Asterisk 13.14.0-rc1 Now Available

The Asterisk Development Team has announced the first release candidate of Asterisk 13.14.0.

This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.14.0-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-21094] – MixMonitorMute mutes through stream if already slinear (e.g. Originate)
  • [ASTERISK-24330] – Requirement for ‘wss’ value in Contact header transport parameter on inbound traffic violates RFC7118
  • [ASTERISK-24499] – Need more explicit debug when PJSIP dialstring is invalid
  • [ASTERISK-24858] – Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
  • [ASTERISK-25083] – Message.c: Message channel becomes saturated with frames leading to spammy log messages
  • [ASTERISK-25494] – build: GCC 5.1.x catches some new const, array bounds and missing paren issues
  • [ASTERISK-25951] – res_agi: run_agi eats frames it shouldn’t
  • [ASTERISK-26343] – ASTERISK-25951 causes issues for callerid manipulation through agi
  • [ASTERISK-26433] – chan_sip: Allows To-tag checks to be bypassed, setting up new calls
  • [ASTERISK-26490] – res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains “_”
  • [ASTERISK-26503] – app_voicemail: Asterisk crashes when MailboxExists is used
  • [ASTERISK-26523] – chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes – rtptimeout behaving badly – regression
  • [ASTERISK-26546] – mips64el and x32 – undefined reference to symbol ‘dlopen@@GLIBC_2.2’
  • [ASTERISK-26566] – res_rtp_asterisk: RTT miscalculation in RTCP
  • [ASTERISK-26579] – codec_opus: Recursiveness when parsing fmtp line
  • [ASTERISK-26586] – chan_sip: Segfaults upon reload if client with MWI wasn’t registered
  • [ASTERISK-26603] – chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
  • [ASTERISK-26604] – chan_sip: sip reload doesn’t apply changes to tlscertfile, tlsciphers, etc.
  • [ASTERISK-26617] – res_rtp_asterisk: Can’t bind on systems without IPv6
  • [ASTERISK-26621] – app_queue: Queue application does not ring members with Local interface
  • [ASTERISK-26632] – core: Possibility of a frame “imbalance” leading to stuck channels.
  • [ASTERISK-26644] – PJSIPShowRegistrationsInbound just dumps all aors
  • [ASTERISK-26653] – pjproject_bundled doesn’t verify already downloaded tarballs
  • [ASTERISK-26655] – pjsip: Transfers Broken with Compact Headers Enabled
  • [ASTERISK-26670] – Outgoing SIP-URI Dialing via PJSIP
  • [ASTERISK-26672] – Crash when setting remote address on RTP instance
  • [ASTERISK-26673] – chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
  • [ASTERISK-26679] – Crash on invalid contact domain (pjsip aor)
  • [ASTERISK-26684] – res_pjsip: Various issues with compact SIP headers
  • [ASTERISK-26691] – Remember SDP negotiation on SIP_CODEC_INBOUND.
  • [ASTERISK-26693] – res_pjsip_endpoint_identifier_ip: Add support for SRV
  • [ASTERISK-26699] – res_pjsip: Assertion when sending OPTIONS request to endpoint
  • [ASTERISK-26704] – res_odbc.conf contains deprecated configuration: ‘pooling’, ‘shared_connections’, ‘limit’, and ‘idlecheck’ options were replaced by ‘max_connections’.
  • [ASTERISK-26710] – res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
  • [ASTERISK-26716] – ari: Channels with pre-dial handlers cannot be hung up via ARI
  • [ASTERISK-26731] – res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
  • [ASTERISK-26735] – res_pjsip_endpoint_identifier_ip: “srv_lookups” after match in .conf has no effect
  • [ASTERISK-26739] – voicemail API test: confuses expected and actual values
  • [ASTERISK-26740] – voicemail API test: uses varlibdir instead of datadir for a sound file
  • [ASTERISK-26743] – PJPROJECT: Detecting compiled max log level does not work.
  • [ASTERISK-26753] – AMI disconnect causes “ast_careful_fwrite: fwrite() returned error: Broken pipe”
  • [ASTERISK-26754] – build_tools: make_build_h does not handle \ in user name
  • [ASTERISK-26755] – app_queue: Random queues disappear on “core reload queue all”

Improvement

  • [ASTERISK-23828] – pjsip – Need a command to list active SIP subscriptions
  • [ASTERISK-26527] – Testsuite: increase timeout to check “core fullybooted wait” up to 30 sec
  • [ASTERISK-26562] – app_controlplayback: Transmit Silence on ControlPlayback pause
  • [ASTERISK-26624] – res_calendar_caldav: Add support for gmail

New Feature

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.14.0-rc1

Thank you for your continued support of Asterisk!

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