata/asterisk/sounds/en/ha/power-failure.gsm
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libdata/asterisk/sounds/en/ha/pressure.gsm
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libdata/asterisk/sounds/en/ha/psi.gsm
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libdata/asterisk/sounds/en/ha/quiet-mode.gsm
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libdata/asterisk/sounds/en/ha/reset.gsm
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libdata/asterisk/sounds/en/ha/roof.gsm
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libdata/asterisk/sounds/en/ha/room.gsm
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libdata/asterisk/sounds/en/ha/rooms.gsm
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libdata/asterisk/sounds/en/ha/second-floor.gsm
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libdata/asterisk/sounds/en/ha/secure.gsm
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libdata/asterisk/sounds/en/ha/security-system.gsm
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libdata/asterisk/sounds/en/ha/set.gsm
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libdata/asterisk/sounds/en/ha/side.gsm
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libdata/asterisk/sounds/en/ha/solar.gsm
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libdata/asterisk/sounds/en/ha/sprinklers.gsm
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libdata/asterisk/sounds/en/ha/still.gsm
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libdata/asterisk/sounds/en/ha/stove.gsm
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libdata/asterisk/sounds/en/ha/sump-pump.gsm
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libdata/asterisk/sounds/en/ha/sun-room.gsm
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libdata/asterisk/sounds/en/ha/system.gsm
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libdata/asterisk/sounds/en/ha/systems.gsm
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libdata/asterisk/sounds/en/ha/thermostat.gsm
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libdata/asterisk/sounds/en/ha/to-control-environ-sys.gsm
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libdata/asterisk/sounds/en/ha/to-control-lights-appl.gsm
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libdata/asterisk/sounds/en/ha/tower.gsm
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libdata/asterisk/sounds/en/ha/unlocked.gsm
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libdata/asterisk/sounds/en/ha/unlocking.gsm
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libdata/asterisk/sounds/en/ha/up.gsm
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libdata/asterisk/sounds/en/ha/volt.gsm
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libdata/asterisk/sounds/en/ha/volts.gsm
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libdata/asterisk/sounds/en/ha/washing-machine.gsm
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libdata/asterisk/sounds/en/ha/water-heater.gsm
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libdata/asterisk/sounds/en/ha/watt.gsm
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libdata/asterisk/sounds/en/ha/watts.gsm
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libdata/asterisk/sounds/en/ha/well-pump.gsm
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libdata/asterisk/sounds/en/ha/window.gsm
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libdata/asterisk/sounds/en/ha/windows.gsm
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libdata/asterisk/sounds/en/ha/xmas-lights.gsm
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libdata/asterisk/sounds/en/ha/xmas-tree.gsm
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libdata/asterisk/sounds/en/ha/yard.gsm
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libdata/asterisk/sounds/en/hail.gsm
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libdata/asterisk/sounds/en/hal_goodbye.gsm
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libdata/asterisk/sounds/en/half.gsm
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libdata/asterisk/sounds/en/hang-on-a-second-angry.gsm
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libdata/asterisk/sounds/en/hang-on-a-second.gsm
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libdata/asterisk/sounds/en/hangup-try-again.gsm
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libdata/asterisk/sounds/en/happy_saved.gsm
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libdata/asterisk/sounds/en/has-arrived-at.gsm
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libdata/asterisk/sounds/en/has-been-changed-to.gsm
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libdata/asterisk/sounds/en/has-been-cleared.gsm
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libdata/asterisk/sounds/en/has-been-disconnected.gsm
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libdata/asterisk/sounds/en/has-been-set-to.gsm
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libdata/asterisk/sounds/en/has-been.gsm
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libdata/asterisk/sounds/en/has-expired.gsm
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libdata/asterisk/sounds/en/has-issued-a.gsm
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libdata/asterisk/sounds/en/has-not-been-seen-for.gsm
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libdata/asterisk/sounds/en/has.gsm
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libdata/asterisk/sounds/en/hash.gsm
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libdata/asterisk/sounds/en/hawaii.gsm
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libdata/asterisk/sounds/en/headed-towards.gsm
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libdata/asterisk/sounds/en/heading.gsm
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libdata/asterisk/sounds/en/health-center.gsm
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libdata/asterisk/sounds/en/hear-odd-noise.gsm
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libdata/asterisk/sounds/en/hear-toilet-flush.gsm
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libdata/asterisk/sounds/en/hectopascal.gsm
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libdata/asterisk/sounds/en/hello-world.gsm
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libdata/asterisk/sounds/en/hello.gsm
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libdata/asterisk/sounds/en/helpdesk.gsm
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libdata/asterisk/sounds/en/hertz.gsm
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libdata/asterisk/sounds/en/high.gsm
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libdata/asterisk/sounds/en/highway.gsm
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libdata/asterisk/sounds/en/hit.gsm
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libdata/asterisk/sounds/en/hold-or-dial-0.gsm
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libdata/asterisk/sounds/en/home.gsm
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libdata/asterisk/sounds/en/honolulu.gsm
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libdata/asterisk/sounds/en/hours.gsm
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libdata/asterisk/sounds/en/housekeeping.gsm
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libdata/asterisk/sounds/en/houston.gsm
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libdata/asterisk/sounds/en/http.gsm
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libdata/asterisk/sounds/en/human-resources.gsm
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libdata/asterisk/sounds/en/humidity.gsm
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libdata/asterisk/sounds/en/hundredth.gsm
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libdata/asterisk/sounds/en/hurricane.gsm
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libdata/asterisk/sounds/en/hz.gsm
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libdata/asterisk/sounds/en/i-dont-understand.gsm
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libdata/asterisk/sounds/en/i-dont-understand2.gsm
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libdata/asterisk/sounds/en/i-dont-understand3.gsm
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libdata/asterisk/sounds/en/i-dont-understand4.gsm
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libdata/asterisk/sounds/en/i-dont-understand5.gsm
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libdata/asterisk/sounds/en/i-grow-bored.gsm
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libdata/asterisk/sounds/en/ice.gsm
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libdata/asterisk/sounds/en/icmp.gsm
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libdata/asterisk/sounds/en/icy.gsm
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libdata/asterisk/sounds/en/idaho.gsm
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libdata/asterisk/sounds/en/if-correct-press.gsm
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libdata/asterisk/sounds/en/if-grtg-played-indefinately.gsm
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libdata/asterisk/sounds/en/if-grtg-should-expire-at.gsm
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libdata/asterisk/sounds/en/if-maint-contract-or-emergency.gsm
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libdata/asterisk/sounds/en/if-rotary-phone.gsm
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libdata/asterisk/sounds/en/if-this-is-correct-press.gsm
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libdata/asterisk/sounds/en/if-this-is-correct.gsm
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libdata/asterisk/sounds/en/if-this-is-not-correct.gsm
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libdata/asterisk/sounds/en/if-u-know-ext-dial.gsm
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libdata/asterisk/sounds/en/if-unsuccessful-speak-to.gsm
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libdata/asterisk/sounds/en/if-you-know-the.gsm
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libdata/asterisk/sounds/en/if-you-need-help.gsm
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libdata/asterisk/sounds/en/if-youd-like-to-make-a-call.gsm
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libdata/asterisk/sounds/en/illinois.gsm
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libdata/asterisk/sounds/en/im-sorry-unable-to-connect-to-eng.gsm
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libdata/asterisk/sounds/en/im-sorry.gsm
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libdata/asterisk/sounds/en/imap.gsm
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libdata/asterisk/sounds/en/in-service.gsm
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libdata/asterisk/sounds/en/in-the-line.gsm
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libdata/asterisk/sounds/en/in-the-queue.gsm
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libdata/asterisk/sounds/en/in-the.gsm
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libdata/asterisk/sounds/en/in-your-city.gsm
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libdata/asterisk/sounds/en/in-your-zip-code.gsm
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libdata/asterisk/sounds/en/inbound.gsm
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libdata/asterisk/sounds/en/indiana.gsm
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libdata/asterisk/sounds/en/indianapolis.gsm
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libdata/asterisk/sounds/en/indicated.gsm
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libdata/asterisk/sounds/en/info-about-last-call.gsm
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libdata/asterisk/sounds/en/information-technology.gsm
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libdata/asterisk/sounds/en/information.gsm
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libdata/asterisk/sounds/en/infuriate-tech-staff.gsm
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libdata/asterisk/sounds/en/initiated.gsm
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libdata/asterisk/sounds/en/initiating.gsm
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libdata/asterisk/sounds/en/inside-sales.gsm
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libdata/asterisk/sounds/en/internal-audit.gsm
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libdata/asterisk/sounds/en/international-call.gsm
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libdata/asterisk/sounds/en/interstate.gsm
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libdata/asterisk/sounds/en/invalid-date.gsm
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libdata/asterisk/sounds/en/invalid-featurecode.gsm
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libdata/asterisk/sounds/en/invalid.gsm
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libdata/asterisk/sounds/en/investor-relations.gsm
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libdata/asterisk/sounds/en/iowa.gsm
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libdata/asterisk/sounds/en/is-at.gsm
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libdata/asterisk/sounds/en/is-curntly-busy.gsm
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libdata/asterisk/sounds/en/is-curntly-unavail.gsm
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libdata/asterisk/sounds/en/is-currently.gsm
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libdata/asterisk/sounds/en/is-in-use.gsm
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libdata/asterisk/sounds/en/is-not-in-the.gsm
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libdata/asterisk/sounds/en/is-not-set.gsm
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libdata/asterisk/sounds/en/is-now-being-recorded.gsm
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libdata/asterisk/sounds/en/is-set-to.gsm
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libdata/asterisk/sounds/en/is.gsm
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libdata/asterisk/sounds/en/it-now.gsm
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libdata/asterisk/sounds/en/it-services.gsm
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libdata/asterisk/sounds/en/jacksonville.gsm
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libdata/asterisk/sounds/en/janitorial.gsm
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libdata/asterisk/sounds/en/jason.gsm
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libdata/asterisk/sounds/en/jedi-extension-trick.gsm
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libdata/asterisk/sounds/en/john.gsm
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libdata/asterisk/sounds/en/just-kidding-not-upset.gsm
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libdata/asterisk/sounds/en/just-kidding-not-upset2.gsm
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libdata/asterisk/sounds/en/kansas-city.gsm
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libdata/asterisk/sounds/en/kansas.gsm
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libdata/asterisk/sounds/en/kentucky.gsm
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libdata/asterisk/sounds/en/keywords_cross_fingers.gsm
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libdata/asterisk/sounds/en/kilobits.gsm
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libdata/asterisk/sounds/en/kilobytes.gsm
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libdata/asterisk/sounds/en/kilohertz.gsm
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libdata/asterisk/sounds/en/kilometer.gsm
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libdata/asterisk/sounds/en/kilometers-per-hour.gsm
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libdata/asterisk/sounds/en/knock-knock.gsm
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libdata/asterisk/sounds/en/knots.gsm
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libdata/asterisk/sounds/en/language.gsm
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libdata/asterisk/sounds/en/las-vegas.gsm
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libdata/asterisk/sounds/en/last-error-was.gsm
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libdata/asterisk/sounds/en/last-num-to-call.gsm
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libdata/asterisk/sounds/en/late.gsm
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libdata/asterisk/sounds/en/later.gsm
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libdata/asterisk/sounds/en/lea-may-request-info.gsm
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libdata/asterisk/sounds/en/left-bracket.gsm
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libdata/asterisk/sounds/en/legal.gsm
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libdata/asterisk/sounds/en/len.gsm
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libdata/asterisk/sounds/en/less-than.gsm
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libdata/asterisk/sounds/en/letters/a.gsm
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libdata/asterisk/sounds/en/letters/ascii123.gsm
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libdata/asterisk/sounds/en/letters/ascii124.gsm
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libdata/asterisk/sounds/en/letters/ascii125.gsm
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libdata/asterisk/sounds/en/letters/ascii126.gsm
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libdata/asterisk/sounds/en/letters/ascii34.gsm
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libdata/asterisk/sounds/en/letters/ascii36.gsm
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libdata/asterisk/sounds/en/letters/ascii37.gsm
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libdata/asterisk/sounds/en/letters/ascii38.gsm
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libdata/asterisk/sounds/en/letters/ascii39.gsm
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libdata/asterisk/sounds/en/letters/ascii40.gsm
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libdata/asterisk/sounds/en/letters/ascii41.gsm
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libdata/asterisk/sounds/en/letters/ascii42.gsm
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libdata/asterisk/sounds/en/letters/ascii44.gsm
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libdata/asterisk/sounds/en/letters/ascii58.gsm
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libdata/asterisk/sounds/en/letters/ascii59.gsm
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libdata/asterisk/sounds/en/letters/ascii60.gsm
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libdata/asterisk/sounds/en/letters/ascii62.gsm
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libdata/asterisk/sounds/en/letters/ascii63.gsm
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libdata/asterisk/sounds/en/letters/ascii91.gsm
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libdata/asterisk/sounds/en/letters/ascii92.gsm
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libdata/asterisk/sounds/en/letters/ascii93.gsm
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libdata/asterisk/sounds/en/letters/ascii94.gsm
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libdata/asterisk/sounds/en/letters/ascii95.gsm
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libdata/asterisk/sounds/en/letters/ascii96.gsm
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libdata/asterisk/sounds/en/letters/asterisk.gsm
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libdata/asterisk/sounds/en/letters/at.gsm
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libdata/asterisk/sounds/en/letters/b.gsm
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libdata/asterisk/sounds/en/letters/c.gsm
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libdata/asterisk/sounds/en/letters/d.gsm
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libdata/asterisk/sounds/en/letters/dash.gsm
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libdata/asterisk/sounds/en/letters/dollar.gsm
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libdata/asterisk/sounds/en/letters/dot.gsm
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libdata/asterisk/sounds/en/letters/e.gsm
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libdata/asterisk/sounds/en/letters/equals.gsm
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libdata/asterisk/sounds/en/letters/exclaimation-point.gsm
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libdata/asterisk/sounds/en/letters/f.gsm
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libdata/asterisk/sounds/en/letters/g.gsm
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libdata/asterisk/sounds/en/letters/h.gsm
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libdata/asterisk/sounds/en/letters/i.gsm
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libdata/asterisk/sounds/en/letters/j.gsm
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libdata/asterisk/sounds/en/letters/k.gsm
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libdata/asterisk/sounds/en/letters/l.gsm
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libdata/asterisk/sounds/en/letters/m.gsm
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libdata/asterisk/sounds/en/letters/n.gsm
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libdata/asterisk/sounds/en/letters/o.gsm
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libdata/asterisk/sounds/en/letters/p.gsm
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libdata/asterisk/sounds/en/letters/plus.gsm
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libdata/asterisk/sounds/en/letters/q.gsm
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libdata/asterisk/sounds/en/letters/r.gsm
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libdata/asterisk/sounds/en/letters/s.gsm
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libdata/asterisk/sounds/en/letters/slash.gsm
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libdata/asterisk/sounds/en/letters/space.gsm
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libdata/asterisk/sounds/en/letters/t.gsm
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libdata/asterisk/sounds/en/letters/u.gsm
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libdata/asterisk/sounds/en/letters/v.gsm
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libdata/asterisk/sounds/en/letters/w.gsm
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libdata/asterisk/sounds/en/letters/x.gsm
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libdata/asterisk/sounds/en/letters/y.gsm
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libdata/asterisk/sounds/en/letters/z.gsm
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libdata/asterisk/sounds/en/letters/zed.gsm
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libdata/asterisk/sounds/en/library.gsm
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libdata/asterisk/sounds/en/lightning.gsm
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libdata/asterisk/sounds/en/like_to_tell_valid_ext.gsm
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libdata/asterisk/sounds/en/limit-simul-calls.gsm
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libdata/asterisk/sounds/en/lines-complaining-customers.gsm
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libdata/asterisk/sounds/en/linux.gsm
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libdata/asterisk/sounds/en/list.gsm
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libdata/asterisk/sounds/en/load-average.gsm
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libdata/asterisk/sounds/en/local-authorities.gsm
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libdata/asterisk/sounds/en/location.gsm
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libdata/asterisk/sounds/en/login-fail.gsm
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libdata/asterisk/sounds/en/long-beach.gsm
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libdata/asterisk/sounds/en/los-angeles.gsm
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libdata/asterisk/sounds/en/loss-prevention.gsm
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libdata/asterisk/sounds/en/loss.gsm
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libdata/asterisk/sounds/en/lots-o-monkeys.gsm
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libdata/asterisk/sounds/en/louisiana.gsm
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libdata/asterisk/sounds/en/low.gsm
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libdata/asterisk/sounds/en/lowercase.gsm
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libdata/asterisk/sounds/en/lunch.gsm
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libdata/asterisk/sounds/en/lyrics-louie-louie.gsm
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libdata/asterisk/sounds/en/machine.gsm
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libdata/asterisk/sounds/en/made-it-up.gsm
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libdata/asterisk/sounds/en/mail.gsm
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libdata/asterisk/sounds/en/mailroom.gsm
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libdata/asterisk/sounds/en/main-menu.gsm
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libdata/asterisk/sounds/en/maine.gsm
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libdata/asterisk/sounds/en/maintenance.gsm
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libdata/asterisk/sounds/en/male.gsm
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libdata/asterisk/sounds/en/management.gsm
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libdata/asterisk/sounds/en/manufacturing.gsm
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libdata/asterisk/sounds/en/marketing.gsm
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libdata/asterisk/sounds/en/marryme.gsm
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libdata/asterisk/sounds/en/martini.gsm
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libdata/asterisk/sounds/en/maryland.gsm
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libdata/asterisk/sounds/en/massachusetts.gsm
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libdata/asterisk/sounds/en/mathematics.gsm
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libdata/asterisk/sounds/en/maximum.gsm
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libdata/asterisk/sounds/en/megabits.gsm
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libdata/asterisk/sounds/en/megabytes.gsm
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libdata/asterisk/sounds/en/megahertz.gsm
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libdata/asterisk/sounds/en/memory.gsm
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libdata/asterisk/sounds/en/memphis.gsm
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libdata/asterisk/sounds/en/menu.gsm
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libdata/asterisk/sounds/en/mesa.gsm
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libdata/asterisk/sounds/en/message-from.gsm
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libdata/asterisk/sounds/en/message-number.gsm
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libdata/asterisk/sounds/en/messages_curious.gsm
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libdata/asterisk/sounds/en/meter.gsm
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libdata/asterisk/sounds/en/meters.gsm
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libdata/asterisk/sounds/en/miami.gsm
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libdata/asterisk/sounds/en/michigan.gsm
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libdata/asterisk/sounds/en/midnight-tomorrow-night.gsm
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libdata/asterisk/sounds/en/midnight-tonight.gsm
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libdata/asterisk/sounds/en/midnight.gsm
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libdata/asterisk/sounds/en/mike.gsm
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libdata/asterisk/sounds/en/miles-per-hour.gsm
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libdata/asterisk/sounds/en/miles.gsm
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libdata/asterisk/sounds/en/millionth.gsm
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libdata/asterisk/sounds/en/milwaukee.gsm
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libdata/asterisk/sounds/en/mind_repeating.gsm
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libdata/asterisk/sounds/en/minimum.gsm
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libdata/asterisk/sounds/en/minions-not-answering-leave-message.gsm
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libdata/asterisk/sounds/en/minneapolis.gsm
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libdata/asterisk/sounds/en/minnesota.gsm
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libdata/asterisk/sounds/en/minute.gsm
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libdata/asterisk/sounds/en/minutes.gsm
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libdata/asterisk/sounds/en/missed.gsm
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libdata/asterisk/sounds/en/mississippi.gsm
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libdata/asterisk/sounds/en/missouri.gsm
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libdata/asterisk/sounds/en/misty.gsm
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libdata/asterisk/sounds/en/mode.gsm
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libdata/asterisk/sounds/en/monitored.gsm
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libdata/asterisk/sounds/en/montana.gsm
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libdata/asterisk/sounds/en/month.gsm
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libdata/asterisk/sounds/en/months.gsm
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libdata/asterisk/sounds/en/moo1.gsm
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libdata/asterisk/sounds/en/moo2.gsm
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libdata/asterisk/sounds/en/morning.gsm
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libdata/asterisk/sounds/en/moron.gsm
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libdata/asterisk/sounds/en/mostly.gsm
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libdata/asterisk/sounds/en/motor-pool.gsm
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libdata/asterisk/sounds/en/mountain.gsm
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libdata/asterisk/sounds/en/moving.gsm
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libdata/asterisk/sounds/en/ms.gsm
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libdata/asterisk/sounds/en/nashville.gsm
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libdata/asterisk/sounds/en/national-weather-service.gsm
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libdata/asterisk/sounds/en/nautical-miles.gsm
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libdata/asterisk/sounds/en/nbdy-avail-to-take-call.gsm
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libdata/asterisk/sounds/en/near.gsm
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libdata/asterisk/sounds/en/nebraska.gsm
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libdata/asterisk/sounds/en/negative.gsm
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libdata/asterisk/sounds/en/network-operations-center.gsm
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libdata/asterisk/sounds/en/network-operations.gsm
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libdata/asterisk/sounds/en/nevada.gsm
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libdata/asterisk/sounds/en/new-accounts.gsm
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libdata/asterisk/sounds/en/new-hampshire.gsm
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libdata/asterisk/sounds/en/new-jersey.gsm
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libdata/asterisk/sounds/en/new-mexico.gsm
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libdata/asterisk/sounds/en/new-orleans.gsm
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libdata/asterisk/sounds/en/new-york.gsm
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libdata/asterisk/sounds/en/night.gsm
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libdata/asterisk/sounds/en/nineteenth.gsm
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libdata/asterisk/sounds/en/ninetieth.gsm
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libdata/asterisk/sounds/en/ninth.gsm
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libdata/asterisk/sounds/en/no-112-1.gsm
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libdata/asterisk/sounds/en/no-112-2.gsm
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libdata/asterisk/sounds/en/no-911-1.gsm
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libdata/asterisk/sounds/en/no-911-2.gsm
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libdata/asterisk/sounds/en/no-empty-conferences.gsm
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libdata/asterisk/sounds/en/no-info-about-number.gsm
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libdata/asterisk/sounds/en/no-longer-in-service.gsm
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libdata/asterisk/sounds/en/no-reply-no-mailbox.gsm
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libdata/asterisk/sounds/en/no-route-exists-to-dest.gsm
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libdata/asterisk/sounds/en/no_invite_to_conf.gsm
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libdata/asterisk/sounds/en/no_longer_conf.gsm
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libdata/asterisk/sounds/en/no_problem_help.gsm
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libdata/asterisk/sounds/en/no_worries_try_again.gsm
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libdata/asterisk/sounds/en/nobody-but-chickens.gsm
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libdata/asterisk/sounds/en/node.gsm
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libdata/asterisk/sounds/en/none_of_my_business1.gsm
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libdata/asterisk/sounds/en/none_of_my_business2.gsm
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libdata/asterisk/sounds/en/north-carolina.gsm
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libdata/asterisk/sounds/en/north-dakota.gsm
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libdata/asterisk/sounds/en/north.gsm
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libdata/asterisk/sounds/en/northerly.gsm
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libdata/asterisk/sounds/en/not-auth-pstn.gsm
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libdata/asterisk/sounds/en/not-enough-credit.gsm
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libdata/asterisk/sounds/en/not-necessary-1.gsm
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libdata/asterisk/sounds/en/not-necessary-ac.gsm
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libdata/asterisk/sounds/en/not-necessary-dial-1-or-ac.gsm
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libdata/asterisk/sounds/en/not-rqsted-wakeup.gsm
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libdata/asterisk/sounds/en/not-taking-your-call.gsm
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libdata/asterisk/sounds/en/not-yet-assigned.gsm
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libdata/asterisk/sounds/en/not-yet-connected.gsm
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libdata/asterisk/sounds/en/not_me.gsm
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libdata/asterisk/sounds/en/not_pass.gsm
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libdata/asterisk/sounds/en/not_siri.gsm
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libdata/asterisk/sounds/en/not_you.gsm
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libdata/asterisk/sounds/en/nothing-recorded.gsm
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libdata/asterisk/sounds/en/now.gsm
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libdata/asterisk/sounds/en/num-not-in-db.gsm
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libdata/asterisk/sounds/en/num-outside-area.gsm
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libdata/asterisk/sounds/en/num-was-successfully.gsm
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libdata/asterisk/sounds/en/number-not-answering.gsm
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libdata/asterisk/sounds/en/number.gsm
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libdata/asterisk/sounds/en/oakland.gsm
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libdata/asterisk/sounds/en/octothorpe.gsm
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libdata/asterisk/sounds/en/off-duty.gsm
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libdata/asterisk/sounds/en/off.gsm
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libdata/asterisk/sounds/en/office-code.gsm
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libdata/asterisk/sounds/en/office-iguanas.gsm
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libdata/asterisk/sounds/en/office.gsm
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libdata/asterisk/sounds/en/ogm_home.gsm
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libdata/asterisk/sounds/en/ohio.gsm
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libdata/asterisk/sounds/en/oklahoma-city.gsm
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libdata/asterisk/sounds/en/oklahoma.gsm
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libdata/asterisk/sounds/en/omaha.gsm
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libdata/asterisk/sounds/en/on-busy.gsm
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libdata/asterisk/sounds/en/on-monthly-tel-stment.gsm
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libdata/asterisk/sounds/en/on-no-answer.gsm
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libdata/asterisk/sounds/en/on.gsm
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libdata/asterisk/sounds/en/one-moment-please.gsm
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libdata/asterisk/sounds/en/one-small-step.gsm
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libdata/asterisk/sounds/en/one-small-step2.gsm
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libdata/asterisk/sounds/en/oops1.gsm
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libdata/asterisk/sounds/en/oops2.gsm
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libdata/asterisk/sounds/en/oops3.gsm
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libdata/asterisk/sounds/en/open-parenthesis.gsm
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libdata/asterisk/sounds/en/open.gsm
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libdata/asterisk/sounds/en/operations.gsm
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libdata/asterisk/sounds/en/option-is-invalid.gsm
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libdata/asterisk/sounds/en/option-not-implemented.gsm
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libdata/asterisk/sounds/en/or-press.gsm
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libdata/asterisk/sounds/en/or.gsm
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libdata/asterisk/sounds/en/order-desk.gsm
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libdata/asterisk/sounds/en/orders.gsm
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libdata/asterisk/sounds/en/oregon.gsm
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libdata/asterisk/sounds/en/other-options-exercise.gsm
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libdata/asterisk/sounds/en/otherwise-press.gsm
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libdata/asterisk/sounds/en/otherwise.gsm
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libdata/asterisk/sounds/en/our-business-hours-are.gsm
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libdata/asterisk/sounds/en/outbound.gsm
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libdata/asterisk/sounds/en/outside-sales.gsm
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libdata/asterisk/sounds/en/outside-transfer.gsm
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libdata/asterisk/sounds/en/pacific.gsm
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libdata/asterisk/sounds/en/packet.gsm
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libdata/asterisk/sounds/en/panic.gsm
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libdata/asterisk/sounds/en/partially.gsm
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libdata/asterisk/sounds/en/partly.gsm
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libdata/asterisk/sounds/en/pascal.gsm
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libdata/asterisk/sounds/en/pascal2.gsm
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libdata/asterisk/sounds/en/passwords_not_match.gsm
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libdata/asterisk/sounds/en/patchy.gsm
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libdata/asterisk/sounds/en/pbx-invalid.gsm
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libdata/asterisk/sounds/en/pbx-invalidpark.gsm
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libdata/asterisk/sounds/en/pbx-parkingfailed.gsm
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libdata/asterisk/sounds/en/pbx-transfer.gsm
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libdata/asterisk/sounds/en/pence.gsm
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libdata/asterisk/sounds/en/pennies.gsm
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libdata/asterisk/sounds/en/pennsylvania.gsm
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libdata/asterisk/sounds/en/penny.gsm
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libdata/asterisk/sounds/en/percent.gsm
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libdata/asterisk/sounds/en/perhaps-we-are.gsm
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libdata/asterisk/sounds/en/perhaps-we-are2.gsm
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libdata/asterisk/sounds/en/period.gsm
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libdata/asterisk/sounds/en/personnel.gsm
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libdata/asterisk/sounds/en/persons-in-path-of.gsm
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libdata/asterisk/sounds/en/philadelphia.gsm
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libdata/asterisk/sounds/en/phoenix.gsm
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libdata/asterisk/sounds/en/phonetic/9_p.gsm
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libdata/asterisk/sounds/en/phonetic/a_p.gsm
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libdata/asterisk/sounds/en/phonetic/b_p.gsm
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libdata/asterisk/sounds/en/phonetic/c_p.gsm
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libdata/asterisk/sounds/en/phonetic/d_p.gsm
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libdata/asterisk/sounds/en/phonetic/e_p.gsm
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libdata/asterisk/sounds/en/phonetic/f_p.gsm
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libdata/asterisk/sounds/en/phonetic/g_p.gsm
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libdata/asterisk/sounds/en/phonetic/h_p.gsm
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libdata/asterisk/sounds/en/phonetic/i_p.gsm
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libdata/asterisk/sounds/en/phonetic/j_p.gsm
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libdata/asterisk/sounds/en/phonetic/k_p.gsm
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libdata/asterisk/sounds/en/phonetic/l_p.gsm
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libdata/asterisk/sounds/en/phonetic/m_p.gsm
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libdata/asterisk/sounds/en/phonetic/n_p.gsm
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libdata/asterisk/sounds/en/phonetic/o_p.gsm
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libdata/asterisk/sounds/en/phonetic/p_p.gsm
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libdata/asterisk/sounds/en/phonetic/q_p.gsm
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libdata/asterisk/sounds/en/phonetic/r_p.gsm
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libdata/asterisk/sounds/en/phonetic/s_p.gsm
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libdata/asterisk/sounds/en/phonetic/t_p.gsm
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libdata/asterisk/sounds/en/phonetic/u_p.gsm
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libdata/asterisk/sounds/en/phonetic/v_p.gsm
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libdata/asterisk/sounds/en/phonetic/w_p.gsm
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libdata/asterisk/sounds/en/phonetic/x_p.gsm
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libdata/asterisk/sounds/en/phonetic/y_p.gsm
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libdata/asterisk/sounds/en/phonetic/z_p.gsm
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libdata/asterisk/sounds/en/physics.gsm
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libdata/asterisk/sounds/en/pin-invalid.gsm
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libdata/asterisk/sounds/en/pin-number-accepted.gsm
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libdata/asterisk/sounds/en/pin_number.gsm
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libdata/asterisk/sounds/en/ping.gsm
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libdata/asterisk/sounds/en/pipe.gsm
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libdata/asterisk/sounds/en/planning.gsm
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libdata/asterisk/sounds/en/please-answer-the-following.gsm
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libdata/asterisk/sounds/en/please-contact-tech-supt.gsm
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libdata/asterisk/sounds/en/please-enter-first-three-letters.gsm
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libdata/asterisk/sounds/en/please-enter-the.gsm
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libdata/asterisk/sounds/en/please-enter-your.gsm
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libdata/asterisk/sounds/en/please-hang-up-and-dial-operator.gsm
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libdata/asterisk/sounds/en/please-hang-up-and-try-again.gsm
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libdata/asterisk/sounds/en/please-hold-minion-connect.gsm
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libdata/asterisk/sounds/en/please-hold-while-minion.gsm
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libdata/asterisk/sounds/en/please-try-again-later.gsm
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libdata/asterisk/sounds/en/please-try-again.gsm
|
2016-05-06 09:41:06 +02:00
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libdata/asterisk/sounds/en/please-try-call-later.gsm
|
2015-12-06 00:29:05 +01:00
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libdata/asterisk/sounds/en/please-try.gsm
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libdata/asterisk/sounds/en/please-wait-connect-oncall-eng.gsm
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libdata/asterisk/sounds/en/pls-ent-num-transfer.gsm
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libdata/asterisk/sounds/en/pls-enter-conf-password.gsm
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libdata/asterisk/sounds/en/pls-enter-num-message-after-tone.gsm
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libdata/asterisk/sounds/en/pls-enter-vm-password.gsm
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libdata/asterisk/sounds/en/pls-entr-num-uwish2-call.gsm
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libdata/asterisk/sounds/en/pls-hold-process-tx.gsm
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libdata/asterisk/sounds/en/pls-hold-silent30.gsm
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libdata/asterisk/sounds/en/pls-hold-while-try.gsm
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libdata/asterisk/sounds/en/pls-listen-options-changed.gsm
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libdata/asterisk/sounds/en/pls-lv-msg-will-contact.gsm
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libdata/asterisk/sounds/en/pls-rcrd-name-at-tone.gsm
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libdata/asterisk/sounds/en/pls-stay-on-line.gsm
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libdata/asterisk/sounds/en/pls-try-again.gsm
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libdata/asterisk/sounds/en/pls-try-area-code.gsm
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libdata/asterisk/sounds/en/pls-try-manually.gsm
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libdata/asterisk/sounds/en/pls-wait-connect-call.gsm
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libdata/asterisk/sounds/en/plugh.gsm
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libdata/asterisk/sounds/en/pm-announcement-number.gsm
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libdata/asterisk/sounds/en/pm-invalid-option.gsm
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libdata/asterisk/sounds/en/pm-phrase-management.gsm
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libdata/asterisk/sounds/en/pm-prompt-number.gsm
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libdata/asterisk/sounds/en/pm-to-record-phrase.gsm
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libdata/asterisk/sounds/en/pm-to-review-phrase.gsm
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libdata/asterisk/sounds/en/point.gsm
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libdata/asterisk/sounds/en/pop.gsm
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libdata/asterisk/sounds/en/port.gsm
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libdata/asterisk/sounds/en/portland.gsm
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libdata/asterisk/sounds/en/portnumber.gsm
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libdata/asterisk/sounds/en/position.gsm
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libdata/asterisk/sounds/en/post-entry-pound.gsm
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libdata/asterisk/sounds/en/pounds.gsm
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libdata/asterisk/sounds/en/presales-support.gsm
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libdata/asterisk/sounds/en/presidents-office.gsm
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libdata/asterisk/sounds/en/press-0.gsm
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libdata/asterisk/sounds/en/press-1.gsm
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libdata/asterisk/sounds/en/press-2.gsm
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libdata/asterisk/sounds/en/press-3.gsm
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libdata/asterisk/sounds/en/press-4.gsm
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libdata/asterisk/sounds/en/press-5.gsm
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libdata/asterisk/sounds/en/press-6-to-eject.gsm
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libdata/asterisk/sounds/en/press-6.gsm
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libdata/asterisk/sounds/en/press-7.gsm
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libdata/asterisk/sounds/en/press-8.gsm
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libdata/asterisk/sounds/en/press-9.gsm
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libdata/asterisk/sounds/en/press-button-again.gsm
|
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libdata/asterisk/sounds/en/press-enter.gsm
|
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libdata/asterisk/sounds/en/press-escape.gsm
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libdata/asterisk/sounds/en/press-hash.gsm
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libdata/asterisk/sounds/en/press-or-say-0.gsm
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libdata/asterisk/sounds/en/press-or-say-1.gsm
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libdata/asterisk/sounds/en/press-or-say-2.gsm
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libdata/asterisk/sounds/en/press-or-say-3.gsm
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libdata/asterisk/sounds/en/press-or-say-4.gsm
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libdata/asterisk/sounds/en/press-or-say-5.gsm
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libdata/asterisk/sounds/en/press-or-say-6.gsm
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libdata/asterisk/sounds/en/press-or-say-7.gsm
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libdata/asterisk/sounds/en/press-or-say-8.gsm
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libdata/asterisk/sounds/en/press-or-say-9.gsm
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libdata/asterisk/sounds/en/press-pound-save-changes.gsm
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libdata/asterisk/sounds/en/press-pound-to-login-star-to-hangup.gsm
|
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libdata/asterisk/sounds/en/press-pound.gsm
|
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libdata/asterisk/sounds/en/press-return.gsm
|
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libdata/asterisk/sounds/en/press-star-cancel.gsm
|
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libdata/asterisk/sounds/en/press-star.gsm
|
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libdata/asterisk/sounds/en/press-the-space-bar.gsm
|
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libdata/asterisk/sounds/en/press-tilde.gsm
|
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libdata/asterisk/sounds/en/press.gsm
|
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libdata/asterisk/sounds/en/press_pound_hash.gsm
|
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libdata/asterisk/sounds/en/press_pound_hash_key.gsm
|
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libdata/asterisk/sounds/en/pressure.gsm
|
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libdata/asterisk/sounds/en/prime_number.gsm
|
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libdata/asterisk/sounds/en/printing.gsm
|
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libdata/asterisk/sounds/en/priv-callee-options.gsm
|
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libdata/asterisk/sounds/en/priv-callpending.gsm
|
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libdata/asterisk/sounds/en/priv-introsaved.gsm
|
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libdata/asterisk/sounds/en/priv-recordintro.gsm
|
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libdata/asterisk/sounds/en/privacy-blacklisted.gsm
|
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libdata/asterisk/sounds/en/privacy-blocked.gsm
|
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libdata/asterisk/sounds/en/privacy-if-error-leave-message-or-hangup.gsm
|
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libdata/asterisk/sounds/en/privacy-if-error.gsm
|
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libdata/asterisk/sounds/en/privacy-if-sales-call-contact-in-writing.gsm
|
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libdata/asterisk/sounds/en/privacy-incorrect.gsm
|
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libdata/asterisk/sounds/en/privacy-last-caller-was.gsm
|
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libdata/asterisk/sounds/en/privacy-not.gsm
|
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libdata/asterisk/sounds/en/privacy-please-dial.gsm
|
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libdata/asterisk/sounds/en/privacy-please-stay-on-line-to-be-connected.gsm
|
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libdata/asterisk/sounds/en/privacy-prompt.gsm
|
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libdata/asterisk/sounds/en/privacy-restricted-by-req.gsm
|
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libdata/asterisk/sounds/en/privacy-screening-unidentified-calls.gsm
|
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libdata/asterisk/sounds/en/privacy-stop-calling-not-welcome.gsm
|
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libdata/asterisk/sounds/en/privacy-stop-calling-not-welcome2.gsm
|
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libdata/asterisk/sounds/en/privacy-thankyou.gsm
|
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libdata/asterisk/sounds/en/privacy-this-number-is.gsm
|
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libdata/asterisk/sounds/en/privacy-to-blacklist-last-caller.gsm
|
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libdata/asterisk/sounds/en/privacy-to-blacklist-this-number.gsm
|
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libdata/asterisk/sounds/en/privacy-to-hear-our-contact-details.gsm
|
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libdata/asterisk/sounds/en/privacy-to-whitelist-last-caller.gsm
|
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libdata/asterisk/sounds/en/privacy-to-whitelist-this-number.gsm
|
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libdata/asterisk/sounds/en/privacy-unident.gsm
|
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libdata/asterisk/sounds/en/privacy-whitelisted.gsm
|
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libdata/asterisk/sounds/en/privacy-you-are-blacklisted.gsm
|
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libdata/asterisk/sounds/en/privacy-you-are-calling-from.gsm
|
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libdata/asterisk/sounds/en/privacy-your-callerid-is.gsm
|
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libdata/asterisk/sounds/en/product.gsm
|
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libdata/asterisk/sounds/en/production.gsm
|
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libdata/asterisk/sounds/en/projects.gsm
|
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libdata/asterisk/sounds/en/prompt-not-found.gsm
|
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libdata/asterisk/sounds/en/protocol.gsm
|
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libdata/asterisk/sounds/en/public-relations.gsm
|
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libdata/asterisk/sounds/en/purchasing.gsm
|
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libdata/asterisk/sounds/en/purposes.gsm
|
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libdata/asterisk/sounds/en/q-dot-931.gsm
|
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libdata/asterisk/sounds/en/q-dot-9thirty1.gsm
|
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libdata/asterisk/sounds/en/quality-assurance.gsm
|
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libdata/asterisk/sounds/en/quality-control.gsm
|
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libdata/asterisk/sounds/en/quality_control.gsm
|
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libdata/asterisk/sounds/en/quarter.gsm
|
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libdata/asterisk/sounds/en/queue-callswaiting.gsm
|
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libdata/asterisk/sounds/en/queue-holdtime.gsm
|
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libdata/asterisk/sounds/en/queue-less-than.gsm
|
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libdata/asterisk/sounds/en/queue-minute.gsm
|
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libdata/asterisk/sounds/en/queue-minutes.gsm
|
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libdata/asterisk/sounds/en/queue-periodic-announce.gsm
|
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libdata/asterisk/sounds/en/queue-quantity1.gsm
|
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libdata/asterisk/sounds/en/queue-quantity2.gsm
|
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libdata/asterisk/sounds/en/queue-reporthold.gsm
|
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libdata/asterisk/sounds/en/queue-seconds.gsm
|
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libdata/asterisk/sounds/en/queue-thankyou.gsm
|
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libdata/asterisk/sounds/en/queue-thereare.gsm
|
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libdata/asterisk/sounds/en/queue-youarenext.gsm
|
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libdata/asterisk/sounds/en/quickly.gsm
|
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libdata/asterisk/sounds/en/quote.gsm
|
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libdata/asterisk/sounds/en/race.gsm
|
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libdata/asterisk/sounds/en/rain.gsm
|
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libdata/asterisk/sounds/en/rainfall.gsm
|
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libdata/asterisk/sounds/en/rainy.gsm
|
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libdata/asterisk/sounds/en/range.gsm
|
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libdata/asterisk/sounds/en/reassigned-new-areacode.gsm
|
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libdata/asterisk/sounds/en/rebates.gsm
|
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libdata/asterisk/sounds/en/received.gsm
|
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libdata/asterisk/sounds/en/receiving.gsm
|
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libdata/asterisk/sounds/en/reception.gsm
|
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libdata/asterisk/sounds/en/recorded.gsm
|
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libdata/asterisk/sounds/en/registrar.gsm
|
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libdata/asterisk/sounds/en/regret_not_saved.gsm
|
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libdata/asterisk/sounds/en/remote-already-in-this-mode-2.gsm
|
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libdata/asterisk/sounds/en/remote-already-in-this-mode.gsm
|
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libdata/asterisk/sounds/en/remote-base.gsm
|
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libdata/asterisk/sounds/en/removed.gsm
|
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libdata/asterisk/sounds/en/repair.gsm
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libdata/asterisk/sounds/en/repeat-only.gsm
|
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libdata/asterisk/sounds/en/repeat_pin.gsm
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libdata/asterisk/sounds/en/repeater.gsm
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libdata/asterisk/sounds/en/research-and-development.gsm
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libdata/asterisk/sounds/en/research.gsm
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libdata/asterisk/sounds/en/reservations.gsm
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libdata/asterisk/sounds/en/restarting.gsm
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libdata/asterisk/sounds/en/rhode-island.gsm
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libdata/asterisk/sounds/en/right-bracket.gsm
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libdata/asterisk/sounds/en/rising.gsm
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libdata/asterisk/sounds/en/risk-management.gsm
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libdata/asterisk/sounds/en/roaming.gsm
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libdata/asterisk/sounds/en/room-service.gsm
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libdata/asterisk/sounds/en/route-sip.gsm
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libdata/asterisk/sounds/en/rqsted-wakeup-for.gsm
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libdata/asterisk/sounds/en/sacramento.gsm
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libdata/asterisk/sounds/en/said_hold_on.gsm
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libdata/asterisk/sounds/en/saint-louis.gsm
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libdata/asterisk/sounds/en/sales-floor.gsm
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libdata/asterisk/sounds/en/sales.gsm
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libdata/asterisk/sounds/en/saleshell.gsm
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libdata/asterisk/sounds/en/san-antonio.gsm
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libdata/asterisk/sounds/en/san-diego.gsm
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libdata/asterisk/sounds/en/san-francisco.gsm
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libdata/asterisk/sounds/en/san-jose.gsm
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libdata/asterisk/sounds/en/save-announce-press.gsm
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libdata/asterisk/sounds/en/say-temp-msg-prs-pound.gsm
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libdata/asterisk/sounds/en/says-thats-stupid.gsm
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libdata/asterisk/sounds/en/scattered.gsm
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libdata/asterisk/sounds/en/sciences.gsm
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libdata/asterisk/sounds/en/screen-callee-options.gsm
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libdata/asterisk/sounds/en/seattle.gsm
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libdata/asterisk/sounds/en/second.gsm
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libdata/asterisk/sounds/en/seconds.gsm
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libdata/asterisk/sounds/en/secretary.gsm
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libdata/asterisk/sounds/en/security.gsm
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libdata/asterisk/sounds/en/self-destruct-in.gsm
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libdata/asterisk/sounds/en/self-destruct.gsm
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libdata/asterisk/sounds/en/sendhelp.gsm
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libdata/asterisk/sounds/en/server.gsm
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libdata/asterisk/sounds/en/service-not-implemented.gsm
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libdata/asterisk/sounds/en/service.gsm
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libdata/asterisk/sounds/en/seventeenth.gsm
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libdata/asterisk/sounds/en/seventh.gsm
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libdata/asterisk/sounds/en/seventieth.gsm
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libdata/asterisk/sounds/en/severe.gsm
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libdata/asterisk/sounds/en/shall-i-try-again.gsm
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libdata/asterisk/sounds/en/shiny-brass-lamp.gsm
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libdata/asterisk/sounds/en/shipping.gsm
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libdata/asterisk/sounds/en/shop.gsm
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libdata/asterisk/sounds/en/show-office.gsm
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libdata/asterisk/sounds/en/sighted.gsm
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libdata/asterisk/sounds/en/silence/1.gsm
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libdata/asterisk/sounds/en/silence/10.gsm
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libdata/asterisk/sounds/en/silence/2.gsm
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libdata/asterisk/sounds/en/silence/3.gsm
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libdata/asterisk/sounds/en/silence/4.gsm
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libdata/asterisk/sounds/en/silence/5.gsm
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libdata/asterisk/sounds/en/silence/6.gsm
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libdata/asterisk/sounds/en/silence/7.gsm
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libdata/asterisk/sounds/en/silence/8.gsm
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libdata/asterisk/sounds/en/silence/9.gsm
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libdata/asterisk/sounds/en/simplex.gsm
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libdata/asterisk/sounds/en/simul-call-limit-reached.gsm
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libdata/asterisk/sounds/en/sixteenth.gsm
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libdata/asterisk/sounds/en/sixth.gsm
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libdata/asterisk/sounds/en/sixtieth.gsm
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libdata/asterisk/sounds/en/sleet.gsm
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libdata/asterisk/sounds/en/sleeting.gsm
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libdata/asterisk/sounds/en/slow.gsm
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libdata/asterisk/sounds/en/slowly.gsm
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libdata/asterisk/sounds/en/snow.gsm
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libdata/asterisk/sounds/en/snowing.gsm
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libdata/asterisk/sounds/en/snowy.gsm
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libdata/asterisk/sounds/en/software.gsm
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libdata/asterisk/sounds/en/someone-you-trust1.gsm
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libdata/asterisk/sounds/en/someone-you-trust2.gsm
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libdata/asterisk/sounds/en/someone-you-trust3.gsm
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libdata/asterisk/sounds/en/something-terribly-wrong.gsm
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libdata/asterisk/sounds/en/sorry-cant-let-you-do-that.gsm
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libdata/asterisk/sounds/en/sorry-cant-let-you-do-that2.gsm
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libdata/asterisk/sounds/en/sorry-cant-let-you-do-that3.gsm
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libdata/asterisk/sounds/en/sorry-mailbox-full.gsm
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libdata/asterisk/sounds/en/sorry-youre-having-problems.gsm
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libdata/asterisk/sounds/en/sorry.gsm
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libdata/asterisk/sounds/en/sorry2.gsm
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libdata/asterisk/sounds/en/sorry_caller_number.gsm
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libdata/asterisk/sounds/en/sorry_didnt_get.gsm
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libdata/asterisk/sounds/en/sorry_didnt_quite_get.gsm
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libdata/asterisk/sounds/en/sorry_login_incorrect.gsm
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libdata/asterisk/sounds/en/sorry_missed.gsm
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libdata/asterisk/sounds/en/sorry_no_messages.gsm
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libdata/asterisk/sounds/en/sorrydave.gsm
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libdata/asterisk/sounds/en/south-carolina.gsm
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libdata/asterisk/sounds/en/south-dakota.gsm
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libdata/asterisk/sounds/en/south.gsm
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libdata/asterisk/sounds/en/southerly.gsm
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libdata/asterisk/sounds/en/spam.gsm
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libdata/asterisk/sounds/en/spam2.gsm
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libdata/asterisk/sounds/en/speak-louder-into-phone.gsm
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libdata/asterisk/sounds/en/speak-louder.gsm
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libdata/asterisk/sounds/en/speak-to-the-operator.gsm
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libdata/asterisk/sounds/en/speed-dial-empty.gsm
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libdata/asterisk/sounds/en/speed-dial.gsm
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libdata/asterisk/sounds/en/speed.gsm
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libdata/asterisk/sounds/en/splat.gsm
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libdata/asterisk/sounds/en/spy-agent.gsm
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libdata/asterisk/sounds/en/spy-console.gsm
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libdata/asterisk/sounds/en/spy-dahdi.gsm
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libdata/asterisk/sounds/en/spy-h323.gsm
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libdata/asterisk/sounds/en/spy-iax.gsm
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libdata/asterisk/sounds/en/spy-iax2.gsm
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libdata/asterisk/sounds/en/spy-jingle.gsm
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libdata/asterisk/sounds/en/spy-local.gsm
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libdata/asterisk/sounds/en/spy-mgcp.gsm
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libdata/asterisk/sounds/en/spy-misdn.gsm
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libdata/asterisk/sounds/en/spy-mobile.gsm
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libdata/asterisk/sounds/en/spy-nbs.gsm
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libdata/asterisk/sounds/en/spy-sip.gsm
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libdata/asterisk/sounds/en/spy-skinny.gsm
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libdata/asterisk/sounds/en/spy-unistim.gsm
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libdata/asterisk/sounds/en/spy-usbradio.gsm
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libdata/asterisk/sounds/en/spy-zap.gsm
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libdata/asterisk/sounds/en/ss-noservice.gsm
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libdata/asterisk/sounds/en/ssh.gsm
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libdata/asterisk/sounds/en/staff.gsm
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libdata/asterisk/sounds/en/staffing.gsm
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libdata/asterisk/sounds/en/standard.gsm
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libdata/asterisk/sounds/en/star-for-menu-again.gsm
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libdata/asterisk/sounds/en/starting-with-either.gsm
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libdata/asterisk/sounds/en/station.gsm
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libdata/asterisk/sounds/en/status.gsm
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libdata/asterisk/sounds/en/step-in-stream.gsm
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libdata/asterisk/sounds/en/sterling.gsm
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libdata/asterisk/sounds/en/still_on_phone.gsm
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libdata/asterisk/sounds/en/store-accounting.gsm
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libdata/asterisk/sounds/en/storm.gsm
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libdata/asterisk/sounds/en/sun.gsm
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libdata/asterisk/sounds/en/sunny.gsm
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libdata/asterisk/sounds/en/support.gsm
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libdata/asterisk/sounds/en/sure_help.gsm
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libdata/asterisk/sounds/en/swap.gsm
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libdata/asterisk/sounds/en/system-crashed.gsm
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libdata/asterisk/sounds/en/system-status-msg.gsm
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libdata/asterisk/sounds/en/system.gsm
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libdata/asterisk/sounds/en/systems.gsm
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libdata/asterisk/sounds/en/talkin_me.gsm
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libdata/asterisk/sounds/en/talking-to-myself.gsm
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libdata/asterisk/sounds/en/target-attendant.gsm
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libdata/asterisk/sounds/en/tcp.gsm
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libdata/asterisk/sounds/en/technical-support.gsm
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libdata/asterisk/sounds/en/telemarketercalling.gsm
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libdata/asterisk/sounds/en/telephone-in-your-pocket.gsm
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libdata/asterisk/sounds/en/telephone-in-your-pocket2.gsm
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libdata/asterisk/sounds/en/telephone-number.gsm
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libdata/asterisk/sounds/en/telesales.gsm
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libdata/asterisk/sounds/en/teletubbie-murder.gsm
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libdata/asterisk/sounds/en/telnet.gsm
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libdata/asterisk/sounds/en/temp-disconnected.gsm
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libdata/asterisk/sounds/en/temperature.gsm
|
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libdata/asterisk/sounds/en/tennessee.gsm
|
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libdata/asterisk/sounds/en/tenth.gsm
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libdata/asterisk/sounds/en/terabits.gsm
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libdata/asterisk/sounds/en/terabytes.gsm
|
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libdata/asterisk/sounds/en/terminated.gsm
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libdata/asterisk/sounds/en/terminating.gsm
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libdata/asterisk/sounds/en/test-tones-follow.gsm
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libdata/asterisk/sounds/en/texas.gsm
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libdata/asterisk/sounds/en/thank-you-cooperation.gsm
|
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libdata/asterisk/sounds/en/thank-you-for-calling.gsm
|
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libdata/asterisk/sounds/en/thank_you_calling.gsm
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libdata/asterisk/sounds/en/thanks-annoyance.gsm
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libdata/asterisk/sounds/en/thanks-for-calling-today.gsm
|
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libdata/asterisk/sounds/en/thanks-for-using.gsm
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libdata/asterisk/sounds/en/that-is-not-rec-phn-num.gsm
|
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libdata/asterisk/sounds/en/that-number.gsm
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libdata/asterisk/sounds/en/that-tickles.gsm
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libdata/asterisk/sounds/en/that-you-require.gsm
|
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libdata/asterisk/sounds/en/the-mailbox.gsm
|
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libdata/asterisk/sounds/en/the-monkeys-twice.gsm
|
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libdata/asterisk/sounds/en/the-new-number-is.gsm
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libdata/asterisk/sounds/en/the-next.gsm
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libdata/asterisk/sounds/en/the-num-i-have-is.gsm
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libdata/asterisk/sounds/en/the-number-u-dialed.gsm
|
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libdata/asterisk/sounds/en/the-party-you-are-calling.gsm
|
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libdata/asterisk/sounds/en/the-weather-at.gsm
|
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libdata/asterisk/sounds/en/then-press-pound.gsm
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libdata/asterisk/sounds/en/there-are.gsm
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libdata/asterisk/sounds/en/there-is-no-customer-support.gsm
|
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libdata/asterisk/sounds/en/these-are-currently.gsm
|
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libdata/asterisk/sounds/en/third.gsm
|
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libdata/asterisk/sounds/en/thirteenth.gsm
|
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libdata/asterisk/sounds/en/thirtieth.gsm
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libdata/asterisk/sounds/en/this-call-may-be-monitored-or-recorded.gsm
|
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libdata/asterisk/sounds/en/this-call-may-be.gsm
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libdata/asterisk/sounds/en/this-call-will-cost.gsm
|
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libdata/asterisk/sounds/en/this-call-will-end-in.gsm
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libdata/asterisk/sounds/en/this-is-the-voice-mail-system.gsm
|
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libdata/asterisk/sounds/en/this-is-yr-wakeup-call.gsm
|
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libdata/asterisk/sounds/en/this.gsm
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libdata/asterisk/sounds/en/thnk-u-for-patience.gsm
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libdata/asterisk/sounds/en/thousandth.gsm
|
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libdata/asterisk/sounds/en/through.gsm
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libdata/asterisk/sounds/en/thunderstorm.gsm
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libdata/asterisk/sounds/en/tide.gsm
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libdata/asterisk/sounds/en/time.gsm
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libdata/asterisk/sounds/en/times.gsm
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libdata/asterisk/sounds/en/timewarp.gsm
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libdata/asterisk/sounds/en/to-accept-recording.gsm
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libdata/asterisk/sounds/en/to-be-called-back.gsm
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libdata/asterisk/sounds/en/to-blklist-last-caller.gsm
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libdata/asterisk/sounds/en/to-blklist-last-num.gsm
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libdata/asterisk/sounds/en/to-call-num-press.gsm
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libdata/asterisk/sounds/en/to-call-prson-w-sent-msg.gsm
|
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libdata/asterisk/sounds/en/to-call-this-number.gsm
|
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libdata/asterisk/sounds/en/to-cancel-this-msg.gsm
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libdata/asterisk/sounds/en/to-cancel-wakeup.gsm
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libdata/asterisk/sounds/en/to-change-exp-date.gsm
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libdata/asterisk/sounds/en/to-change-your-pin-number.gsm
|
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libdata/asterisk/sounds/en/to-collect-voicemail.gsm
|
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libdata/asterisk/sounds/en/to-compose-a-message.gsm
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libdata/asterisk/sounds/en/to-confirm-wakeup.gsm
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libdata/asterisk/sounds/en/to-dial-by-name-press.gsm
|
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libdata/asterisk/sounds/en/to-dial-by-name.gsm
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libdata/asterisk/sounds/en/to-enter-a-diff-number.gsm
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libdata/asterisk/sounds/en/to-enter-a-number.gsm
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libdata/asterisk/sounds/en/to-erase-yr-temp-grtg.gsm
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libdata/asterisk/sounds/en/to-extension.gsm
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libdata/asterisk/sounds/en/to-hang-up-2.gsm
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libdata/asterisk/sounds/en/to-hang-up.gsm
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libdata/asterisk/sounds/en/to-hear-callerid.gsm
|
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libdata/asterisk/sounds/en/to-hear-menu-again.gsm
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libdata/asterisk/sounds/en/to-hear-msg-again.gsm
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libdata/asterisk/sounds/en/to-hear-msg-envelope.gsm
|
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libdata/asterisk/sounds/en/to-hear-net-status.gsm
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libdata/asterisk/sounds/en/to-hear-weather-status.gsm
|
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libdata/asterisk/sounds/en/to-hear-your-account-balance.gsm
|
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libdata/asterisk/sounds/en/to-join-a-meeting.gsm
|
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libdata/asterisk/sounds/en/to-leave-message-for.gsm
|
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libdata/asterisk/sounds/en/to-listen-to-it.gsm
|
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libdata/asterisk/sounds/en/to-log-in-to-voice-mail.gsm
|
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libdata/asterisk/sounds/en/to-place-outgoing-call.gsm
|
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libdata/asterisk/sounds/en/to-reach-first-rep.gsm
|
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libdata/asterisk/sounds/en/to-reach-operator.gsm
|
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libdata/asterisk/sounds/en/to-rec-yr-temp-grtg.gsm
|
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libdata/asterisk/sounds/en/to-record-call.gsm
|
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libdata/asterisk/sounds/en/to-redial-the-last-number-you-called.gsm
|
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libdata/asterisk/sounds/en/to-report-emergency.gsm
|
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libdata/asterisk/sounds/en/to-report-system-network-down.gsm
|
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libdata/asterisk/sounds/en/to-rerecord-announce.gsm
|
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libdata/asterisk/sounds/en/to-rerecord-it.gsm
|
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libdata/asterisk/sounds/en/to-rerecord-yr-message.gsm
|
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libdata/asterisk/sounds/en/to-rmv-num-blklist.gsm
|
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libdata/asterisk/sounds/en/to-rqst-wakeup-call.gsm
|
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libdata/asterisk/sounds/en/to-send-a-reply.gsm
|
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libdata/asterisk/sounds/en/to-snooze-for.gsm
|
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libdata/asterisk/sounds/en/to-use-def-attendant.gsm
|
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libdata/asterisk/sounds/en/tomorrow-night.gsm
|
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libdata/asterisk/sounds/en/tone_time.gsm
|
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libdata/asterisk/sounds/en/tones-that-follow-are-for-the-deaf.gsm
|
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libdata/asterisk/sounds/en/tonight.gsm
|
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libdata/asterisk/sounds/en/too-low.gsm
|
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libdata/asterisk/sounds/en/tornado.gsm
|
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libdata/asterisk/sounds/en/touchtone1.gsm
|
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libdata/asterisk/sounds/en/touchtone2.gsm
|
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libdata/asterisk/sounds/en/touchtone3.gsm
|
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libdata/asterisk/sounds/en/towards.gsm
|
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libdata/asterisk/sounds/en/trading-desk.gsm
|
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libdata/asterisk/sounds/en/traffic.gsm
|
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libdata/asterisk/sounds/en/training.gsm
|
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libdata/asterisk/sounds/en/transfer.gsm
|
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libdata/asterisk/sounds/en/transfer_to_agent.gsm
|
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libdata/asterisk/sounds/en/transportation.gsm
|
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libdata/asterisk/sounds/en/travel.gsm
|
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libdata/asterisk/sounds/en/treasury.gsm
|
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libdata/asterisk/sounds/en/tt-allbusy.gsm
|
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libdata/asterisk/sounds/en/tt-codezone.gsm
|
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libdata/asterisk/sounds/en/tt-codezone_MIXDOWN.gsm
|
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libdata/asterisk/sounds/en/tt-hangup.gsm
|
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libdata/asterisk/sounds/en/tt-monkeys.gsm
|
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libdata/asterisk/sounds/en/tt-monkeysintro.gsm
|
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libdata/asterisk/sounds/en/tt-monty-knights.gsm
|
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libdata/asterisk/sounds/en/tt-somethingwrong.gsm
|
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libdata/asterisk/sounds/en/tt-weasels.gsm
|
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libdata/asterisk/sounds/en/tucson.gsm
|
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libdata/asterisk/sounds/en/tulsa.gsm
|
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libdata/asterisk/sounds/en/turn-off-recording.gsm
|
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libdata/asterisk/sounds/en/turning-to.gsm
|
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libdata/asterisk/sounds/en/twelveth.gsm
|
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libdata/asterisk/sounds/en/twentieth.gsm
|
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libdata/asterisk/sounds/en/twisty-maze.gsm
|
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libdata/asterisk/sounds/en/tx-has-been-approved.gsm
|
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libdata/asterisk/sounds/en/typhoon.gsm
|
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libdata/asterisk/sounds/en/udp.gsm
|
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libdata/asterisk/sounds/en/uh-oh1.gsm
|
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libdata/asterisk/sounds/en/uh-oh2.gsm
|
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libdata/asterisk/sounds/en/uk.gsm
|
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libdata/asterisk/sounds/en/unavailable.gsm
|
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libdata/asterisk/sounds/en/unconditional.gsm
|
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libdata/asterisk/sounds/en/unicorn_blood.gsm
|
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libdata/asterisk/sounds/en/unicorn_blood_MIXDOWN.gsm
|
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libdata/asterisk/sounds/en/unidentified-no-callback.gsm
|
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libdata/asterisk/sounds/en/units.gsm
|
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libdata/asterisk/sounds/en/unix.gsm
|
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libdata/asterisk/sounds/en/unwelcomecall.gsm
|
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libdata/asterisk/sounds/en/up.gsm
|
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libdata/asterisk/sounds/en/uppercase.gsm
|
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libdata/asterisk/sounds/en/uptime.gsm
|
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libdata/asterisk/sounds/en/user.gsm
|
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libdata/asterisk/sounds/en/users.gsm
|
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libdata/asterisk/sounds/en/utah.gsm
|
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libdata/asterisk/sounds/en/variable.gsm
|
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libdata/asterisk/sounds/en/vermont.gsm
|
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libdata/asterisk/sounds/en/virginia-beach.gsm
|
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libdata/asterisk/sounds/en/virginia.gsm
|
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libdata/asterisk/sounds/en/visibility.gsm
|
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libdata/asterisk/sounds/en/visit-asterisk-website.gsm
|
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libdata/asterisk/sounds/en/vm-Cust1.gsm
|
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libdata/asterisk/sounds/en/vm-Cust2.gsm
|
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libdata/asterisk/sounds/en/vm-Cust3.gsm
|
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libdata/asterisk/sounds/en/vm-Cust4.gsm
|
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libdata/asterisk/sounds/en/vm-Cust5.gsm
|
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libdata/asterisk/sounds/en/vm-Family.gsm
|
|
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|
libdata/asterisk/sounds/en/vm-Friends.gsm
|
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|
libdata/asterisk/sounds/en/vm-INBOX.gsm
|
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|
libdata/asterisk/sounds/en/vm-Old.gsm
|
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|
libdata/asterisk/sounds/en/vm-Urgent.gsm
|
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libdata/asterisk/sounds/en/vm-Work.gsm
|
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libdata/asterisk/sounds/en/vm-advopts.gsm
|
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libdata/asterisk/sounds/en/vm-and.gsm
|
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libdata/asterisk/sounds/en/vm-calldiffnum.gsm
|
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libdata/asterisk/sounds/en/vm-changeto.gsm
|
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libdata/asterisk/sounds/en/vm-delete.gsm
|
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libdata/asterisk/sounds/en/vm-deleted.gsm
|
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libdata/asterisk/sounds/en/vm-dialout.gsm
|
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|
libdata/asterisk/sounds/en/vm-duration.gsm
|
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|
libdata/asterisk/sounds/en/vm-enter-num-to-call.gsm
|
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|
libdata/asterisk/sounds/en/vm-extension.gsm
|
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libdata/asterisk/sounds/en/vm-first.gsm
|
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libdata/asterisk/sounds/en/vm-for.gsm
|
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|
libdata/asterisk/sounds/en/vm-forward-multiple.gsm
|
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|
libdata/asterisk/sounds/en/vm-forward.gsm
|
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|
libdata/asterisk/sounds/en/vm-forwardoptions.gsm
|
|
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|
libdata/asterisk/sounds/en/vm-from-extension.gsm
|
|
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|
libdata/asterisk/sounds/en/vm-from-phonenumber.gsm
|
|
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|
libdata/asterisk/sounds/en/vm-from.gsm
|
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|
libdata/asterisk/sounds/en/vm-goodbye.gsm
|
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|
libdata/asterisk/sounds/en/vm-helpexit.gsm
|
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|
libdata/asterisk/sounds/en/vm-incorrect-mailbox.gsm
|
|
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|
libdata/asterisk/sounds/en/vm-incorrect.gsm
|
|
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|
libdata/asterisk/sounds/en/vm-instructions.gsm
|
|
|
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|
libdata/asterisk/sounds/en/vm-intro.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-invalid-password.gsm
|
|
|
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|
libdata/asterisk/sounds/en/vm-invalidpassword.gsm
|
|
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|
|
libdata/asterisk/sounds/en/vm-isonphone.gsm
|
|
|
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|
libdata/asterisk/sounds/en/vm-isunavail.gsm
|
|
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|
libdata/asterisk/sounds/en/vm-last.gsm
|
|
|
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|
libdata/asterisk/sounds/en/vm-leavemsg.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-login.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-mailboxfull.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-marked-nonurgent.gsm
|
|
|
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|
libdata/asterisk/sounds/en/vm-marked-urgent.gsm
|
|
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|
libdata/asterisk/sounds/en/vm-message.gsm
|
|
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|
libdata/asterisk/sounds/en/vm-messages.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-minutes.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-mismatch.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-msgforwarded.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-msginstruct.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-msgsaved.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-newpassword.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-newuser.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-next.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-no.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-nobodyavail.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-nobox.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-nomore.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-nonumber.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-num-i-have.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-onefor-full.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-onefor.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-options.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-opts-full.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-opts.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-passchanged.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-password.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-pls-try-again.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-press.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-prev.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-reachoper.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-rec-busy.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-rec-name.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-rec-temp.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-rec-unv.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-received.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-record-prepend.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-reenterpassword.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-repeat.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-review-nonurgent.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-review-urgent.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-review.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-saved.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-savedto.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-savefolder.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-savemessage.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-saveoper.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-sorry.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-star-cancel.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-starmain.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-tempgreetactive.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-tempgreeting.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-tempgreeting2.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-tempremoved.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-then-pound.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-theperson.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-tmpexists.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-tocallback.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-tocallnum.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-tocancel.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-tocancelmsg.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-toenternumber.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-toforward.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-tohearenv.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-tomakecall.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-tooshort.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-toreply.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-torerecord.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-undelete.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-undeleted.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-unknown-caller.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-whichbox.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/vm-youhave.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/voice-mail-system.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/wait-moment.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/wait-offensive-sounds.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/wakeup-call-cancelled.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/wakeup-call.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/wakeup-daily.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/wakeup-for-daily.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/wakeup-for-one-time.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/wakeup-onetime.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/walks-into-bar-mail.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/warning.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/was-last-seen.gsm
|
|
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|
libdata/asterisk/sounds/en/was.gsm
|
|
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|
libdata/asterisk/sounds/en/washington-dc.gsm
|
|
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|
|
libdata/asterisk/sounds/en/washington.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/watch.gsm
|
|
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|
|
libdata/asterisk/sounds/en/watson.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/we-apologize.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/we-dont-have-tech-support.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/weasels-eaten-phonesys.gsm
|
|
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|
|
libdata/asterisk/sounds/en/weather-station.gsm
|
|
|
|
|
libdata/asterisk/sounds/en/weather.gsm
|
|
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|
libdata/asterisk/sounds/en/web.gsm
|
|
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|
libdata/asterisk/sounds/en/weeks.gsm
|
|
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|
libdata/asterisk/sounds/en/welcome.gsm
|
|
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|
libdata/asterisk/sounds/en/were-sorry.gsm
|
|
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|
libdata/asterisk/sounds/en/west-virginia.gsm
|
|
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|
libdata/asterisk/sounds/en/west.gsm
|
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|
libdata/asterisk/sounds/en/westerly.gsm
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libdata/asterisk/sounds/en/what-are-you-wearing.gsm
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libdata/asterisk/sounds/en/what-time-it-is.gsm
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libdata/asterisk/sounds/en/what-time-it-is2.gsm
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libdata/asterisk/sounds/en/when-dialing-this-number.gsm
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libdata/asterisk/sounds/en/who-would-you-like-to-call.gsm
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libdata/asterisk/sounds/en/whoareyou.gsm
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libdata/asterisk/sounds/en/why-no-answer-mystery.gsm
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libdata/asterisk/sounds/en/wichita.gsm
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libdata/asterisk/sounds/en/will-apply.gsm
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libdata/asterisk/sounds/en/will-expire.gsm
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libdata/asterisk/sounds/en/will-not-expire.gsm
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libdata/asterisk/sounds/en/will-reflect-charge-of.gsm
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libdata/asterisk/sounds/en/wind.gsm
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libdata/asterisk/sounds/en/windows.gsm
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libdata/asterisk/sounds/en/windy.gsm
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libdata/asterisk/sounds/en/wisconsin.gsm
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libdata/asterisk/sounds/en/wish-to-continue.gsm
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libdata/asterisk/sounds/en/wish_command_totally.gsm
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libdata/asterisk/sounds/en/with.gsm
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libdata/asterisk/sounds/en/within.gsm
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libdata/asterisk/sounds/en/woo_hoo_call_first.gsm
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libdata/asterisk/sounds/en/work.gsm
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libdata/asterisk/sounds/en/wrong-try-again-smarty.gsm
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libdata/asterisk/sounds/en/wtng-to-spk-w-rep.gsm
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libdata/asterisk/sounds/en/www-switchboard-com.gsm
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libdata/asterisk/sounds/en/wx/around.gsm
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libdata/asterisk/sounds/en/wx/barometer.gsm
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libdata/asterisk/sounds/en/wx/ceiling.gsm
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libdata/asterisk/sounds/en/wx/dew-point.gsm
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libdata/asterisk/sounds/en/wx/falling.gsm
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libdata/asterisk/sounds/en/wx/feet.gsm
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libdata/asterisk/sounds/en/wx/foot.gsm
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libdata/asterisk/sounds/en/wx/gust.gsm
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libdata/asterisk/sounds/en/wx/gusting-to.gsm
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libdata/asterisk/sounds/en/wx/gusts.gsm
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libdata/asterisk/sounds/en/wx/gusty.gsm
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libdata/asterisk/sounds/en/wx/heat-index.gsm
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libdata/asterisk/sounds/en/wx/humidity.gsm
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libdata/asterisk/sounds/en/wx/inch.gsm
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libdata/asterisk/sounds/en/wx/inches.gsm
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libdata/asterisk/sounds/en/wx/kilometer.gsm
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libdata/asterisk/sounds/en/wx/large.gsm
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libdata/asterisk/sounds/en/wx/meter.gsm
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libdata/asterisk/sounds/en/wx/mist.gsm
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libdata/asterisk/sounds/en/wx/northeast.gsm
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libdata/asterisk/sounds/en/wx/northwest.gsm
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libdata/asterisk/sounds/en/wx/percent.gsm
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libdata/asterisk/sounds/en/wx/point.gsm
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libdata/asterisk/sounds/en/wx/rising.gsm
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libdata/asterisk/sounds/en/wx/southeast.gsm
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libdata/asterisk/sounds/en/wx/southwest.gsm
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libdata/asterisk/sounds/en/wx/steady.gsm
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libdata/asterisk/sounds/en/wx/temperature.gsm
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libdata/asterisk/sounds/en/wx/wind-chill.gsm
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libdata/asterisk/sounds/en/wx/winds.gsm
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libdata/asterisk/sounds/en/wyoming.gsm
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libdata/asterisk/sounds/en/yeah.gsm
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libdata/asterisk/sounds/en/year.gsm
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libdata/asterisk/sounds/en/years.gsm
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libdata/asterisk/sounds/en/yes-dear.gsm
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libdata/asterisk/sounds/en/yes-dear2.gsm
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libdata/asterisk/sounds/en/you-are-caller-num.gsm
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libdata/asterisk/sounds/en/you-are-curr-call-num.gsm
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libdata/asterisk/sounds/en/you-can-press.gsm
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libdata/asterisk/sounds/en/you-dialed-wrong-number.gsm
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libdata/asterisk/sounds/en/you-entered.gsm
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libdata/asterisk/sounds/en/you-have-dialed.gsm
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libdata/asterisk/sounds/en/you-have-reached-a-test-number.gsm
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libdata/asterisk/sounds/en/you-have-these-options.gsm
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libdata/asterisk/sounds/en/you-must-first-dial.gsm
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libdata/asterisk/sounds/en/you-seem-impatient.gsm
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libdata/asterisk/sounds/en/you-sound-cute.gsm
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libdata/asterisk/sounds/en/you-wish-to-join.gsm
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libdata/asterisk/sounds/en/you_say_yes.gsm
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libdata/asterisk/sounds/en/your-account.gsm
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libdata/asterisk/sounds/en/your-msg-has-been-saved.gsm
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libdata/asterisk/sounds/en/your-msg-is-too-short.gsm
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libdata/asterisk/sounds/en/your-notifications.gsm
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libdata/asterisk/sounds/en/your-req-notification.gsm
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libdata/asterisk/sounds/en/your-temp-greeting.gsm
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libdata/asterisk/sounds/en/your.gsm
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libdata/asterisk/sounds/en/yourcallisimportant.gsm
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libdata/asterisk/sounds/en/yourcallisimpotent.gsm
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libdata/asterisk/sounds/en/zip-code.gsm
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libdata/asterisk/sounds/en/zombies.gsm
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libdata/asterisk/static-http/ajamdemo.html
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libdata/asterisk/static-http/appdocsxml.xslt
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libdata/asterisk/static-http/astman.css
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libdata/asterisk/static-http/astman.js
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libdata/asterisk/static-http/core-en_US.xml
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libdata/asterisk/static-http/mantest.html
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libdata/asterisk/static-http/prototype.js
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libexec/agi-bin/agi-test.agi
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libexec/agi-bin/eagi-sphinx-test
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libexec/agi-bin/eagi-test
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libexec/agi-bin/jukebox.agi
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${PLIST.webvmail}libexec/cgi-bin/vmail
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man/man8/astdb2bdb.8
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man/man8/astdb2sqlite3.8
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man/man8/asterisk.8
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man/man8/astgenkey.8
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man/man8/autosupport.8
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man/man8/safe_asterisk.8
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sbin/astcanary
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sbin/astdb2bdb
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sbin/astdb2sqlite3
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sbin/asterisk
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sbin/astgenkey
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sbin/astversion
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sbin/autosupport
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sbin/rasterisk
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sbin/safe_asterisk
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share/doc/asterisk/Asterisk-13-Reference.pdf
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share/doc/asterisk/Asterisk-Admin-Guide.pdf
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGI-Commands_29394277.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_answer_29394808.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_asyncagi-break_29394809.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_channel-status_29394810.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_control-stream-file_29394811.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_database-del_29394812.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_database-deltree_29394813.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_database-get_29394814.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_database-put_29394815.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_exec_29394816.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_get-data_29394817.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_get-full-variable_29394818.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_get-option_29394819.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_get-variable_29394820.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_gosub_29394571.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_hangup_29394821.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_noop_29394822.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_receive-char_29394823.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_receive-text_29394824.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_record-file_29394825.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_say-alpha_29394826.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_say-date_29394830.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_say-datetime_29394832.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_say-digits_29394827.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_say-number_29394828.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_say-phonetic_29394829.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_say-time_29394831.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_send-image_29394833.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_send-text_29394834.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_set-autohangup_29394835.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_set-callerid_29394836.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_set-context_29394837.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_set-extension_29394838.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_set-music_29394839.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_set-priority_29394840.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_set-variable_29394841.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_speech-activate-grammar_29394851.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_speech-create_29394846.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_speech-deactivate-grammar_29394852.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_speech-destroy_29394848.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_speech-load-grammar_29394849.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_speech-recognize_29394853.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_speech-set_29394847.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_speech-unload-grammar_29394850.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_stream-file_29394842.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_tdd-mode_29394843.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_verbose_29394844.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AGICommand_wait-for-digit_29394845.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AMI-Actions_29394279.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-AMI-Events_29394281.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-ARI_29394283.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_ADSIProg_29394514.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_AELSub_29394369.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_AGI_29394854.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_AMD_29394497.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_AddQueueMember_29394418.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_AgentLogin_29394501.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_AgentRequest_29394502.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_AlarmReceiver_29394513.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_Answer_29394698.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_Authenticate_29394377.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_BackGround_29394699.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_BackgroundDetect_29394399.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_BridgeWait_29394444.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_Bridge_29394782.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_Busy_29394700.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_CELGenUserEvent_29394382.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_CallCompletionCancel_29394731.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_CallCompletionRequest_29394730.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_ChanIsAvail_29394558.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_ChanSpy_29394507.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_ChangeMonitor_29394910.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_ChannelRedirect_29394512.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_ClearHash_29394607.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_ConfBridge_29394402.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_Congestion_29394701.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_ContinueWhile_29394381.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_ControlPlayback_29394520.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_DAHDIAcceptR2Call_29394351.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_DAHDIRAS_29394557.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_DAHDIScan_29394509.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_DAHDISendCallreroutingFacility_29394350.html
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share/doc/asterisk/Asterisk-Admin-Guide/Asterisk-13-Application_DAHDISendKeypadFacility_29394349.html
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update to Asterisk 13.19.0 -- this contains both security fixes
and general bug fixes, fixes AST-2017-005, AST-2017-006, AST-2017-007,
AST-2017-008, AST-2017-009, AST-2017-10, AST-2017-11, AST-2017-12,
AST-2017-13, and AST-2017-14 (note that a number of these only
pertain to PJSIP which isn't used in pkgsrc)
----- 13.19.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.19.0.
The release of Asterisk 13.19.0 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:
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-27413 - Add cache_media_frames debugging option.
(Reported by Richard Mudgett)
* ASTERISK-27206 - res_pjsip: No mechanism exists to limit
endpoint identification to IP only
(Reported by Ben Merrills)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27531 - Compiler optimizations can break module load
sequence.
(Reported by abelbeck)
* ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
Contact crashes asterisk
(Reported by Ross Beer)
* ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
read()
(Reported by Abhay Gupta)
* ASTERISK-25079 - AMI bridge of channels results in MOH not
destroyed and robotic audio on one channel
(Reported by Zane Conkle)
* ASTERISK-27490 - chan_console: 'set active' fails to work
(Reported by Tzafrir Cohen)
* ASTERISK-24756 - ConfBridge sound_muted does not work from
CLI or AMI
(Reported by Thomas Frederiksen)
* ASTERISK-25649 - Transfer application does not work with
Local channels - documentation misleading
(Reported by Ivan Ullmann)
* ASTERISK-25869 - chan_sip: "rejected because extension not
found" should be logged as a security event
(Reported by Brian J. Murrell)
* ASTERISK-27440 - Strictrtp has issues to qualify video rtp
streams
(Reported by Wim De Vlaminck)
* ASTERISK-24329 - Music On Hold announcement cuts intro of
music the first time it is played
(Reported by Thomas Frederiksen)
* ASTERISK-19657 - Coverity Report: Fix issues for error type
CHAR_IO
(Reported by Matt Jordan)
* ASTERISK-27175 - iax.conf demo peer is invalid
(Reported by Tzafrir Cohen)
* ASTERISK-27430 - README refers to security documents that do
not exist.
(Reported by Corey Farrell)
* ASTERISK-20281 - "core set verbose" behaves strangely, can't
alias it, cli.conf example broken
(Reported by Tim Ringenbach at Asteria Solutions Group)
* 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-27408 - Identify causes and fix
pjsip/resolver/srv/failover/in_dialog/transport_tcp
(Reported by Corey Farrell)
* ASTERISK-18411 - Queue members with hints for state_interface
get stuck in "In Use" state.
(Reported by Steven T. Wheeler)
* ASTERISK-26131 - chan_sip: Crash Asterisk (in
sip_request_call at chan_sip.c) by making a call to a single
character in a dot pattern match
(Reported by Dwayne Hubbard)
* ASTERISK-27475 - codec_opus requires libcurl
(Reported by Samuel For)
* ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
not applied on reload
(Reported by John Bigelow)
* ASTERISK-27465 - CLI Completion Not Working
(Reported by Ross Beer)
* ASTERISK-27460 - CDR: Deadlock using AMI Originate with
Variable CDR(amaflags)=...
(Reported by Richard Mudgett)
* ASTERISK-27453 - RTP: Blind transfer direct media scenario
results in one way audio.
(Reported by Richard Mudgett)
* ASTERISK-20643 - SIP ICE support - remove hardcoded
limitation on SDP size, make ICE support disabled by default in
SIP, maybe provide a better warning message
(Reported by Roy)
* ASTERISK-26980 - pjsip: Clean up WebRTC disables
(Reported by abelbeck)
* 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-23735 - Transcoding makes bad choice in high-rate
translations
(Reported by Richard Kenner)
* ASTERISK-27445 - ARI: Updating a bridge gives wrong error
message.
(Reported by Frank Durden)
* ASTERISK-24662 - [patch] column and row headers for Signed
Linear format variants in output of 'core show translation' are
ambiguous
(Reported by Rusty Newton)
* ASTERISK-27353 - H323 audio starts with a delay of 2
seconds.
(Reported by Marco Giordani)
* ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
media
(Reported by Kevin Harwell)
* ASTERISK-27437 - [patch] ICE: server-reflexive candidates
(srflx) with Dual-Stack.
(Reported by Alexander Traud)
* ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
IPv6 addresses.
(Reported by Alexander Traud)
* ASTERISK-27435 - [patch] configure:
pjsip_evsub_set_uas_timeout not found.
(Reported by Alexander Traud)
* ASTERISK-27431 - Asterisk fails to build when openssl headers
are not installed.
(Reported by Corey Farrell)
* ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
(Reported by Ivan Larionov)
* ASTERISK-27421 - RTP source learning not working with devices
that have some clock issues
(Reported by nappsoft)
* ASTERISK-27361 - Attended transfer crashes in Asterisk
13.17.2
(Reported by Alessandro Pimenta)
* ASTERISK-27238 - Bridging: Crash freeing a frame that's
already been freed
(Reported by Richard Kenner)
* ASTERISK-27412 - core: Audiohook freeing interpolated frame
when it shouldn't.
(Reported by Mikhail)
* ASTERISK-27423 - app_record: We set the RECORD_STATUS
channel variable before closing the file
(Reported by George Joseph)
* ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
insert same ip address in "source ip address" and "destination
ip address" fields in HEP packets
(Reported by Max Norba)
* ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
is equal to RemoteAddress)
(Reported by Vasilii Rogin)
* ASTERISK-27415 - asterisk.conf: Setting astctl without
setting astrundir is ineffective.
(Reported by Corey Farrell)
* 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-27393 - res_pjsip: Crash occurs when an empty
contact read from astdb or database
(Reported by Aaron An)
* ASTERISK-27290 - res_pjsip: PIDF contact field has
malformed/invalid XML
(Reported by basildane)
* ASTERISK-27032 - res_pjsip: TLS options do not handle empty
values
(Reported by seanchann.zhou)
* ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
enabled but not configured.
(Reported by Alexander Traud)
* ASTERISK-26426 - format_ogg_opus: remove from source
(Reported by Kevin Harwell)
* ASTERISK-27378 - Modules: Fix issues with CLI completion.
(Reported by Corey Farrell)
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27390 - Audit menuselect module dependencies
(Reported by Corey Farrell)
* ASTERISK-27389 - Optional API modules should not allow
unload.
(Reported by Corey Farrell)
* ASTERISK-27369 - Bridge() dialplan application fails without
setting BRIDGERESULT channel variable
(Reported by James Terhune)
* ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
documentation
(Reported by Igor Goncharovsky)
* ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
'imap_delete_old_greeting'
(Reported by Anthony Messina)
* ASTERISK-27194 - jitterbuffer: Does not handle case where
translator returns null frame.
(Reported by Joshua Elson)
* ASTERISK-26639 - core: Disabling xmldoc support does not
work. Also results in abort during Asterisk startup.
(Reported by Mr Dini)
* ASTERISK-27372 - ARI: Node ARI client broken in latest
versions of 13 and 14
(Reported by Benjamin Keith Ford)
* ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
absence of the Expires header field with an unsubscribe action.
(Reported by Jonathan Cloots)
* ASTERISK-25960 - The config_hook unit test causes Asterisk to
crash if run a second time
(Reported by George Joseph)
* ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
when rtp_ipv6 set to yes
(Reported by Martin Cisárik)
* ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
curl is loaded
(Reported by Ronald Raikes)
* ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
but first on SDP media level.
(Reported by Alexander Traud)
* ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
Assertion on un/re-load: mod.id == -1
(Reported by Tzafrir Cohen)
* ASTERISK-23462 - Cannot disable SIP debugging via CLI after
enabling with conf file option - also 'sip set debug off'
reports debugging disabled, when it really isn't
(Reported by Rusty Newton)
* ASTERISK-27328 - Missing openssl dependencies in
res_rtp_asterisk and tcptls
(Reported by Tzafrir Cohen)
* ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
(o=) contains local address.
(Reported by Alexander Traud)
* ASTERISK-27343 - Fails to build in FreeBSD due to
sys/sysmacros.h not existing there
(Reported by Guido Falsi)
* ASTERISK-27340 - backtrace.c: Crash due to double-free.
(Reported by Corey Farrell)
* ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
stopping.
(Reported by Alexander Traud)
* ASTERISK-27333 - sip_to_pjsip not correctly handling
disallow=all directive
(Reported by Torrey Searle)
Improvements made in this release:
-----------------------------------
* ASTERISK-24297 - cdr.c: Minor code optimizations.
(Reported by Richard Mudgett)
* ASTERISK-27449 - [PATCH] When failing to acquire target
during attended transfer, display wanted extension
(Reported by Niklas Larsson)
* ASTERISK-27456 - app_voicemail: Add new object for
VoicemailUserEntry
(Reported by sungtae kim)
* ASTERISK-27380 - ast_coredumper: allow pointing out the
asterisk binary explicitly
(Reported by Tzafrir Cohen)
* ASTERISK-23556 - Compilation warning for invert.c (array
subscript is above array bounds)
(Reported by Marcello Ceschia)
* ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
(Reported by Richard Mudgett)
* ASTERISK-27335 - CDR performance needs improvement.
(Reported by Richard Mudgett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.0
Thank you for your continued support of Asterisk!
----- 13.18.5 -----
The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18.
The available releases are released as versions 13.18.5, 14.7.5,
15.1.5 and 13.18-cert2.
The following security vulnerabilities were resolved in these versions:
* AST-2017-014: Crash in PJSIP resource when missing a contact header
A select set of SIP messages create a dialog in Asterisk. Those
SIP messages must contain a contact header. For those messages,
if the header was not present and using the PJSIP channel driver,
it would cause Asterisk to crash. The severity of this vulnerability
is somewhat mitigated if authentication is enabled. If authentication
is enabled a user would have to first be authorized before reaching
the crash point.
For a full list of changes in the current releases, please see the ChangeLogs:
https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
The security advisory is available at:
https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
Thank you for your continued support of Asterisk!
----- 13.18.4 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert9, 13.18.4,
14.7.4 and 15.1.4.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-012: Remote Crash Vulnerability in RTCP Stack
If a compound RTCP packet is received containing more than
one report (for example a Receiver Report and a Sender
Report) the RTCP stack will incorrectly store report
information outside of allocated memory potentially causing
a crash.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-012.html
http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
Thank you for your continued support of Asterisk!
----- 13.18.3 -----
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
security releases are released as versions 13.13-cert8, 13.18.3,
14.7.3 and 15.1.3.
The release of these versions resolves the following security
vulnerabilities:
* AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
If the chan_skinny (AKA SCCP protocol) channel driver is
flooded with certain requests it can cause the asterisk
process to use excessive amounts of virtual memory
eventually causing asterisk to stop processing requests of
any kind.
For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
The security advisories are available at:
http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
Thank you for your continued support of Asterisk!
----- 13.18.2 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.18.2.
The release of Asterisk 13.18.2 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
character isn't allowed any more
(Reported by Michael Maier)
* ASTERISK-27391 - Regression: Deadlock between AOR named lock
and pjproject grp lock
(Reported by shaurya jain)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.2
Thank you for your continued support of Asterisk!
----- 13.18.0 -----
The Asterisk Development Team would like to announce the release
of Asterisk 13.18.0.
The release of Asterisk 13.18.0 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-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)
Bugs fixed in this release:
-----------------------------------
* 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 Benoît 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 Engström)
* 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 Seán 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-27093 - ODBC deadlocks when app_directory tries to
play back non-existent voicemail greeting
(Reported by James Terhune)
New Features made in this release:
-----------------------------------
* 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)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.18.0
Thank you for your continued support of Asterisk!
----- 13.17.0 ----
The Asterisk Development Team would like to announce the release
of Asterisk 13.17.0.
The release of Asterisk 13.17.0 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:
Bugs fixed in this release:
-----------------------------------
* 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-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-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-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-27060 - Comment typo format_g729.c
(Reported by Matthew Fredrickson)
* ASTERISK-27026 - res_ari: Crash when no ari.conf
configuration file exists
(Reported by Ronald Raikes)
* 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-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
get_write_timeout
(Reported by Jørgen 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 Engström)
* ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
when request and To URI differ
(Reported by Yasin CANER)
* ASTERISK-26789 - Audit manipulation of channel flags without
locks
(Reported by Joshua Colp)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
6869i)
(Reported by Matthias Binder)
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-26124 - res_agi: Set audio format for EAGI audio
stream
(Reported by John Fawcett)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0
Thank you for your continued support of Asterisk!
2018-01-23 09:26:08 +01:00
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2015-12-06 00:29:05 +01:00
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Update to Asterisk 13.15.0. This is mostly a bug fix release with a few
minor enhancements. 13.14.1 was released to fix AST-2017-001.
----- 13.15.0
The Asterisk Development Team would like to announce the release of
Asterisk 13.15.0.
The release of Asterisk 13.15.0 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:
*New Features made in this release:*
-----------------------------------
- [ASTERISK-26878 <https://issues.asterisk.org/jira/browse/ASTERISK-26878>]
- func_channel: Add ability to get the callid so dialplan has access to it.
(Reported by Richard Mudgett)
- [ASTERISK-26863 <https://issues.asterisk.org/jira/browse/ASTERISK-26863>]
- res_pjsip: Add endpoint identification scheme based on a configured SIP
header/value
(Reported by Matt Jordan)
- [ASTERISK-17428 <https://issues.asterisk.org/jira/browse/ASTERISK-17428>]
- [patch] Allow "Comedian Mail" branding to be removed
(Reported by John Covert)
*Bugs fixed in this release:*
-----------------------------------
- [ASTERISK-26851 <https://issues.asterisk.org/jira/browse/ASTERISK-26851>]
- res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
(Reported by Richard Begg)
- [ASTERISK-26897 <https://issues.asterisk.org/jira/browse/ASTERISK-26897>]
- chan_sip: Security vulnerability with client code header
(Reported by Alex Villacís Lasso)
- [ASTERISK-26916 <https://issues.asterisk.org/jira/browse/ASTERISK-26916>]
- res_pjsip: Excessive refcount reached on transport ao2 object
(Reported by Ross Beer)
- [ASTERISK-26705 <https://issues.asterisk.org/jira/browse/ASTERISK-26705>]
- libasteriskssl.so not found when asterisk is installed for the 1st time
(Reported by George Joseph)
- [ASTERISK-26850 <https://issues.asterisk.org/jira/browse/ASTERISK-26850>]
- res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field
in HEP packets
(Reported by Max Norba)
- [ASTERISK-26484 <https://issues.asterisk.org/jira/browse/ASTERISK-26484>]
- res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from'
argument.
(Reported by Vinod Dharashive)
- [ASTERISK-26776 <https://issues.asterisk.org/jira/browse/ASTERISK-26776>]
- res_pjsip_pubsub: Crash when generating xpidf content
(Reported by Andrew Green)
- [ASTERISK-26880 <https://issues.asterisk.org/jira/browse/ASTERISK-26880>]
- Asterisk crashes when multiple speex users join confbridge with pp_vad
and dtx enabled
(Reported by Kirsty Tyerman)
- [ASTERISK-26862 <https://issues.asterisk.org/jira/browse/ASTERISK-26862>]
- app_queue: Queue stops calling members with local interface after
forwarding in previous call
(Reported by Robert Mordec)
- [ASTERISK-26732 <https://issues.asterisk.org/jira/browse/ASTERISK-26732>]
- res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
(Reported by Dan Jenkins)
- [ASTERISK-26879 <https://issues.asterisk.org/jira/browse/ASTERISK-26879>]
- PJSIP external_media_address ignored if no local_net options are provided
(Reported by Matt Jordan)
- [ASTERISK-26867 <https://issues.asterisk.org/jira/browse/ASTERISK-26867>]
- autochan: Locking in a function ast_autochan_destroy() on destroyed
channel (after masquerade).
(Reported by Krzysztof Trempala)
- [ASTERISK-26869 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-26668>]
- core: Malformed pattern matching extension (various factors) results in
crash
(Reported by xrobau)
- [ASTERISK-26865 <https://issues.asterisk.org/jira/browse/ASTERISK-26865>]
- chan_iax2: Reload of iax peer results in loss of host address/port
(Reported by Richard Begg)
- [ASTERISK-26872 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-26717>]
- Document the fact that Asterisk HEP support only works with the PJSIP
channel driver
(Reported by Olivier Krief)
- [ASTERISK-26643 <https://issues.asterisk.org/jira/browse/ASTERISK-26643>]
- Extra new line in Device field of DeviceStateChange AMI Event after
restart of Asterisk
(Reported by Roman Bedros)
- [ASTERISK-25237 <https://issues.asterisk.org/jira/browse/ASTERISK-25237>]
- stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
(Reported by Smirnov Aleksey)
- [ASTERISK-26857 <https://issues.asterisk.org/jira/browse/ASTERISK-26857>]
- chan_pjsip: Dialplan function race condition
(Reported by Joshua Colp)
- [ASTERISK-26841 <https://issues.asterisk.org/jira/browse/ASTERISK-26841>]
- chan_sip: Call not cancelled after receiving a 422 response
(Reported by Jean Aunis - Prescom)
- [ASTERISK-26822 <https://issues.asterisk.org/jira/browse/ASTERISK-26822>]
- pjsip/cli_commands: pjsip show channelstats shows wrong codec
(Reported by Kevin Harwell)
- [ASTERISK-26685 <https://issues.asterisk.org/jira/browse/ASTERISK-26685>]
- res_pjsip: Crash when using IPv6 and Transport ws,wss
(Reported by Michael Balen)
- [ASTERISK-24562 <https://issues.asterisk.org/jira/browse/ASTERISK-24562>]
- app_voicemail: Cannot set fromstring on a per-mailbox basis
(Reported by Mark Scholten)
- [ASTERISK-26598 <https://issues.asterisk.org/jira/browse/ASTERISK-26598>]
- Saynumber is trying to get "and" from "digits/" subfolder
(Reported by Jonathan Harris)
- [ASTERISK-17067 <https://issues.asterisk.org/jira/browse/ASTERISK-17067>]
- Long lines in call files cause spurious syntax error
(Reported by Dave Olszewski)
- [ASTERISK-26796 <https://issues.asterisk.org/jira/browse/ASTERISK-26796>]
- res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
(Reported by Jørgen H)
- [ASTERISK-25628 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-26825>]
- pjsip.conf.sample: user_agent: still refers to branch 12
(Reported by Tzafrir Cohen)
- [ASTERISK-26823 <https://issues.asterisk.org/jira/browse/ASTERISK-26823>]
- PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not
exist
(Reported by Mark Michelson)
- [ASTERISK-26623 <https://issues.asterisk.org/jira/browse/ASTERISK-26623>]
- res_pjsip: Crash when calling PJSIPShowEndpoint
(Reported by Jørgen H)
- [ASTERISK-26808 <https://issues.asterisk.org/jira/browse/ASTERISK-26808>]
- res_pjsip_outbound_registration doesn't know about network change events
(Reported by George Joseph)
- [ASTERISK-26313 <https://issues.asterisk.org/jira/browse/ASTERISK-26313>]
- chan_sip : Asterisk restart seems to be required for changing encryption
option
(Reported by benasse)
- [ASTERISK-26781 <https://issues.asterisk.org/jira/browse/ASTERISK-26781>]
- bridge: Passing the 'p' (play tone) flag to Bridge() application results
in garbled audio
(Reported by Sean Bright)
- [ASTERISK-26782 <https://issues.asterisk.org/jira/browse/ASTERISK-26782>]
- res_pjsip: URI requirement for fields is not consistently documented and
error does not provide indication
(Reported by Peter Sokolov)
- [ASTERISK-26812 <https://issues.asterisk.org/jira/browse/ASTERISK-26812>]
- [patch] Fix download_externals To Allow The Use Of curl Or wget
(Reported by Michael L. Young)
- [ASTERISK-18271 <https://issues.asterisk.org/jira/browse/ASTERISK-18271>]
- Pattern matching with res_config_mysql extensions does not behave as
expected
(Reported by Charlie Smurthwaite)
- [ASTERISK-26669 <https://issues.asterisk.org/jira/browse/ASTERISK-26669>]
- PJSIP Segfault 13.13.1 (Bundled PJSIP)
(Reported by Nic Colledge)
- [ASTERISK-18731 <https://issues.asterisk.org/jira/browse/ASTERISK-18731>]
- [patch] DUNDi weight parameter not processed correctly
(Reported by Peter Racz)
- [ASTERISK-26580 <https://issues.asterisk.org/jira/browse/ASTERISK-26580>]
- [patch] Error during LDAP modify action when user unregisters
(Reported by Nicholas John Koch)
- [ASTERISK-26799 <https://issues.asterisk.org/jira/browse/ASTERISK-26799>]
- res_pjsip: Using an auth object for inbound and outbound authentication
fails.
(Reported by Richard Mudgett)
- [ASTERISK-26738 <https://issues.asterisk.org/jira/browse/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 <https://issues.asterisk.org/jira/browse/ASTERISK-25893>]
- Function vmauthenticate accesses uninitialized memory
(Reported by Filip Jenicek)
- [ASTERISK-26802 <https://issues.asterisk.org/jira/browse/ASTERISK-26802>]
- [patch] Integrity Check Of PJSIP Download Fails
(Reported by Michael L. Young)
- [ASTERISK-15858 <https://issues.asterisk.org/jira/browse/ASTERISK-15858>]
- [patch] Fix query with double backslash in string literals and stop log
warnings
(Reported by Humberto Figuera)
- [ASTERISK-26057 <https://issues.asterisk.org/jira/browse/ASTERISK-26057>]
- res_config_sqlite3 uses incorrect query - unnecessary escape
(Reported by Stepan)
- [ASTERISK-23457 <https://issues.asterisk.org/jira/browse/ASTERISK-23457>]
- SQlite3: Realtime queue loading fails after PRAGMA query result
(Reported by Scott Griepentrog)
- [ASTERISK-26794 <https://issues.asterisk.org/jira/browse/ASTERISK-26794>]
- http: Crash on Reload Only in ast_tcptls_server_start
(Reported by Joshua Elson)
- [ASTERISK-26714 <https://issues.asterisk.org/jira/browse/ASTERISK-26714>]
- Phone default have not ringing on ARM
(Reported by Igor Goncharovsky)
- [ASTERISK-26696 <https://issues.asterisk.org/jira/browse/ASTERISK-26696>]
- pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on
subscription refresh
(Reported by Zach R)
- [ASTERISK-26756 <https://issues.asterisk.org/jira/browse/ASTERISK-26756>]
- res_pjsip_mwi: Asterisk does not terminate MWI subscription
(Reported by Carl Fortin)
- [ASTERISK-26109 <https://issues.asterisk.org/jira/browse/ASTERISK-26109>]
- Asterisk fails building with OpenSSL 1.1.0
(Reported by Tzafrir Cohen)
- [ASTERISK-26723 <https://issues.asterisk.org/jira/browse/ASTERISK-26723>]
- VoiceMailPlayMsg not playing messages via realtime
(Reported by Ryan Rittgarn)
- [ASTERISK-18286 <https://issues.asterisk.org/jira/browse/ASTERISK-18286>]
- [patch] 'Silence' is truncated in Record()
(Reported by var)
- [ASTERISK-26248 <https://issues.asterisk.org/jira/browse/ASTERISK-26248>]
- chan_pjsip: Error when calling PJSIP client with domain specified
(Reported by Norbert Varga)
- [ASTERISK-26788 <https://issues.asterisk.org/jira/browse/ASTERISK-26788>]
- core: Protect flags during ast_waitfor
(Reported by Joshua Colp)
- [ASTERISK-26115 <https://issues.asterisk.org/jira/browse/ASTERISK-26115>]
- pbx: AMI Originate ignore "failed" extension on call failure
(Reported by Nasir Iqbal)
- [ASTERISK-26785 <https://issues.asterisk.org/jira/browse/ASTERISK-26785>]
- configs/samples: The 'identify' entry is in the wrong section in
sorcery.conf.sample
(Reported by Torrey Searle)
- [ASTERISK-26772 <https://issues.asterisk.org/jira/browse/ASTERISK-26772>]
- Crash in srv.c on startup with pjsip
(Reported by nappsoft)
- [ASTERISK-26770 <https://issues.asterisk.org/jira/browse/ASTERISK-26770>]
- res_stasis_device_state: Duplicate subscriptions when multiple received
at same time
(Reported by Joshua Colp)
*Improvements made in this release:*
-----------------------------------
- [ASTERISK-26864 <https://issues.asterisk.org/jira/browse/ASTERISK-26864>]
- res_pjsip_session: Add support for overlap dialling
(Reported by Richard Begg)
- [ASTERISK-26846 <https://issues.asterisk.org/jira/browse/ASTERISK-26846>]
- chan_sip: Add rtcp-mux support
(Reported by Sean Bright)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.15.0
*Thank you for your continued support of Asterisk!*
----- 13.14.0
The Asterisk Development Team has announced the release of Asterisk 13.14.0.
The release of Asterisk 13.14.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following are the issues resolved in this release:
New Features made in this release:
-----------------------------------
* ASTERISK-26630 - Make logging PJPROJECT messages a bit easier
(Reported by Richard Mudgett)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported
by nappsoft)
* 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-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in
wrong byte order on Intel platform when using slin codec
(Reported by Frankie Chin)
* 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 Jørgen 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)
Improvements made in this release:
-----------------------------------
* 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)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.14.0
Thank you for your continued support of Asterisk!
-----
2017-05-14 00:39:13 +02:00
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