Asterisk 13.14.0-rc2 Now Available
The Asterisk Development Team has announced the second 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-rc2 resolves several issues reported by the community and would have not been possible without your participation.
The following are the issues resolved in this release candidate:
- [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] - [patch]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] - [patch] 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] - [patch]pjsip: Transfers Broken with Compact Headers Enabled
- [ASTERISK-26670] - [patch] 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] - [patch] 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"
- [ASTERISK-26772] - Crash in srv.c on startup with pjsip
- [ASTERISK-26777] - res_sorcery_memory_cache deadlocks
- [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
- [ASTERISK-26630] - Make logging PJPROJECT messages a bit easier
For a full list of changes in this release candidate, please see the ChangeLog:
Thank you for your continued support of Asterisk!