Asterisk 13.4.0-rc1 Now Available

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

The release of Asterisk 13.4.0-rc1 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 candidate:

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

New Feature

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

For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.4.0-rc1

Thank you for your continued support of Asterisk!

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