pkglint -r --network --only "migrate"
As a side-effect of migrating the homepages, pkglint also fixed a few
indentations in unrelated lines. These and the new homepages have been
checked manually.
Performing substitutions during post-patch breaks tools such as mkpatches,
making it very difficult to regenerate correct patches after making changes,
and often leading to substituted string replacements being committed.
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.htmlhttp://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!
The Asterisk Development Team would like to announce the release
of Asterisk 13.16.0.
The release of Asterisk 13.16.0 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-26982 - chan_sip: rtcp_mux setting may cause ice
completion failure/delay if client offers rtcp-mux as
negotiable
(Reported by Stefan Engström)
* ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with
authentication failure 10 or 110
(Reported by Javier Riveros)
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
events
(Reported by Ove Aursand)
* ASTERISK-26998 - res_pjsip_session: INVITE retransmissions
could still setup the same call again.
(Reported by Richard Mudgett)
* ASTERISK-26143 - res_rtp_asterisk: One way audio when
transcoding
(Reported by Henning Holtschneider)
* ASTERISK-26606 - tcptls: Incorrect OpenSSL function call
leads to misleading error report
(Reported by Bob Ham)
* ASTERISK-26983 - Crash in Manager Reload when TLS Config
Changes
(Reported by Joshua Elson)
* ASTERISK-25032 - [patch]cel_odbc sometimes inserts CEL with
wrong eventtime
(Reported by Etienne Lessard)
* ASTERISK-26173 - func_cdr: CDR function does not permit empty
values to be assigned
(Reported by gkloepfer)
* ASTERISK-25506 - [patch]CONFBRIDGE failure after an
app_confbrige.so module reload results in segfault or
error/warning messages.
(Reported by Frederic LE FOLL)
* ASTERISK-24529 - Using AMI Action Bridge to on an already
bridged channel causes the incorrect return priority to be used
(Reported by Corey Farrell)
* ASTERISK-26860 - Upon RTCP reception, netsock2.c:210
ast_sockaddr_split_hostport: Port missing in (null)
(Reported by Evers Lab)
* ASTERISK-26922 - chan_sip: tcpbind uses wrong source address
(Reported by Ksenia)
* ASTERISK-26974 - res_pjsip: Deadlock in T.38 framehook
(Reported by Richard Mudgett)
* ASTERISK-26908 - res_pjsip: The ChanIsAvail causes a
res_pjsip session to be leaked.
(Reported by Richard Mudgett)
* ASTERISK-25823 - SIGSEGV, Segmentation fault. -
../sysdeps/x86_64/strlen.S: No such file or directory.
(Reported by Andreas Krüger)
* ASTERISK-26951 - chan_sip: ACK with SDP does not update a
direct media bridge
(Reported by Jean Aunis - Prescom)
* ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build
fails for non-SSE2 instrunction Linux
(Reported by abelbeck)
* ASTERISK-26926 - func_speex: Crash caused by frame with no
datalen
(Reported by Richard Kenner)
* ASTERISK-26929 - pjsip: Add database tables for RLS
(Reported by Joshua Colp)
* ASTERISK-26953 - Asterisk crash if hep.conf have some missing
parameters
(Reported by Joel Vandal)
* ASTERISK-26890 - STUN server with non-default-route transport
causes INVITE delay
(Reported by George Joseph)
* ASTERISK-26692 - res_rtp_asterisk: Crash in
dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
(Reported by scgm11)
* ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP
address string
(Reported by Niklas Larsson)
* ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when
receiving packet
(Reported by Adagio)
* ASTERISK-26613 - format_wav: wav16 format read file only by
320 - half of frame
(Reported by Vitaly K)
* ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in
MixMonitor
(Reported by Ivan Myalkin)
* ASTERISK-21856 - STUN never works when asterisk started
without internet access
(Reported by Jeremy Kister)
* ASTERISK-20984 - Audible clicks when playing sox encoded au
file with STREAM FILE AGI command
(Reported by Roman S.)
* ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use
same IP as explicit transport
(Reported by Richard Begg)
* ASTERISK-26903 - Listening TCP/TLS sockets stop when
temporarily out of open files
(Reported by Walter Doekes)
* ASTERISK-26528 - [UBSAN] strings.h:signed integer overflow in
ast_str_case_hash
(Reported by Badalian Vyacheslav)
* ASTERISK-26928 - pjsip: Add database tables for PUBLISH
support
(Reported by Joshua Colp)
* ASTERISK-26927 - pjproject_bundled: Crash on
pj_ssl_get_info() while ioqueue_on_read_complete().
(Reported by Alexander Traud)
* ASTERISK-26905 - pjproject_bundled: Merge 3 upstream
deadlock patches into bundled
(Reported by Ross Beer)
* ASTERISK-26897 - chan_sip: Security vulnerability with client
code header
(Reported by Alex Villacís Lasso)
* ASTERISK-25974 - Unused realtime MOH classes not purged on
'moh reload'
(Reported by Sébastien Couture)
* ASTERISK-26916 - res_pjsip: Excessive refcount reached on
transport ao2 object
(Reported by Ross Beer)
* ASTERISK-21721 - SIP Failed to parse multiple Supported:
headers
(Reported by Olle Johansson)
* ASTERISK-26915 - chan_sip: Session Timers required but
refused wrongly.
(Reported by Alexander Traud)
* ASTERISK-26363 - res_pjsip: Bye sent to sip trunk is not
authenticated even after receiving a 407 error code
(Reported by Yaacov Akiba Slama)
* ASTERISK-26896 - Overflow of buffer to PQEscapeStringConn
with large app_args causes ABRT
(Reported by twisted)
* ASTERISK-26705 - libasteriskssl.so not found when asterisk is
installed for the 1st time
(Reported by George Joseph)
* ASTERISK-21009 - xmpp_pubsub_unsubscribe: Could not create IQ
when creating pubsub unsubscription on client
(Reported by
Marcello Ceschia)
* ASTERISK-25490 - [patch]SDP crypto tag is validated
incorrectly
(Reported by Joerg Sonnenberger)
* ASTERISK-24712 - xmpp: starttls problem causes connection
spew
(Reported by Matthias Urlichs)
* ASTERISK-26086 - res_musiconhold: format option is not
documented adequately
(Reported by Jens Bürger)
* ASTERISK-23996 - No core dumps because of res_musiconhold
chdir.
(Reported by Walter Doekes)
* ASTERISK-26814 - pjproject_bundled build fails to download
pjproject source when using cURL
(Reported by Gergely Dömsödi)
* ASTERISK-23510 - JABBER_STATUS fails with improper code 7 for
unavailable clients
(Reported by Anthony Critelli)
* ASTERISK-21855 - Asterisk crashes when XMPP message is sent
(JabberSend) and no internet connection is available
(Reported by Jeremy Kister)
* ASTERISK-25622 - WARNING for "JABBER: socket read error"
should be more specific
(Reported by Sean Darcy)
* ASTERISK-26818 - cdr: Problem setting variables in h exten
(Reported by scgm11)
* ASTERISK-26875 - app_mixmonitor: Recording out of sync when
183 but no RTP
(Reported by Aaron An)
Improvements made in this release:
-----------------------------------
* ASTERISK-26088 - Investigate heavy memory utilization by
res_pjsip_pubsub
(Reported by Richard Mudgett)
* ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report
channel name with res_hep_rtcp when using chan_sip
(Reported by Nir Simionovich (GreenfieldTech - Israel))
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.16.0
Thank you for your continued support of Asterisk!
that the first two don't affect pkgsrc as we are using chan_sip
not PJSIP. The last only affects users of SCCP, which is Cisco's
proprietary protocol.
----- AST-2017-002
A remote crash can be triggered by sending a SIP packet to
Asterisk with a specially crafted CSeq header and a Via
header with no branch parameter. The issue is that the
PJSIP RFC 2543 transaction key generation algorithm does
not allocate a large enough buffer. By overrunning the
buffer, the memory allocation table becomes corrupted,
leading to an eventual crash.
This issue is in PJSIP, and so the issue can be fixed
without performing an upgrade of Asterisk at all. However,
we are releasing a new version of Asterisk with the bundled
PJProject updated to include the fix.
If you are running Asterisk with chan_sip, this issue does
not affect you.
----- AST-2017-003
The multi-part body parser in PJSIP contains a logical
error that can make certain multi-part body parts attempt
to read memory from outside the allowed boundaries. A
specially-crafted packet can trigger these invalid reads
and potentially induce a crash.
The issue is within the PJSIP project and not in Asterisk.
Therefore, the problem can be fixed without upgrading
Asterisk. However, we will be releasing a new version of
Asterisk where the bundled version of PJSIP has been
updated to have the bug patched.
If you are using Asterisk with chan_sip, this issue does
not affect you.
----- AST-2017-004
A remote memory exhaustion can be triggered by sending an
SCCP packet to Asterisk system with chan_skinny enabled
that is larger than the length of the SCCP header but
smaller than the packet length specified in the header. The
loop that reads the rest of the packet doesn't detect that
the call to read() returned end-of-file before the expected
number of bytes and continues infinitely. The partial
data message logging in that tight loop causes Asterisk to
exhaust all available memory.
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!
-----
MASTER_SITES= site1 \
site2
style continuation lines to be simple repeated
MASTER_SITES+= site1
MASTER_SITES+= site2
lines. As previewed on tech-pkg. With thanks to rillig for fixing pkglint
accordingly.
minor improvements.
The Asterisk Development Team has announced the release of Asterisk 13.13.0.
The release of Asterisk 13.13.0 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-26595 - ARI: Add the ability to control the source of
video in a multi-party mixing bridge (Reported by Matt Jordan)
* ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing
events (Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26608 - Compile and link failures on OpenBSD (Reported
by snuffy)
* ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
manipulation through agi (Reported by Morten Tryfoss)
* ASTERISK-26520 - codec_opus: Generated fmtp line has no content
(Reported by scgm11)
* ASTERISK-26605 - codec_opus: Spammed warning when Opus
negotiated but codec_opus not loaded. (Reported by Richard
Mudgett)
* ASTERISK-26516 - pjsip: Memory corruption with possible memory
leak. (Reported by Richard Mudgett)
* ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and
makes asterisk CLI read garbage (Reported by George Joseph)
* ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold
temporarily locks up set (Reported by Jason)
* ASTERISK-26575 - testsuite: Need to check PJSIP functionality
when res_srtp is not loaded. (Reported by Joshua Colp)
* ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by
Dmitry Melekhov)
* ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11
regressions (Reported by Matt Jordan)
* ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George
Joseph)
* ASTERISK-26509 - A few non-critical deprecation warnings when
building on Ubuntu 16.10 (Reported by Jonathan Harris)
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
calls after 2 minutes - rtptimeout behaving badly - regression
(Reported by Michael Keuter)
* ASTERISK-26468 - ari: Bridge events stop working after this
sequence of ARI calls (Reported by Daniele Pallastrelli)
* ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic
payload types. (Reported by Alexander Traud)
* ASTERISK-26549 - app_dial: When PickupChan() is used some
channels may have incorrect device state (Reported by Joshua
Colp)
* ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats
to maximum (Reported by Joshua Colp)
* ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele
Giacone)
* ASTERISK-26476 - chan_sip: Incorrect display option "Outbound
reg. retry 403" in "sip show settings" (Reported by Sergey
Grachev)
* ASTERISK-26537 - AMI: NewConnectedLine event is not documented
(Reported by Etienne Lessard)
* ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as
argument 2 to memcpy (Reported by Badalian Vyacheslav)
* ASTERISK-26524 - astobj2: data_size variable is wasted space
when AO2_DEBUG is not enabled. (Reported by Corey Farrell)
* ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian
Gilmour)
* ASTERISK-26387 - Asterisk segfaults shortly after starting even
with no active calls. (Reported by Harley Peters)
* ASTERISK-26514 - Super Awesome Company: Don't specify transport
in pjsip.conf (Reported by Rusty Newton)
* ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing
enough to be a nuisance (Reported by Joshua Colp)
* ASTERISK-26510 - pjproject_bundled uses the --strip-components
option of tar which isn't supported in older versions (Reported
by George Joseph)
* ASTERISK-22480 - Embedded pjproject: build.mak contains
hardcoded full path to version.mak (Reported by Matt Jordan)
* ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change
(Reported by Bill Brigden)
* ASTERISK-26503 - app_voicemail: Asterisk crashes when
MailboxExists is used (Reported by Doug Lytle)
* ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can
cause audio loss and wonkiness (Reported by Andreas Wetzel)
* ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack)
installations. (Reported by Alexander Traud)
* ASTERISK-26421 - Segmentation Fault with ARI originate into
mixing bridge with 43 clients (Reported by Andrew Nagy)
* ASTERISK-26444 - 'features show' command in CLI does not return
prompt. (Reported by John Kiniston)
* ASTERISK-26482 - [patch] chan_pjsip: segfault on already
disconnected session (Reported by Alexei Gradinari)
* ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes
File not Module (Reported by Alexander Traud)
* ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by
Tzafrir Cohen)
* ASTERISK-26439 - chan_rtp: Crash when originating (Reported by
Kayode)
* ASTERISK-26462 - [patch] app_queue: While using queues with
realtime, setting back to an empty context doesn't stop the exit
key usage (Reported by Leandro Dardini)
* ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT
detection triggered. (Reported by Alexander Traud)
* ASTERISK-26618 - build: Backport addition of librt check to
configure.ac (Reported by Kevin Harwell)
Improvements made in this release:
-----------------------------------
* ASTERISK-25063 - [patch]add X.509 subject alternative name
support to Asterisk TLS support (Reported by Maciej Szmigiero)
* ASTERISK-26558 - app_queue: add variable to know if the call is
not answered after a queue (Reported by scgm11)
* ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
(Reported by scgm11)
* ASTERISK-26538 - codec_opus: Add sample to
configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
* ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and
'ari set debug' CLI commands (Reported by Matt Jordan)
* ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by
blacklisting host subnets that are not involved in RTP (Reported
by Michael Walton)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.13.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 13.12.2.
The release of Asterisk 13.12.2 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-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
calls after 2 minutes - rtptimeout behaving badly - regression
(Reported by Michael Keuter)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.2
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 13.12.1.
The release of Asterisk 13.12.1 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-26503 - app_voicemail: Asterisk crashes when
MailboxExists is used (Reported by Doug Lytle)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.1
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 13.12.0.
The release of Asterisk 13.12.0 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-26277 - Add dialplan function
PJSIP_SEND_SESSION_REFRESH that sends a session refresh to
update formats on a channel after session establishment
(Reported by Matt Jordan)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26477 - pjproject: SEGV during SSL operations
(Reported by George Joseph)
* ASTERISK-17470 - [patch] - When videosupport=yes, asterisk
allows one end peer to send video, even though the other end
supports only audio. (Reported by effie mouzeli)
* ASTERISK-26416 - pjproject-bundled: configure fails to check for
all required utilities (Reported by Corey Farrell)
* ASTERISK-26466 - core: Be forgiving on external callerid that
may be flawed so we don't drop events (Reported by Richard
Mudgett)
* ASTERISK-26362 - res_config_mysql: Broken after 13.10 (Reported
by Carlos Chavez)
* ASTERISK-26446 - app_dial: There's no way to override the
hangupcause on unanswered channels (Reported by George Joseph)
* ASTERISK-24311 - Populating database via Alembic fails when
using same database for multiple schema sets (Reported by Dafi
Ni)
* ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No
Symmetric Response. (Reported by Alexander Traud)
* ASTERISK-26426 - format_ogg_opus: remove from source (Reported
by Kevin Harwell)
* ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
bindaddr=[::] (Reported by Jacek)
* ASTERISK-25468 - Deadlock in chan_sip - core show locks shows
do_monitor lock (Reported by Barry Flanagan)
* ASTERISK-26397 - manager: PresenceState action crashes Asterisk
14 (Reported by Andrew Nagy)
* ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by
Joshua Colp)
* ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so
instructed (Reported by Tzafrir Cohen)
* ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled
(Reported by Anthony Messina)
* ASTERISK-26263 - SQL error when using realtime and registering
extension / inserting into ps_contacts (Reported by Jeppe Ryskov
Larsen)
* ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten
for connectionful protocols (Reported by Joshua Colp)
* ASTERISK-26367 - rtp: Timestamps broken when video frame is
across multiple RTP packets (Reported by Joshua Colp)
* ASTERISK-26375 - res_pjsip_transport_management: Log message
states seconds, but time value is milliseconds (Reported by
Joshua Colp)
* ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
by Aaron Hamstra)
* ASTERISK-26360 - app_queue: "queue show" output gets "failed to
extend from 240 to 327" msgs. (Reported by Richard Mudgett)
* ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes
unexpected callerid (Reported by Kevin Harwell)
* ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request
'REGISTER' failed (Reported by Dmitry Melekhov)
* ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets)
(Reported by Etienne Lessard)
* ASTERISK-26264 - res_pjsip: Crash when applying ACL from
non-existent endpoint (Reported by nappsoft)
* ASTERISK-26288 - followme: fails to reset config items to
default values on reload (Reported by Tzafrir Cohen)
* ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p'
argument) is enabled and callee rejects a call or hangs up.
(Reported by Etienne Lessard)
* ASTERISK-26331 - Crash on "core show channeltype Surrogate" in
ast_format_cap_get_names (Reported by CGI.NET)
* ASTERISK-26282 - AEL: macro-call in Dial application, macro
"lacks 's' extension" (Reported by chris de rock)
* ASTERISK-26226 - pbx: Asterisk crash on AMI action
"ShowDialplan" when there's a circular dependency between
contexts (Reported by Etienne Lessard)
* ASTERISK-26279 - pjproject-bundled: Fails to compile on Debian
6 (Reported by George Joseph)
* ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not
cleaning up properly (Reported by Alexander Traud)
* ASTERISK-26299 - app_queue: Queue application sometimes stops
calling members with Local interface (Reported by Etienne
Lessard)
* ASTERISK-26203 - res_fax: Deadlock when using
FAXOPT(gateway)=yes with Local channels (Reported by Etienne
Lessard)
* ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking
inversion in T.38 query option with features bridging code
(Reported by David Brillert)
* ASTERISK-22732 - Deadlock potential in res_fax and CCSS with
local channels. (Reported by Richard Mudgett)
* ASTERISK-26269 - res_pjsip: Wrong state for aors without
registered contacts after startup (Reported by nappsoft)
* ASTERISK-22374 - Finish mapping the sip.conf parameters to
res_sip.conf parameters (Reported by Matt Jordan)
* ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
abelbeck)
* ASTERISK-25472 - Swagger scripts are not replacing format
variable in file brief (Reported by Corey Farrell)
* ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include
annexb=no attribute. (Reported by Ali Ghavidel)
* ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but
it's not mandatory to compile it (Reported by József Dudás)
* ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities
not detected in PJProject. (Reported by Alexander Traud)
* ASTERISK-25492 - ARI: Path parameters are case sensitive
(Reported by Joshua Colp)
* ASTERISK-26233 - pbx: Failure to remove inconsistent extension
names (Reported by Corey Farrell)
* ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via
chan_pjsip (Reported by Ross Beer)
* ASTERISK-26246 - Security: Privilege escalation by AMI adding
dialplan extensions. (Reported by Richard Mudgett)
* ASTERISK-26267 - ast_register_atexit callbacks should be run on
failed startup. (Reported by Corey Farrell)
* ASTERISK-26241 - res_pjsip: When using compact headers, rpid
and pai are incorrectly generated (Reported by George Joseph)
* ASTERISK-26239 - res_pjsip_logger: An empty global/debug option
is treated as a "match all" hostname (Reported by George Joseph)
* ASTERISK-26238 - res_pjsip: Empty global default_from_user
causes crash (Reported by Joshua Colp)
* ASTERISK-25797 - app_queue: Crash when calling a queue with a
member with a forward to an nonexistent extension (Reported by
Etienne Lessard)
* ASTERISK-26268 - alembic: 'auth_username' not in PJSIP
'identify_by' enum (Reported by Joshua Colp)
* ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade +
indicate (Reported by Ross Beer)
* ASTERISK-26183 - alembic: error when using sqlalchemy version
1.1.0b2 (Reported by Kevin Harwell)
* ASTERISK-26280 - DNS lookups can block channel media paths
(Reported by Mark Michelson)
* ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a
similar treatment for module unloading as
res_pjsip_outbound_registration.c (Reported by Richard Mudgett)
* ASTERISK-26265 - Errors ignored from some parts of system
initialization. (Reported by Corey Farrell)
* ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex
for get all (Reported by Dmitry)
* ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets
with IP6 (Reported by Alexander Traud)
* ASTERISK-25996 - Remove "live_dangerously" requirement on
DB(read) (Reported by Andrew Nagy)
* ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1:
"libasteriskpj.so: undefined reference to..." (Reported by Hans
van Eijsden)
Improvements made in this release:
-----------------------------------
* ASTERISK-26409 - codec_opus: Update Asterisk to support the
translation codec. (Reported by Kevin Harwell)
* ASTERISK-26289 - Announcer channels in ConfBridges cause
inefficiencies (Reported by Mark Michelson)
* ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let
dialplan know what fax transport was used (Reported by Alexei
Gradinari)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.0
Thank you for your continued support of Asterisk!
including two security issues: AST-2016-006 and AST-2016-007.
Note that AST-2016-006 only affected setups using PJSIP, which
pkgsrc Asterisk does not.
pkgsrc changes:
- don't use gethostbyname_r on NetBSD
- eliminte conflict with new hmac(1) function on NetBSD
----- AST-2016-006
Asterisk can be crashed remotely by sending an ACK to it from an
endpoint username that Asterisk does not recognize. Most SIP
request types result in an "artificial" endpoint being looked up,
but ACKs bypass this lookup. The resulting NULL pointer results in
a crash when attempting to determine if ACLs should be applied.
This issue was introduced in the Asterisk 13.10 release and only
affects that release.
This issue only affects users using the PJSIP stack with Asterisk.
Those users that use chan_sip are unaffected.
----- AST-2016-007
The overlap dialing feature in chan_sip allows chan_sip to report
to a device that the number that has been dialed is incomplete and
more digits are required. If this functionality is used with a
device that has performed username/password authentication RTP
resources are leaked. This occurs because the code fails to release
the old RTP resources before allocating new ones in this scenario.
If all resources are used then RTP port exhaustion will occur and
no RTP sessions are able to be set up.
----- 13.11.2
The Asterisk Development Team has announced the release of Asterisk 13.11.2.
The release of Asterisk 13.11.2 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-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request
'REGISTER' failed (Reported by Dmitry Melekhov)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.2
Thank you for your continued support of Asterisk!
----- 13.11.0
The Asterisk Development Team has announced the release of Asterisk 13.11.0.
The release of Asterisk 13.11.0 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-25904 - PJSIP: add contact.updated event (Reported by
Alexei Gradinari)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26269 - res_pjsip: Wrong state for aors without
registered contacts after startup (Reported by nappsoft)
* ASTERISK-26299 - app_queue: Queue application sometimes stops
calling members with Local interface (Reported by Etienne
Lessard)
* ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1:
"libasteriskpj.so: undefined reference to..." (Reported by Hans
van Eijsden)
* ASTERISK-26237 - Fax is detected on regular calls. (Reported by
Richard Mudgett)
* ASTERISK-26227 - sqlalchemy error due to long identifier name
(Reported by Mark Michelson)
* ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
by Aaron Hamstra)
* ASTERISK-26214 - Allow arbitrary time for fax detection to end
on a channel (Reported by Richard Mudgett)
* ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
command and attended transfer handling (Reported by Ben
Smithurst)
* ASTERISK-26216 - res_fax: Deadlock when detect fax while channel
executing Playback (Reported by Richard Mudgett)
* ASTERISK-26212 - [patch] Makefile: Retain XML Declaration and
DTD in docs. (Reported by Alexander Traud)
* ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in
conditional code. (Reported by Corey Farrell)
* ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence
number even on lost packets. (Reported by Alexander Traud)
* ASTERISK-26038 - 'make install' doesn't seem to install OS/X
init files (Reported by Tzafrir Cohen)
* ASTERISK-26200 - [patch] res_pjsip_mwi: improve realtime
performance - remove unneeded check on endpoint's contacts.
(Reported by Alexei Gradinari)
* ASTERISK-26133 - app_queue: Queue members receive multiple calls
(Reported by Richard Miller)
* ASTERISK-26196 - pbx: Time based includes can leak timezone
string (Reported by Corey Farrell)
* ASTERISK-26193 - chan_sip: reference leak in mwi_event_cb
(Reported by Corey Farrell)
* ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing
DTLS failure occurred on RTP instance (Reported by Edwin
Vandamme)
* ASTERISK-26191 - threadpool: Leak on duplicate taskprocessor for
ast_threadpool_serializer_group (Reported by Corey Farrell)
* ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
(Reported by Alexander Traud)
* ASTERISK-26160 - pjsip: Updated->Reachable during qualify
(Reported by Matt Jordan)
* ASTERISK-25289 - Build System does not respect CFLAGS and
CXXFLAGS when building menuselect (Reported by Jeffrey Walton)
* ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out
of bounds and bugs (Reported by Alexei Gradinari)
* ASTERISK-26177 - func_odbc: Database handle is kept when it
should be released (Reported by Leandro Dardini)
* ASTERISK-26184 - chan_sip: Reference leaks in error paths.
(Reported by Corey Farrell)
* ASTERISK-26181 - REF_DEBUG: Node object incorrectly logged
during duplicate replacement (Reported by Corey Farrell)
* ASTERISK-26180 - PJSIP: provide valid tcp nodelay option for
reuse (Reported by Scott Griepentrog)
* ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported
by Joshua Colp)
* ASTERISK-26172 - res_sorcery_realtime: fix bug when successful
sql UPDATE is treated as failed if there is no affected rows.
(Reported by Alexei Gradinari)
* ASTERISK-25772 - res_pjsip: Unexpected two BYE when answered
(Reported by Dmitriy Serov)
* ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
due to server timeout (Reported by Ross Beer)
* ASTERISK-26144 - Crash on loading codecs g729/g723 (Reported by
Alexei Gradinari)
* ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x
(Reported by George Joseph)
* ASTERISK-26021 - Build codecs siren7 and siren14 for Asterisk 13
(Reported by Daniel Denson)
* ASTERISK-26326 - Crash when dialing MulticastRTP channel
(Reported by George Joseph)
Improvements made in this release:
-----------------------------------
* ASTERISK-26220 - Add support for noreturn function attributes.
(Reported by Corey Farrell)
* ASTERISK-22131 - Update the make dependencies script to pull,
build, and install the correct pjproject (Reported by Matt
Jordan)
* ASTERISK-25471 - [patch]Add subscribe_context to res_pjsip
(Reported by JoshE)
* ASTERISK-26159 - res_hep: enabled by default and information
sent to default address (Reported by Ross Beer)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 13.10.0.
The release of Asterisk 13.10.0 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:
Improvements made in this release:
-----------------------------------
* ASTERISK-26088 - Investigate heavy memory utilization by
res_pjsip_pubsub (Reported by Richard Mudgett)
* ASTERISK-26011 - [patch]PJSIP: add "via_addr", "via_port",
"call_id" to contacts (Reported by Alexei Gradinari)
* ASTERISK-25994 - [patch]res_pjsip: module load priority
(Reported by Alexei Gradinari)
* ASTERISK-25931 - PJSIP: add "reg_server" to contacts. (Reported
by Alexei Gradinari)
* ASTERISK-25835 - Authentication using 'Username' field from
Digest (Reported by Ross Beer)
* ASTERISK-25930 - PJSIP: disable multi domain to improve realtime
performace (Reported by Alexei Gradinari)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26160 - pjsip: Updated->Reachable during qualify
(Reported by Matt Jordan)
* ASTERISK-26177 - func_odbc: Database handle is kept when it
should be released (Reported by Leandro Dardini)
* ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
due to server timeout (Reported by Ross Beer)
* ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to
v21_details (Reported by Corey Farrell)
* ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a
self-comparison (Reported by George Joseph)
* ASTERISK-26138 - chan_unistim: Under FreeBSD, chan_unistim
generates a compile error (Reported by George Joseph)
* ASTERISK-26128 - Alembic scripts are failing (Reported by Mark
Michelson)
* ASTERISK-26139 - test_res_pjsip_scheduler: Compile failure if
pjproject isn't installed in a system location (Reported by
George Joseph)
* ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
(Reported by Alexander Traud)
* ASTERISK-26127 - res_pjsip_session: Crash due to race condition
between res_pjsip_session unload and timer (Reported by Joshua
Colp)
* ASTERISK-26083 - ARI: Announcer channels staying around after
playback to a bridge is finished (Reported by Per Jensen)
* ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in
http.conf (Reported by Alexander Traud)
* ASTERISK-26069 - Asterisk truncates To: header, dropping the
closing '>' (Reported by Vasil Kolev)
* ASTERISK-26097 - [patch] CLI: show maximum file descriptors
(Reported by Alexander Traud)
* ASTERISK-25262 - Memory leak when a caller channel does multiple
dials and CEL is enabled (Reported by Etienne Lessard)
* ASTERISK-26092 - [Segfault] in res_rtp_asterisk.c:4268 after
Remotely bridged channels (Reported by Niklas Larsson)
* ASTERISK-26096 - res_hep: Crash when configuration file is
missing (Reported by Niklas Larsson)
* ASTERISK-26089 - Invalid security events during boot using PJSIP
Realtime (Reported by Scott Griepentrog)
* ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by
Ross Beer)
* ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B.
Davis)
* ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
against libsrtp-1.5.0 (Reported by Patrick Laimbock)
* ASTERISK-26091 - [patch] ar cru creates warning, instead use ar
cr (Reported by Alexander Traud)
* ASTERISK-26070 - ari/channels: Creating a local channel without
an originator adds all audio formats to it's capabilities
(Reported by George Joseph)
* ASTERISK-26078 - core: Memory leak in logging (Reported by
Etienne Lessard)
* ASTERISK-26065 - chan_pjsip: MWI NOTIFY contents not ordered
properly (Reported by Ross Beer)
* ASTERISK-26063 - ${PJSIP_HEADER(read,Call-ID)} does not work -
documentation needs clarification for when read/write is
possible (Reported by Private Name)
* ASTERISK-25777 - data race in threadpool (Reported by Badalian
Vyacheslav)
* ASTERISK-26038 - 'make install' doesn't seem to install OS/X
init files (Reported by Tzafrir Cohen)
* ASTERISK-26029 - parking: ast_parking_park_call should return
parking_space instead of parking_exten (Reported by Diederik de
Groot)
* ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
LAST_INSERT_ID() always returns zero. (Reported by Edwin
Vandamme)
* ASTERISK-25941 - chan_pjsip: Crash on an immediate SIP final
response (Reported by Javier Riveros )
* ASTERISK-26014 - res_sorcery_astdb: Make tolerant of unknown
fields (Reported by Joshua Colp)
* ASTERISK-24986 - keepalive INFO packages ignored by asterisk
(Reported by Ilya Trikoz)
* ASTERISK-26034 - T.38 passthrough problem behind firewall due to
early nosignal packet (Reported by George Joseph)
* ASTERISK-26030 - call cut because of double Session-Expires
header in re-invite after proxy authentication is required
(Reported by George Joseph)
* ASTERISK-25964 - Outbound registrations created via ARI/push
configuration do not clean up outbound registrations currently
in flight (Reported by Matt Jordan)
* ASTERISK-26005 - res_pjsip: Multiple SIP messages are combined
into 1 TCP packet (Reported by Ross Beer)
* ASTERISK-25352 - res_hep_rtcp correlation_id is different then
res_hep (Reported by Kevin Scott Adams)
* ASTERISK-26008 - app_followme does not delete recorded name
prompt (Reported by Tzafrir Cohen)
* ASTERISK-26007 - res_pjsip: Endpoints deleting early after
upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)
* ASTERISK-25990 - PJSIP TLS registration should respect
client_uri scheme when generating Contact URI (Reported by
Sebastian Damm)
* ASTERISK-25978 - res_pjsip_authenticator_digest: Should not use
source port in nonce verification (Reported by Mark Michelson)
* ASTERISK-25993 - pjproject: Allow bundling to not require
everything it does (Reported by Joshua Colp)
* ASTERISK-25956 - Compilation error in conditionally compiled
code in config_options.c (Reported by Chris Trobridge)
* ASTERISK-25998 - file: Crash when using nativeformats (Reported
by Joshua Colp)
* ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
(Reported by Ross Beer)
* ASTERISK-25968 - pjproject_bundled: Configure and make need to
be re-tested (Reported by George Joseph)
* ASTERISK-24463 - Voicemail email address corrupt or not sent
when message is in the process of being recorded during reload
(Reported by John Campbell)
* ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by
Dmitriy Serov)
* ASTERISK-25963 - func_odbc requires reconnect checks for stale
connections (Reported by Ross Beer)
* ASTERISK-25961 - tests/channels/SIP/sip_tls_call: Sporadic crash
when running test (Reported by Joshua Colp)
* ASTERISK-16115 - [patch] problem with ringinuse=no, queue
members receive sometimes two calls (Reported by nik600)
* ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir
only works if you manually add secret.conf yourself (Reported by
Jonathan R. Rose)
* ASTERISK-25950 - [patch]SIP channel does not send PeerStatus
events for autocreated peers (Reported by Kirill Katsnelson)
* ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions
are case sensitive to QueueName (Reported by Javier Acosta)
New Features made in this release:
-----------------------------------
* ASTERISK-25904 - PJSIP: add contact.updated event (Reported by
Alexei Gradinari)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.10.0
Thank you for your continued support of Asterisk!
rillig
jperkin
ryoon
joerg
gdt
wiz
adam
kleink
darcy
jnemeth
maya
agc