Asterisk 13.14.0 Now Available
he Asterisk Development Team has announced the release of Asterisk 13.14.0.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.14.0 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:
- [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 '[email protected]@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, please see the ChangeLog:
Thank you for your continued support of Asterisk!