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 14.1.1.
The release of Asterisk 14.1.1 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-14.1.1
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 11.24.1.
The release of Asterisk 11.24.1 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-11.24.1
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!
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!
The Asterisk Development Team has announced the release of Asterisk 11.24.0.
The release of Asterisk 11.24.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No
Symmetric Response. (Reported by Alexander Traud)
* ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
bindaddr=[::] (Reported by Jacek)
* ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so
instructed (Reported by Tzafrir Cohen)
* 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-26272 - chan_sip: File descriptors leak (UDP sockets)
(Reported by Etienne Lessard)
* ASTERISK-26288 - followme: fails to reset config items to
default values on reload (Reported by Tzafrir Cohen)
* 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-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-24841 - ConfBridge: Strange sampling rates chosen when
channels have multiple native formats (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-25706 - pbx: Abort asterisk on features reload
(handle_hint_change) (Reported by Krzysztof Trempala)
* ASTERISK-26233 - pbx: Failure to remove inconsistent extension
names (Reported by Corey Farrell)
* ASTERISK-26267 - ast_register_atexit callbacks should be run on
failed startup. (Reported by Corey Farrell)
* ASTERISK-26265 - Errors ignored from some parts of system
initialization. (Reported by Corey Farrell)
* ASTERISK-25996 - Remove "live_dangerously" requirement on
DB(read) (Reported by Andrew Nagy)
* ASTERISK-26237 - Fax is detected on regular calls. (Reported by
Richard Mudgett)
* ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
command and attended transfer handling (Reported by Ben
Smithurst)
* 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-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-25659 - res_rtp_asterisk: ECDH not negotiated causing
DTLS failure occurred on RTP instance (Reported by Edwin
Vandamme)
* ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
(Reported by Alexander Traud)
* 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-26179 - chan_sip: Second T.38 request fails (Reported
by Joshua Colp)
* ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x
(Reported by George Joseph)
Improvements made in this release:
-----------------------------------
* ASTERISK-26220 - Add support for noreturn function attributes.
(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-11.24.0
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!
AST-2016-007. Note that on Oct. 25th, this branch of Asterisk will
switch to security fixes, and one year later it will read end-of-life.
pkgsrc changes:
- don't use gethostbyname_r on NetBSD
- eliminate conflict with new hmac(1) function on NetBSd
----- 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.
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!
Upstream changes:
1.61 Tue Jun 21 21:05:12 CEST 2016
- Fixed RT#115491, remove the use of the encodings pragma, now deprecated.
- Plenty of style, test and functionality fixes contributed by Joel Maslak
and Paul Cochrane, as part of the CPAN PR Challenge. Awesome job, thanks!
- Amended the main module documentation to make it clear this module is
in maintenance mode and hasn't seen any major development work in years.
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!
The Asterisk Development Team has announced the release of Asterisk 11.23.0.
The release of Asterisk 11.23.0 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:
Bugs fixed in this release:
-----------------------------------
* 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-26130 - [patch] WebRTC: Should use latest DTLS version.
(Reported by Alexander Traud)
* 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-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-26038 - 'make install' doesn't seem to install OS/X
init files (Reported by Tzafrir Cohen)
* 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-26008 - app_followme does not delete recorded name
prompt (Reported by Tzafrir Cohen)
* 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-25917 - [patch]app_voicemail: passwordlocation=spooldir
only works if you manually add secret.conf yourself (Reported by
Jonathan R. Rose)
* ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions
are case sensitive to QueueName (Reported by Javier Acosta)
* ASTERISK-16115 - [patch] problem with ringinuse=no, queue
members receive sometimes two calls (Reported by nik600)
* ASTERISK-25934 - chan_sip should not require sipregs or
updateable sippeers table unless rt (Reported by Jaco Kroon)
* ASTERISK-25888 - Frequent segfaults in function can_ring_entry()
of app_queue.c (Reported by Sébastien Couture)
* ASTERISK-25874 - app_voicemail: Stack buffer overflow in
test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
* ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE
without adding them to the local hangupcauses via
ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)
* ASTERISK-25407 - Asterisk fails to log to multiple syslog
destinations (Reported by Elazar Broad)
* ASTERISK-25510 - [patch]Log to syslog failing (Reported by
Michael Newton)
Improvements made in this release:
-----------------------------------
* ASTERISK-25444 - [patch]Music On Hold Warning misleading
(Reported by Conrad de Wet)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
Thank you for your continued support of Asterisk!
Don't default on inet6, since the inet6 code in conserver8 depends
on some Linux-isms (ipv6 sockets can accept ipv4 packets.)
PLIST:
add some example configurations that were missing.
since the package doesn't support PJSIP (yet), all reference to
PJSIP bugs are not applicable.
----- 13.9.1
The Asterisk Development Team has announced the release of Asterisk 13.9.1.
The release of Asterisk 13.9.1 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-26007 - res_pjsip: Endpoints deleting early after
upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.1
Thank you for your continued support of Asterisk!
----- 13.9.0
The Asterisk Development Team has announced the release of Asterisk 13.9.0.
The release of Asterisk 13.9.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25963 - func_odbc requires reconnect checks for stale
connections (Reported by Ross Beer)
* ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by
Dmitriy Serov)
* ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
LAST_INSERT_ID() always returns zero. (Reported by Edwin
Vandamme)
* ASTERISK-25927 - Removed option "registertrying" is still
documented in sip.conf.sample (Reported by Etienne Lessard)
* ASTERISK-25947 - Protocol transfers to stasis applications are
missing the StasisStart with the replace_channel object.
(Reported by Richard Mudgett)
* ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
fails to get app name (Reported by John Bigelow)
* ASTERISK-24782 - StasisEnd event not present for channel that
was swapped out for another after completing attended transfer
(Reported by John Bigelow)
* ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed
ConnectedLine information (Reported by George Joseph)
* ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
thread (Reported by Joshua Colp)
* ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
not raised (Reported by Joshua Colp)
* ASTERISK-25934 - chan_sip should not require sipregs or
updateable sippeers table unless rt (Reported by Jaco Kroon)
* ASTERISK-25888 - Frequent segfaults in function can_ring_entry()
of app_queue.c (Reported by Sébastien Couture)
* ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets
exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George
Joseph)
* ASTERISK-25707 - Long contact URIs or hostnames can crash
pjproject/Asterisk under certain conditions (Reported by George
Joseph)
* ASTERISK-25123 - Bracketed IPv6 Contact header parameter
unparsable with Asterisk/PJSIP (Reported by Anthony Messina)
* ASTERISK-25874 - app_voicemail: Stack buffer overflow in
test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
* ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE
without adding them to the local hangupcauses via
ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)
* ASTERISK-25885 - res_pjsip: Race condition between adding
contact and automatic expiration (Reported by Joshua Colp)
* ASTERISK-25910 - pjproject: Via headers are not parsed when
"received" contains an IPv6 address (Reported by George Joseph)
* ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails
(Reported by Harley Peters)
* ASTERISK-25894 - [patch] webrtc video broken due to missing
marker bits in RTP streams (Reported by Jacek Konieczny)
* ASTERISK-25854 - No audio after HOLD/RESUME - incorrect
a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)
* ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error,
cannot find -lasteriskpj (Reported by Hans van Eijsden)
* ASTERISK-25882 - ARI: Crash can occur due to race condition when
attempting to operate on a hung up channel (Part 2) (Reported by
Richard Mudgett)
* ASTERISK-25867 - [patch] Video delay on app_echo (Reported by
Jacek Konieczny)
* ASTERISK-24605 - res_parking option parkeddynamic does not work
with the core Features 'parkcall' (DTMF initiated parking)
(Reported by Philip Correia)
* ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
(Reported by Ross Beer)
* ASTERISK-24596 - Unclear how to use Park application with
res_parking 'parkeddynamic' enabled. Documentation? (Reported by
Philip Correia)
* ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
possible codecs configured for peer as opposed to intersection
of configured codecs and offered codecs (Reported by Taylor
Hawkes)
* ASTERISK-25825 - Crashes during shutdown when running CLI
commands (Reported by Mark Michelson)
* ASTERISK-25407 - Asterisk fails to log to multiple syslog
destinations (Reported by Elazar Broad)
* ASTERISK-25510 - [patch]Log to syslog failing (Reported by
Michael Newton)
* ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
data corruption (Reported by Gianluca Merlo)
Improvements made in this release:
-----------------------------------
* ASTERISK-25865 - Message-Account Missing From PJSIP MWI
(Reported by Ross Beer)
* ASTERISK-25444 - [patch]Music On Hold Warning misleading
(Reported by Conrad de Wet)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0
Thank you for your continued support of Asterisk!
also contains fixes for AST-2016-004 and AST-2016-005. However,
those issues only affected the pjsip module. Since Asterisk in
pkgsrc doesn't (yet) use pjsip, it wasn't affected.
----- 13.8.2
The Asterisk Development Team has announced the release of Asterisk 13.8.2.
The release of Asterisk 13.8.2 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-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
not raised (Reported by Joshua Colp)
* ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
thread (Reported by Joshua Colp)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.2
Thank you for your continued support of Asterisk!
----- 13.8.0
The Asterisk Development Team has announced the release of Asterisk 13.8.0.
The release of Asterisk 13.8.0 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-24919 - res_pjsip_config_wizard: Ability to write
contents to file (Reported by Ray Crumrine)
* ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
Journo)
* ASTERISK-25480 - [patch]Add field PauseReason on
QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25849 - chan_pjsip: transfers with direct media
sometimes drops audio (Reported by Kevin Harwell)
* ASTERISK-25113 - install_prereq in Debian 8 without "standard
system utilities" (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
(Reported by Sergio Medina Toledo)
* ASTERISK-25023 - Deadlock in chan_sip in
update_provisional_keepalive (Reported by Arnd Schmitter)
* ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
channel (Reported by Filip Frank)
* ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
separating multiple AORs (Reported by Mateusz Kowalski)
* ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
Stasis application. (Reported by Javier Riveros )
* ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
Bright)
* ASTERISK-25582 - Testsuite: Reactor timeout error in
tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
Jordan)
* ASTERISK-25811 - Unable to delete object from sorcery cache
(Reported by Ross Beer)
* ASTERISK-25800 - [patch] Calculate talktime when is first call
answered (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
PJSIP requirement (Reported by Gergely Dömsödi)
* ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
when calling from Gosub (Reported by Jacques Peacock)
* ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
OutboundSubscriptionDetail ami action (Reported by Kevin
Harwell)
* ASTERISK-25721 - [patch] res_phoneprov: memory leak and
heap-use-after-free (Reported by Badalian Vyacheslav)
* ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
returns garbage (Reported by Etienne Lessard)
* ASTERISK-25751 - res_pjsip: Support
pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
* ASTERISK-25606 - Core dump when using transports in sorcery
(Reported by Martin Moučka)
* ASTERISK-20987 - non-admin users, who join muted conference are
not being muted (Reported by hristo)
* ASTERISK-25737 - res_pjsip_outbound_registration: line option
not in Alembic (Reported by Joshua Colp)
* ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
udptl_rx_packet cause ast_frdup crash (Reported by Walter
Doekes)
* ASTERISK-25742 - Secondary IFP Packets can result in accessing
uninitialized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
Vulnerability - Investigate vulnerability of HTTP server
(Reported by Alex A. Welzl)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
non-default timert1 (Reported by Alexander Traud)
* ASTERISK-25702 - PjSip realtime DB and Cache Errors since
upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
Nic Colledge)
* ASTERISK-25730 - build: make uninstall after make distclean
tries to remove root (Reported by George Joseph)
* ASTERISK-25725 - core: Incorrect XML documentation may result in
weird behavior (Reported by Joshua Colp)
* ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
sip_sipredirect (Reported by Badalian Vyacheslav)
* ASTERISK-25709 - ARI: Crash can occur due to race condition when
attempting to operate on a hung up channel (Reported by Mark
Michelson)
* ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
by Badalian Vyacheslav)
* ASTERISK-25685 - infrastructure: Run alembic in Jenkins build
script (Reported by Joshua Colp)
* ASTERISK-25712 - Second call to already-on-call phone and
Asterisk sends "Ready" (Reported by Richard Mudgett)
* ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
(Reported by Badalian Vyacheslav)
* ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report
incorrect values (Reported by Gianluca Merlo)
* ASTERISK-25611 - core: threadpool thread_timeout_thrash unit
test sporadically failing (Reported by Joshua Colp)
* ASTERISK-24097 - Documentation - CHANNEL function help text
missing 'linkedid' argument (Reported by Steven T. Wheeler)
* ASTERISK-25700 - main/config: Clean config maps on shutdown.
(Reported by Corey Farrell)
* ASTERISK-25696 - bridge_basic: don't cache xferfailsound during
a transfer (Reported by Kevin Harwell)
* ASTERISK-25697 - bridge_basic: don't play an attended transfer
fail sound after target hangs up (Reported by Kevin Harwell)
* ASTERISK-25683 - res_ari: Asterisk fails to start if compiled
with MALLOC_DEBUG (Reported by yaron nahum)
* ASTERISK-25686 - PJSIP: qualify_timeout is a double, database
schema is an integer (Reported by Marcelo Terres)
* ASTERISK-25690 - Hanging up when executing connected line sub
does not cause hangup (Reported by Joshua Colp)
* ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
reload' cause a crash (Reported by Sean Bright)
* ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP
address when multihomed (Reported by Olivier Krief)
* ASTERISK-25637 - Multi homed server using wrong IP (Reported by
Daniel Journo)
* ASTERISK-25394 - pbx: Incorrect device and presence state when
changing hint details (Reported by Joshua Colp)
* ASTERISK-25640 - pbx: Deadlock on features reload and state
change hint. (Reported by Krzysztof Trempala)
* ASTERISK-25681 - devicestate: Engine thread is not shut down
(Reported by Corey Farrell)
* ASTERISK-25680 - manager: manager_channelvars is not cleaned at
shutdown (Reported by Corey Farrell)
* ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
Corey Farrell)
* ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by
Daniel Journo)
* ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
by Corey Farrell)
* ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
Farrell)
* ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by
Mark Michelson)
* ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
(Reported by Corey Farrell)
* ASTERISK-25647 - bug of cel_radius.c: wrong point of
ADD_VENDOR_CODE (Reported by Aaron An)
* ASTERISK-25317 - asterisk sends too many stun requests (Reported
by Stefan Engström)
* ASTERISK-25137 - endpoint stasis messages are delivered twice
(Reported by Vitezslav Novy)
* ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are
sent for every status change (Reported by George Joseph)
* ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on
transfer initiated channel (Reported by Dmitry Melekhov)
* ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
Brandon)
* ASTERISK-25442 - using realtime (mysql) queue members are never
updated in wait_our_turn function (app_queue.c) (Reported by
Carlos Oliva)
* ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
caching (Reported by Joshua Colp)
* ASTERISK-25601 - json: Audit reference usage and thread safety
(Reported by Joshua Colp)
* ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
sungtae kim)
Improvements made in this release:
-----------------------------------
* ASTERISK-25495 - [patch] Prevent old-update packages on
repository Debian systems (Reported by Rodrigo Ramirez
Norambuena)
* ASTERISK-25846 - Gracefully deal with Absent Stasis Apps
(Reported by Andrew Nagy)
* ASTERISK-25791 - res_pjsip_caller_id: Lack of support for
Anonymous <anonymous at anonymous.invalid> (Reported by Anthony
Messina)
* ASTERISK-24813 - asterisk.c: #if statement in listener()
confuses code folding editors (Reported by Corey Farrell)
* ASTERISK-25767 - [patch] Add check to configure for sanitizes
(Reported by Badalian Vyacheslav)
* ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
core set (Reported by Rusty Newton)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.0
Thank you for your continued support of Asterisk!
----- 11.22.0
The Asterisk Development Team has announced the release of Asterisk 11.22.0.
The release of Asterisk 11.22.0 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:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
data corruption (Reported by Gianluca Merlo)
* ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
channel (Reported by Filip Frank)
* ASTERISK-25800 - [patch] Calculate talktime when is first call
answered (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
returns garbage (Reported by Etienne Lessard)
* ASTERISK-20987 - non-admin users, who join muted conference are
not being muted (Reported by hristo)
* ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
Vulnerability - Investigate vulnerability of HTTP server
(Reported by Alex A. Welzl)
* ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
udptl_rx_packet cause ast_frdup crash (Reported by Walter
Doekes)
* ASTERISK-25742 - Secondary IFP Packets can result in accessing
uninitialized pointers and a crash (Reported by Torrey Searle)
* ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
non-default timert1 (Reported by Alexander Traud)
* ASTERISK-25730 - build: make uninstall after make distclean
tries to remove root (Reported by George Joseph)
* ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
sip_sipredirect (Reported by Badalian Vyacheslav)
* ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
by Badalian Vyacheslav)
* ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
(Reported by Badalian Vyacheslav)
* ASTERISK-25701 - core: Endless loop in "core show
taskprocessors" (Reported by ibercom)
* ASTERISK-25700 - main/config: Clean config maps on shutdown.
(Reported by Corey Farrell)
* ASTERISK-25690 - Hanging up when executing connected line sub
does not cause hangup (Reported by Joshua Colp)
* ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
reload' cause a crash (Reported by Sean Bright)
* ASTERISK-25394 - pbx: Incorrect device and presence state when
changing hint details (Reported by Joshua Colp)
* ASTERISK-25640 - pbx: Deadlock on features reload and state
change hint. (Reported by Krzysztof Trempala)
* ASTERISK-25681 - devicestate: Engine thread is not shut down
(Reported by Corey Farrell)
* ASTERISK-25680 - manager: manager_channelvars is not cleaned at
shutdown (Reported by Corey Farrell)
* ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
Corey Farrell)
* ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
by Corey Farrell)
* ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
Farrell)
* ASTERISK-25647 - bug of cel_radius.c: wrong point of
ADD_VENDOR_CODE (Reported by Aaron An)
* ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
Brandon)
* ASTERISK-25442 - using realtime (mysql) queue members are never
updated in wait_our_turn function (app_queue.c) (Reported by
Carlos Oliva)
* ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
sungtae kim)
Improvements made in this release:
-----------------------------------
* ASTERISK-24813 - asterisk.c: #if statement in listener()
confuses code folding editors (Reported by Corey Farrell)
* ASTERISK-25767 - [patch] Add check to configure for sanitizes
(Reported by Badalian Vyacheslav)
* ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
core set (Reported by Rusty Newton)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
Thank you for your continued support of Asterisk!
----- 11.21.2
The Asterisk Development Team has announced the release of Asterisk 11.21.2.
The release of Asterisk 11.21.2 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-25770 - Check for OpenSSL defines before trying to use
them. (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-11.21.2
Thank you for your continued support of Asterisk!
Add patches to work around gcc inline mess.
Note: this package should be updated and the PR contains an update, but
I don't want to do that when I can't compile it (whereas simple mechanical
patches are much less likely to go astray...)
version 8.2.1 (Jun 2, 2015):
- added TCP keepalives between client and server - TCP-based consoles
already had the code - this was mostly an oversight
- patch for SEGV and task execution - patch by Artem Savkov
<asavkov@redhat.com>
- expanded break sequences from [1-9] to [1-9a-z] - based on patch by
Artem Savkov <asavkov@redhat.com>
pkgsrc:
options.mk:
add support inet6
The way the conserver is coded, you get inet6 or you get uds
patches/patch-conserver_readcfg.c:
new patch, fixing a setproctitle() bug with inet6.
This needs to be sent upstream.
Makefile:
install sample configurations
Change maintainer to pkgsrc-users@netbsd.org
old maintainer address bounced.
drop ``--with-regex'' option, it's no longer supported.
This release brings:
- support the latest libSystem
- compatibility with Gtk+ 3
- improved hardware compatibility (GSM)
- improved handling of SMS and USSD messages
- new "console" plug-in
- improved "profiles" plug-in
- further improvements to the user interface
As its name suggests, picocom is a minimal dumb-terminal emulation
program. It is, in principle, very much like minicom, only it's
"pico" instead of "mini"!
It was designed to serve as a simple, manual, modem configuration,
testing, and debugging tool. It has also served (quite well) as a
low-tech serial communications program to allow access to all types
of devices that provide serial consoles. It could also prove useful
in many other similar tasks.
Changes:
2.5
===
* Compatibility with Gammu >= 1.36.7
2.4
===
* Fixed possible crash when initializing SMSD with invalid parameters.
* Fixed crash on handling diverts on certain architectures.
Changes:
20160203 - 1.37.0
[-] * Improved compatibility with ZTE MF190.
[-] * Improved compatibility with Huawei E1750.
[-] * Improved compatibility with Huawei E1752.
[-] * Increased detail of reported errors from SMSD.
20151208 - 1.36.8
[-] * Changed default value for ReceiveFrequency.
[-] * Fixed compatibility for PostgreSQL.
[-] * Fixed build failure with all disabled SMSD backends.
[-] * Documentation improvements.
[-] * Fixed mixing C++ code with SMSD.
20151129 - 1.36.7
[-] * Support devices which do not report full network status.
[-] * Disable Huawei unsolicited messages on startup.
[-] * Various improvements for Huawei modems.
[-] * Fixed compilation on Windows.
[-] * Fixed regression with Siemens AX75.
[-] * Improved decoding of USSD responses.
[-] * Properly decode emojis to console or files backend.
[+] * Added support for proxying the connection through arbitrary command.
[+] * SMSD now tracks retries count per message.
20151012 - 1.36.6
[-] * Fixed installation of bash-completion script.
[-] * Fixed timezone manipulation in SMSD.
[-] * Documentation improvements.
[-] * Fixed licensing of helper/win32-dirent.*.
[*] * Increased default speed for AT connection to 115200.
[*] * Improve AT module initialization.
20150826 - 1.36.5
[-] * Properly use timezones with SQLite in SMSD.
[-] * Improve support for Huawei E1752.
[-] * Fixed compilation on distros with old Glib.
