Asterisk 13.4.0 Now Available

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:

Bug

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

Improvement

• [ASTERISK-24706] – add auto-dtmf mode for pjsip
• [ASTERISK-24744] – Swedish Core Voice prompts
• [ASTERISK-24802] – stasis: set a channel variable on websocket disconnect error
• [ASTERISK-24862] – Support in-dialog OPTIONS
• [ASTERISK-24892] – Super Awesome Company sound prompts
• [ASTERISK-24917] – clang compilation warnings
• [ASTERISK-24918] – pjsip: add CLI options to display global and system configuration
• [ASTERISK-24949] – res_pjsip_outbound_registration: Backport line functionality
• [ASTERISK-24965] – cel_pgsql – log_error string references CDR instead of CEL
• [ASTERISK-25040] – pbx: Improve performance of reloads by making hint destruction more performant
• [ASTERISK-25043] – Avoiding ERR_remove_state in OpenSSL
• [ASTERISK-25044] – sorcery: Add ability to insert a new wizard into an object type’s list
• [ASTERISK-25045] – vector: Add new capabilities and unit tests
• [ASTERISK-25051] – Remove unneeded uses of optional_api providers.

New Feature

• [ASTERISK-24922] – ARI: Add the ability to intercept hold and raise an event

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!