improvements.
pkgsrc change: adapt to new res_resolver_unbound module.
The Asterisk Development Team has announced the release of Asterisk 14.2.0.
The release of Asterisk 14.2.0 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-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)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26608 - Compile and link failures on OpenBSD (Reported
by snuffy)
* ASTERISK-26520 - codec_opus: Generated fmtp line has no content
(Reported by scgm11)
* ASTERISK-26605 - codec_opus: Spammed warning when Opus
negotiated but codec_opus not loaded. (Reported by Richard
Mudgett)
* ASTERISK-26516 - pjsip: Memory corruption with possible memory
leak. (Reported by Richard Mudgett)
* ASTERISK-26556 - manager: AMI version report same in Ast 13 &
14, despite Ast 14 syntax changes (Reported by Michelle Dupuis)
* ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
manipulation through agi (Reported by Morten Tryfoss)
* 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-26571 - res_pjsip: Resolution incorrect when explicit
IPv6 transport configured (Reported by Joshua Colp)
* ASTERISK-26468 - ari: Bridge events stop working after this
sequence of ARI calls (Reported by Daniele Pallastrelli)
* ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by
Dmitry Melekhov)
* ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11
regressions (Reported by Matt Jordan)
* ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George
Joseph)
* ASTERISK-26509 - A few non-critical deprecation warnings when
building on Ubuntu 16.10 (Reported by Jonathan Harris)
* ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
calls after 2 minutes - rtptimeout behaving badly - regression
(Reported by Michael Keuter)
* ASTERISK-26549 - app_dial: When PickupChan() is used some
channels may have incorrect device state (Reported by Joshua
Colp)
* ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP
Media Attributes When SLIN48 Codec Is Used (Reported by Frankie
Chin)
* ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic
payload types. (Reported by Alexander Traud)
* ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash when
publishing, in publisher_client_send at
res_pjsip_outbound_publish.c (Reported by Matt Krokosz)
* 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-26541 - res_pjsip_sdp_rtp: Restrict number of formats
to maximum (Reported by Joshua Colp)
* 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-26513 - tests/channels/pjsip/qualify/auth: Crashing
enough to be a nuisance (Reported by Joshua Colp)
* ASTERISK-26514 - Super Awesome Company: Don't specify transport
in pjsip.conf (Reported by Rusty Newton)
* ASTERISK-26510 - pjproject_bundled uses the --strip-components
option of tar which isn't supported in older versions (Reported
by George Joseph)
* ASTERISK-22480 - Embedded pjproject: build.mak contains
hardcoded full path to version.mak (Reported by Matt Jordan)
* ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change
(Reported by Bill Brigden)
* ASTERISK-26503 - app_voicemail: Asterisk crashes when
MailboxExists is used (Reported by Doug Lytle)
* ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can
cause audio loss and wonkiness (Reported by Andreas Wetzel)
* ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack)
installations. (Reported by Alexander Traud)
* ASTERISK-26482 - [patch] chan_pjsip: segfault on already
disconnected session (Reported by Alexei Gradinari)
* ASTERISK-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-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-26462 - [patch] app_queue: While using queues with
realtime, setting back to an empty context doesn't stop the exit
key usage (Reported by Leandro Dardini)
* ASTERISK-26439 - chan_rtp: Crash when originating (Reported by
Kayode)
* ASTERISK-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)
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-26492 - ARI: Add ability to specify channel variables
on websocket events (Reported by Mark Michelson)
* ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing
events (Reported by Matt Jordan)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.2.0
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 14.1.0.
The release of Asterisk 14.1.0 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-26410 - core: Asterisk 14 doesn't show the header in
the console or verbose when starting (Reported by Dan Jenkins)
* ASTERISK-24311 - Populating database via Alembic fails when
using same database for multiple schema sets (Reported by Dafi
Ni)
* ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No
Symmetric Response. (Reported by Alexander Traud)
* ASTERISK-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-26391 - Consoles do not display verbose logger messages
even when requested. (Reported by Marcelo Terres)
* ASTERISK-26263 - SQL error when using realtime and registering
extension / inserting into ps_contacts (Reported by Jeppe Ryskov
Larsen)
* ASTERISK-26365 - rtp: Offer with multiple payloads for same
codec is incorrectly handled (Reported by Joshua Colp)
* ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
by Aaron Hamstra)
* 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-26364 - res_pjsip: Don't assume a request will have
target addresses (Reported by Joshua Colp)
* ASTERISK-26360 - app_queue: "queue show" output gets "failed to
extend from 240 to 327" msgs. (Reported by Richard Mudgett)
* ASTERISK-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-26341 - ARI: Stopping a media playlist only stops the
current media URI being played back, and not the whole list
(Reported by Matt Jordan)
* ASTERISK-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-26269 - res_pjsip: Wrong state for aors without
registered contacts after startup (Reported by nappsoft)
* 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-26299 - app_queue: Queue application sometimes stops
calling members with Local interface (Reported by Etienne
Lessard)
* ASTERISK-26279 - pjproject-bundled: Fails to compile on Debian
6 (Reported by George Joseph)
* ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not
cleaning up properly (Reported by Alexander Traud)
* ASTERISK-26203 - res_fax: Deadlock when using
FAXOPT(gateway)=yes with Local channels (Reported by Etienne
Lessard)
* ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking
inversion in T.38 query option with features bridging code
(Reported by David Brillert)
* ASTERISK-22732 - Deadlock potential in res_fax and CCSS with
local channels. (Reported by Richard Mudgett)
* ASTERISK-26288 - followme: fails to reset config items to
default values on reload (Reported by Tzafrir Cohen)
* ASTERISK-22374 - Finish mapping the sip.conf parameters to
res_sip.conf parameters (Reported by Matt Jordan)
* ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
abelbeck)
* ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include
annexb=no attribute. (Reported by Ali Ghavidel)
* ASTERISK-25472 - Swagger scripts are not replacing format
variable in file brief (Reported by Corey Farrell)
* ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but
it's not mandatory to compile it (Reported by József Dudás)
* ASTERISK-26305 - Asterisk 14: Two resolver unbound testsuite
tests fail (Reported by Richard Mudgett)
* ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities
not detected in PJProject. (Reported by Alexander Traud)
* ASTERISK-25492 - ARI: Path parameters are case sensitive
(Reported by Joshua Colp)
* ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via
chan_pjsip (Reported by Ross Beer)
* ASTERISK-26233 - pbx: Failure to remove inconsistent extension
names (Reported by Corey Farrell)
* ASTERISK-26246 - Security: Privilege escalation by AMI adding
dialplan extensions. (Reported by Richard Mudgett)
* ASTERISK-26267 - ast_register_atexit callbacks should be run on
failed startup. (Reported by Corey Farrell)
* ASTERISK-26241 - res_pjsip: When using compact headers, rpid
and pai are incorrectly generated (Reported by George Joseph)
* ASTERISK-25797 - app_queue: Crash when calling a queue with a
member with a forward to an nonexistent extension (Reported by
Etienne Lessard)
* ASTERISK-26239 - res_pjsip_logger: An empty global/debug option
is treated as a "match all" hostname (Reported by George Joseph)
* ASTERISK-26238 - res_pjsip: Empty global default_from_user
causes crash (Reported by Joshua Colp)
* ASTERISK-26268 - alembic: 'auth_username' not in PJSIP
'identify_by' enum (Reported by Joshua Colp)
* ASTERISK-26253 - sdp_srtp: libsrtp now a required dependency,
shouldn't be (Reported by Ben Merrills)
* ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade +
indicate (Reported by Ross Beer)
* ASTERISK-26183 - alembic: error when using sqlalchemy version
1.1.0b2 (Reported by Kevin Harwell)
* ASTERISK-26283 - res_resolver_unbound: fails configure on older
Ubuntu and CentOS (Reported by George Joseph)
* ASTERISK-26280 - DNS lookups can block channel media paths
(Reported by Mark Michelson)
* ASTERISK-26278 - asterisk.h should produce a reasonable error
for external modules that fail to define AST_MODULE_SELF_SYM.
(Reported by Corey Farrell)
* ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a
similar treatment for module unloading as
res_pjsip_outbound_registration.c (Reported by Richard Mudgett)
* ASTERISK-26265 - Errors ignored from some parts of system
initialization. (Reported by Corey Farrell)
* ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex
for get all (Reported by Dmitry)
* ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets
with IP6 (Reported by Alexander Traud)
* ASTERISK-25996 - Remove "live_dangerously" requirement on
DB(read) (Reported by Andrew Nagy)
* ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1:
"libasteriskpj.so: undefined reference to..." (Reported by Hans
van Eijsden)
* ASTERISK-26237 - Fax is detected on regular calls. (Reported by
Richard Mudgett)
Improvements made in this release:
-----------------------------------
* ASTERISK-26409 - codec_opus: Update Asterisk to support the
l
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-14.1.0
Thank you for your continued support of Asterisk!
and run, but not a lot of functional testing. This does not have
the new PJSIP, which will be coming in a followup commit. This
also does not have the patches for compiling with Clang. For
upgrading instructions, please see:
https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+14
----- 14.0.0 -----
The Asterisk Development Team is pleased to announce the release of
Asterisk 14.0.0.
Asterisk 14 is the next major release series of Asterisk. It is a
Standard Support release, similar to Asterisk 12. For more information
about support time lines for Asterisk releases, see the Asterisk
versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
For important information regarding upgrading to Asterisk 14, please
see the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+14
A short list of new features includes:
* A complete overhaul of the core DNS support in Asterisk, including
implementing full NAPTR and SRV support in the PJSIP stack via the
libunbound library.
* The ability to publish extension state to a SIP Subscription server,
such as Kamailio. This includes the ability to automatically generate
a hint in the dialplan based on device state changes using the new
autohint setting.
* Playback of media from a remote HTTP server via a URI is now supported
by all dialplan applications and AGI. Media retrieved using a URI is
cached in a media cache and re-used when possible.
* When using ARI to manipulate media on a resource, a list of media
resources can now be supplied. The media resources will be played back
sequentially in the order that they are provided.
* Channels created via ARI can now be created and handed off to Stasis
for external control prior to performing the outbound dial. This
enables applications to set additional state on the channel prior to
dialing, as well as enabling certain early media scenarios.
And much more!
More information about the new features can be found on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Documentation
A full list of all new features can also be found in the CHANGES file:
https://github.com/asterisk/asterisk/blob/14/CHANGES
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.0.0
Thank you for your continued support of Asterisk!
----- 14.0.1 -----
The Asterisk Development Team has announced the release of Asterisk 14.0.1.
The release of Asterisk 14.0.1 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:
Improvements made in this release:
-----------------------------------
* ASTERISK-26409 - codec_opus: Update Asterisk to support the
translation codec. (Reported by Kevin Harwell)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.1
Thank you for your continued support of Asterisk!
----- 14.0.2 -----
The Asterisk Development Team has announced the release of Asterisk 14.0.2.
The release of Asterisk 14.0.2 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26410 - core: Asterisk 14 doesn't show the header in
the console or verbose when starting (Reported by Dan Jenkins)
* ASTERISK-26426 - format_ogg_opus: remove from source (Reported
by Kevin Harwell)
* ASTERISK-26425 - download_externals: ignore xmlstarlet return
code for optional element (Reported by Kevin Harwell)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.2
Thank you for your continued support of Asterisk!
jnemeth