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[asteriskteam at digium...]

The Asterisk Development Team has announced the release of Asterisk 13.4.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.4.0 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-24922 - ARI: Add the ability to intercept hold and
raise an event (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
* ASTERISK-25112 - Logger: Configuration settings are not reset to
default during reload. (Reported by Corey Farrell)
* ASTERISK-24944 - main/audiohook.c change prevents G722 call
recording (Reported by Ronald Raikes)
* ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
or more digits (Reported by Makoto Dei)
* ASTERISK-25086 - [patch]PJSIP crashes if endpoint missing in
Dial() (Reported by snuffy)
* ASTERISK-25089 - res_pjsip_config_wizard: Variable specified in
templates aren't being processed correctly (Reported by George
Joseph)
* ASTERISK-25090 - CLI core show channel truncates cdr variables
(Reported by snuffy)
* ASTERISK-25085 - [patch]Potential crash after unload of
func_periodic_hook or test_message (Reported by Corey Farrell)
* ASTERISK-25083 - Message.c: Message channel becomes saturated
with frames leading to spammy log messages (Reported by Jonathan
Rose)
* ASTERISK-25082 - Asterisk deletes message after doing a playback
of an INBOX message using ast_vm_play when the Old folder is
full for that mailbox. (Reported by Jonathan Rose)
* ASTERISK-25041 - [patch]Broken column type checking in
res_config_mysql addon (Reported by Alexandre Fournier)
* ASTERISK-21893 - Segfault after call hangup, in
ast_channel_hangupcause_set, at channel_internal_api.c (Reported
by Alexandr Gordeev)
* ASTERISK-25074 - Regression: Recent clang-related change broke
cross compiling of Asterisk (Reported by Sebastian Kemper)
* ASTERISK-25042 - asterisk.conf options override command-line
options. (Reported by Corey Farrell)
* ASTERISK-24442 - Outgoing call files don't work properly when
set in the future (Reported by tootai)
* ASTERISK-25057 - res_pjsip_pubsub: Crash in send_notify due to
invalid root pointer in sub_tree (Reported by Matt Jordan)
* ASTERISK-24938 - ARI Snoop Channel results in excessive
escalating CPU usage (Reported by George Ladoff)
* ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
ignore ISDN RESTART requests. (Reported by Richard Mudgett)
* ASTERISK-25003 - Asterisk crashes on attended transfer (using
feature) (Reported by Artem Volodin)
* ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
contain waiting time (Reported by Etienne Lessard)
* ASTERISK-25027 - Build System: Many ARI modules are missing
dependencies. (Reported by Corey Farrell)
* ASTERISK-25061 - pbx_config: Register manager actions with
module version of macro. (Reported by Corey Farrell)
* ASTERISK-25025 - Periodic crashes (in
ast_channel_snapshot_create at stasis_channels.c) with Certified
Asterisk 13. (Reported by Chet Stevens)
* ASTERISK-25053 - Unit test category /main/presence missing
trailing slash. (Reported by Corey Farrell)
* ASTERISK-22708 - res_odbc.conf negative_connection_cache option
not respected, failover between DSNs doesn't work (Reported by
JoshE)
* ASTERISK-25054 - Formats interface's cannot be unregistered,
needs to hold modules until shutdown. (Reported by Corey
Farrell)
* ASTERISK-24896 - [patch] Using force black background leads to
colours not being reset (Reported by dant)
* ASTERISK-25033 - Asterisk 13 (branch head) won't compile without
PJSip (Reported by Peter Whisker)
* ASTERISK-25028 - Build System: Unneeded defines in
asterisk/buildopts.h (Reported by Corey Farrell)
* ASTERISK-25048 - Astobj2: Initialization order wrong when both
refdebug and AO2_DEBUG are both enabled. (Reported by Corey
Farrell)
* ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls with
cause code 44 after some time. (Reported by Denis Alberto
Martinez)
* ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-25037 - res_pjsip_outbound_registration: Potential
crash in off-nominal failure case when sending message (Reported
by Joshua Colp)
* ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
(Reported by Steve Davies)
* ASTERISK-22790 - check_modem_rate() may return incorrect rate
for V.27 (Reported by not here)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
* ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
which is disallowed in res_fax's check_modem_rate (Reported by
Matt Jordan)
* ASTERISK-24996 - chan_pjsip: Creating Channel Causes Asterisk to
Crash When Duplicate AOR Sections Exist in pjsip.conf (Reported
by Ashley Sanders)
* ASTERISK-25020 - Mismatched response to outgoing REGISTER
request (Reported by Mark Michelson)
* ASTERISK-25018 - pjsip show endpoints crashes asterisk when
qualified aors present (Reported by Ivan Poddubny)
* ASTERISK-24749 - ConfBridge: Wrong language on playing
conf-hasjoin and conf-hasleft when played to bridge (Reported by
Philippe Bolduc)
* ASTERISK-24845 - pjsip send notify not working with Cisco phone
(Reported by Carl Fortin)
* ASTERISK-25004 - Crash in authenticated reinvite after
originated T.38 FAX (Reported by Mark Michelson)
* ASTERISK-24999 - PJSIP crashes with malformed contact line
(Reported by snuffy)
* ASTERISK-24998 - res_corosync: res_corosync tries to load even
if res_corosync.conf is missing (Reported by George Joseph)
* ASTERISK-24997 - Astobj2: Some callers of __adjust_lock do not
pre-check the object (Reported by Corey Farrell)
* ASTERISK-24982 - res_pjsip_mwi: Unsolicited MWI NOTIFY only sent
on mailbox changes (Reported by Joshua Colp)
* ASTERISK-24991 - Check for ao2_alloc failure in
__ast_channel_internal_alloc (Reported by Corey Farrell)
* ASTERISK-24895 - After hangup on the side of the ISDN network no
HangupRequest event comes for the dahdi channel. (Reported by
Andrew Zherdin)
* ASTERISK-24977 - Contacts that don't use qualify are being
marked as unavailable (Reported by George Joseph)
* ASTERISK-24774 - Segfault in ast_context_destroy with
extensions.ael and extensions.conf (Reported by Corey Farrell)
* ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when
channels have multiple native formats (Reported by Matt Jordan)
* ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
to Fail (Reported by Ashley Sanders)
* ASTERISK-24958 - Forwarding loop detection inhibits certain
desirable scenarios (Reported by Mark Michelson)
* ASTERISK-24863 - res_pjsip: No endpoint events raised via AMI
when contacts cannot be reached/qualified (Reported by Dmitriy
Serov)
* ASTERISK-24869 - Asterisk segfaults on DAHDI attended transfer
due to application (appl) being NULL on unbridged channel
(Reported by viniciusfontes)
* ASTERISK-24970 - Crash in res_pjsip_pubsub handling of failed
notify (Reported by Scott Griepentrog)
* ASTERISK-24959 - [patch]CLI command cdr show pgsql status
(Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24954 - Git migration: Asterisk version numbers are
incompatible with the Test Suite (Reported by Matt Jordan)
* ASTERISK-17608 - func_aes.so cannot be loaded if res_crypto /
openssl not compiled (Reported by Warren Selby)
* ASTERISK-24928 - [patch]t38_udptl_maxdatagram in pjsip.conf not
honored (Reported by Juergen Spies)
* ASTERISK-24835 - Early Media Not working with Chan SIP and
Asterisk 13 (Reported by Andrew Nagy)
* ASTERISK-21777 - Asterisk tries to transcode video instead of
audio (Reported by Nick Ruggles)
* ASTERISK-24380 - core: Native formats are set to h264 with
certain audio/video codec configuration, resulting in path
translation WARNINGs (Reported by Matt Jordan)
* ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
into account (Reported by Frederic Van Espen)
* ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
short (Reported by Y Ateya)
* ASTERISK-24935 - res_pjsip_phoneprov_provider: Fix leaked
OBJ_MULTIPLE iterator. (Reported by Corey Farrell)
* ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
(Reported by Vadim)
* ASTERISK-24933 - T38 fails negotiation (Reported by Jonathan
Rose)
* ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
byte prefix bug (Reported by Matt Jordan)
* ASTERISK-21211 - chan_iax2 - unprotected access of
iaxs[peer->callno] potentially results in segfault (Reported by
Jaco Kroon)
* ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
(Reported by Christoph Timm)
* ASTERISK-24782 - StasisEnd event not present for channel that
was swapped out for another after completing attended transfer
(Reported by John Bigelow)
* ASTERISK-24910 - "timer=no" and "timer=required" settings in
pjsip.conf fail (Reported by Ray Crumrine)
* ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
(Reported by Jeffrey C. Ollie)
* ASTERISK-24914 - Division by zero in file.c when playback of
voicemail with video as h264 (Reported by Marcello Ceschia)
* ASTERISK-24899 - Parking fall-through behavior different in 13
(Reported by Malcolm Davenport)
* ASTERISK-24937 - [patch]res_pjsip_messaging: Messages may be
sent out of order (Reported by Mark Michelson)
* ASTERISK-24920 - Asterisk handles duplicate SIP requests as if
they were each a new request (Reported by Mark Michelson)
* ASTERISK-24857 - [patch] "timing test", pjsip incoming/outgoing
calls, voicemail prompts and recordings all fail when using the
kqueue timer source on FreeBSD 10.x (Reported by Justin T.
Gibbs)
* ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
detection in ast_malloc (Reported by Timo Teräs)
* ASTERISK-24142 - CCSS: crash during shutdown due to device
lookup in destroyed container (Reported by David Brillert)
* ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
core restart now (Reported by Peter Katzmann)
* ASTERISK-24805 - [patch] - ASAN: Race condition
(heap-use-after-free) on asterisk closing (Reported by Badalian
Vyacheslav)
* ASTERISK-24881 - ast_register_atexit should only be used when
absolutely needed (Reported by Corey Farrell)
* ASTERISK-24731 - res_pjsip_session cannot be unloaded (Reported
by Corey Farrell)
* ASTERISK-24864 - app_confbridge: file playback blocks dtmf
(Reported by Kevin Harwell)
* ASTERISK-14233 - [patch] Buddies are always auto-registered when
processing the roster (Reported by Simon Arlott)
* ASTERISK-24780 - [patch] - Buddies are always auto-registered
when processing the roster (Reported by Simon Arlott)
* ASTERISK-24781 - PJSIP: Unnecessary 180 Ringing messages sent
with undesireabe consequences. (Reported by Richard Mudgett)

Improvements made in this release:
-----------------------------------
* ASTERISK-25044 - sorcery: Add ability to insert a new wizard
into an object type's list (Reported by George Joseph)
* ASTERISK-24892 - Super Awesome Company sound prompts (Reported
by Rusty Newton)
* ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
Hjelm)
* ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
(Reported by Alexander Traud)
* ASTERISK-25045 - vector: Add new capabilities and unit tests
(Reported by George Joseph)
* ASTERISK-24706 - [patch]add auto-dtmf mode for pjsip (Reported
by yaron nahum)
* ASTERISK-25051 - Remove unneeded uses of optional_api providers.
(Reported by Corey Farrell)
* ASTERISK-25040 - pbx: Improve performance of reloads by making
hint destruction more performant (Reported by Matt Jordan)
* ASTERISK-24917 - [patch] clang compilation warnings (Reported by
Diederik de Groot)
* ASTERISK-24949 - res_pjsip_outbound_registration: Backport line
functionality (Reported by Joshua Colp)
* ASTERISK-24965 - cel_pgsql - log_error string references CDR
instead of CEL (Reported by Rodrigo Ramirez Norambuena)
* ASTERISK-24918 - pjsip: add CLI options to display global and
system configuration (Reported by Scott Griepentrog)
* ASTERISK-24862 - [patch] Support in-dialog OPTIONS (Reported by
yaron nahum)
* ASTERISK-24802 - stasis: set a channel variable on websocket
disconnect error (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-13.4.0

Thank you for your continued support of Asterisk!


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