Asterisk News

Asterisk Releases

Asterisk 11.18.0-rc1 Now Available

May 21, 2015

The Asterisk Development Team has announced the first release candidate of Asterisk 11.18.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.18.0-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] - [patch] Buddies are always auto-registered when processing the roster
  • [ASTERISK-18032] - [patch] - IPv6 and IPv4 NAT not working
  • [ASTERISK-19538] - Asterisk segfaults on sippeers realtime redundancy
  • [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-21854] - Long Asterisk-version strings display improperly in the 'Connected to ...' line upon remote console connection
  • [ASTERISK-21893] - Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c
  • [ASTERISK-22352] - [patch] 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] - [patch]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-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] - [patch] - Buddies are always auto-registered when processing the roster
  • [ASTERISK-24805] - [patch] - ASAN: Race condition (heap-use-after-free) on asterisk closing
  • [ASTERISK-24847] - [security] [patch] tcptls: certificate CN NULL byte prefix bug
  • [ASTERISK-24864] - app_confbridge: file playback blocks dtmf
  • [ASTERISK-24881] - ast_register_atexit should only be used when absolutely needed
  • [ASTERISK-24887] - [patch]tags in a=crypto lines do not accept 2 or more digits
  • [ASTERISK-24894] - [patch] 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-24916] - Increasing memory usage when multiple reinvite during call
  • [ASTERISK-24932] - Asterisk 13.x does not build with GCC 5.0
  • [ASTERISK-24942] - Voicemail API: message is deleted when destination mailbox is at maxmsg
  • [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-24959] - [patch]CLI command cdr show pgsql status
  • [ASTERISK-24975] - Enabling 'DEBUG_THREADLOCALS' Causes the Build to Fail
  • [ASTERISK-24976] - cdr_odbc not include new columns added on 1.8
  • [ASTERISK-24991] - Check for ao2_alloc failure in __ast_channel_internal_alloc
  • [ASTERISK-25022] - Memory leak setting up DTLS/SRTP calls
  • [ASTERISK-25028] - Build System: Unneeded defines in asterisk/buildopts.h
  • [ASTERISK-25034] - chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests.
  • [ASTERISK-25038] - Queue log "EXITWITHTIMEOUT" does not always contain waiting time
  • [ASTERISK-25041] - [patch]Broken column type checking in res_config_mysql addon
  • [ASTERISK-25042] - asterisk.conf options override command-line options.
  • [ASTERISK-25074] - Regression: Recent clang-related change broke cross compiling of Asterisk
  • [ASTERISK-25083] - Message.c: Message channel becomes saturated with frames leading to spammy log messages
  • [ASTERISK-25112] - Logger: Configuration settings are not reset to default during reload.

Improvement

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0-rc1

Thank you for your continued support of Asterisk!


Asterisk 1.8.28-cert5, 1.8.32.3, 11.6-cert11, 11.17.1, 12.8.2, 13.1-cert2, 13.3.2 Now Available (Security Release)

Apr 8, 2015

The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11, 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of these versions resolves the following security vulnerability:

  • AST-2015-003: TLS Certificate Common name NULL byte exploit
    When Asterisk registers to a SIP TLS device and and verifies the server, Asterisk will accept signed certificates that match a common name other than the one Asterisk is expecting if the signed certificate has a common name containing a null byte after the portion of the common name that Asterisk expected. This potentially allows for a man in the middle attack.

For more information about the details of this vulnerability, please read security advisory AST-2015-003, which was released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs:

The security advisory is available at:

Thank you for your continued support of Asterisk!


Asterisk 13.3.1 Now Available

Apr 6, 2015

The Asterisk Development Team has announced the release of Asterisk 13.3.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.3.1 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:

  • pjsip: resolve ABI compatibility problem with external modules
    (Closes issue ASTERISK-24941. Reported by Matt Jordan)

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

 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.1

Thank you for your continued support of Asterisk!

 


Asterisk 13.2.1 Now Available

Apr 6, 2015

The Asterisk Development Team has announced the release of Asterisk 13.2.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.2.1 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:

  • pjsip: resolve ABI compatibility problem with external modules
    (Closes issue ASTERISK-24941. Reported by Matt Jordan)

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.2.1

Thank you for your continued support of Asterisk!


Asterisk 13.3.0 Now Available

Apr 1, 2015

The Asterisk Development Team has announced the release of Asterisk 13.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.3.0 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-15434] - [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller
  • [ASTERISK-16779] - Cannot disallow unknown format ''
  • [ASTERISK-17721] - Incoming SRTP calls that specify a key lifetime fail
  • [ASTERISK-18105] - most of asterisk modules are unbuildable in cygwin environment
  • [ASTERISK-18708] - func_curl hangs channel under load
  • [ASTERISK-19470] - Documentation on app_amd is incorrect
  • [ASTERISK-20850] - [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
  • [ASTERISK-21038] - Bad command completion of "core set debug channel"
  • [ASTERISK-22670] - Asterisk crashes when processing ISDN AoC Events
  • [ASTERISK-23214] - chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases
  • [ASTERISK-23390] - NewExten Event with application AGI shows up before and after AGI runs
  • [ASTERISK-24015] - app_transfer fails with PJSIP channels
  • [ASTERISK-24085] - Documentation - We should remove or further document the 'contact' section in pjsip.conf
  • [ASTERISK-24451] - chan_iax2: reference leak in sched_delay_remove
  • [ASTERISK-24479] - Enable REF_DEBUG for module references
  • [ASTERISK-24499] - Need more explicit debug when PJSIP dialstring is invalid
  • [ASTERISK-24612] - res_pjsip: No information if a required sorcery wizard is not loaded
  • [ASTERISK-24616] - Crash in res_format_attr_h264 due to invalid string copy
  • [ASTERISK-24632] - install_prereq script installs pjproject without IPv6 support
  • [ASTERISK-24677] - ARI GET variable on channel provides unhelpful response on non-existent variable
  • [ASTERISK-24685] - "pjsip show version" CLI command
  • [ASTERISK-24689] - Segfault on hangup after outgoing PRI-Euroisdn call
  • [ASTERISK-24700] - CRASH: NULL channel is being passed to ast_bridge_transfer_attended()
  • [ASTERISK-24701] - Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information
  • [ASTERISK-24716] - Improve pjsip log messages for presence subscription failure
  • [ASTERISK-24724] - 'httpstatus' Web Page Produces Incomplete HTML
  • [ASTERISK-24727] - PJSIP: Crash experienced during multi-Asterisk transfer scenario.
  • [ASTERISK-24739] - [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules
  • [ASTERISK-24740] - [patch]Segmentation fault on aoc-e event
  • [ASTERISK-24741] - dtls_handler causes Asterisk to crash
  • [ASTERISK-24742] - [patch] Fix ast_odbc_find_table function in res_odbc
  • [ASTERISK-24748] - res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur
  • [ASTERISK-24751] - Integer values in json payload to ARI cause asterisk to crash
  • [ASTERISK-24752] - Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown
  • [ASTERISK-24755] - Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge
  • [ASTERISK-24768] - res_timing_pthread: file descriptor leak
  • [ASTERISK-24769] - res_pjsip_sdp_rtp: Local ICE candidates leaked
  • [ASTERISK-24771] - ${CHANNEL(pjsip)} - segfault
  • [ASTERISK-24772] - ODBC error in realtime sippeers when device unregisters under MariaDB
  • [ASTERISK-24785] - 'Expires' header missing from 200 OK on REGISTER
  • [ASTERISK-24786] - [patch] - Asterisk terminates when playing a voicemail stored in LDAP
  • [ASTERISK-24787] - [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime
  • [ASTERISK-24791] - Crash in ast_rtcp_write_report
  • [ASTERISK-24796] - Codecs and bucket schema's prevent module unload
  • [ASTERISK-24797] - bridge_softmix: G.729 codec license held
  • [ASTERISK-24799] - [patch] make fails with undefined reference to SSLv3_client_method
  • [ASTERISK-24800] - Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill
  • [ASTERISK-24807] - Missing mandatory field Max-Forwards
  • [ASTERISK-24808] - res_config_odbc: Improper escaping of backslashes occurs with MySQL
  • [ASTERISK-24812] - ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding
  • [ASTERISK-24814] - asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers
  • [ASTERISK-24817] - init_logger_chain: unreachable code block
  • [ASTERISK-24825] - Caller ID not recognized using Centrex/Distinctive dialing
  • [ASTERISK-24828] - Fix Frame Leaks
  • [ASTERISK-24830] - res_rtp_asterisk.c checks USE_PJPROJECT not HAVE_PJPROJECT
  • [ASTERISK-24838] - chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling
  • [ASTERISK-24840] - res_pjsip: conflicting endpoint identifiers
  • [ASTERISK-24872] - [patch] AMI PJSIPShowEndpoint closes AMI connection on error
  • [ASTERISK-24876] - Investigate reference leaks from tests/channels/local/local_optimize_away
  • [ASTERISK-24879] - [patch]Compilation fails due to 64bit time under OpenBSD
  • [ASTERISK-24880] - [patch]Compilation under OpenBSD
  • [ASTERISK-24882] - chan_sip: Improve usage of REF_DEBUG

Improvement

  • [ASTERISK-24745] - [patch]Add no_answer to ARI hangup causes
  • [ASTERISK-24790] - Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context
  • [ASTERISK-24811] - asterisk-publication sorcery object does not use realtime

New Feature

  • [ASTERISK-17899] - Handle crypto lifetime in SDES-SRTP negotiation
  • [ASTERISK-24703] - ARI: Add the ability to "transfer" (redirect) a channel


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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0

Thank you for your continued support of Asterisk!

 

 

 

 


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