The Asterisk Development Team has announced the release of Asterisk 13.12.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
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 âcore show channeltype Surrogateâ 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-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-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!
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