Asterisk Development Team
2017-Apr-07 16:38 UTC
[asterisk-users] Asterisk 14.4.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 14.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.4.0 resolves several issues reported by the community and would have not been possible without your participation. *Thank you!* The following issues are resolved in this release: *New Features made in this release:* ----------------------------------- - [ASTERISK-26878 <https://issues.asterisk.org/jira/browse/ASTERISK-26878>] - func_channel: Add ability to get the callid so dialplan has access to it. (Reported by Richard Mudgett) - [ASTERISK-26863 <https://issues.asterisk.org/jira/browse/ASTERISK-26863>] - res_pjsip: Add endpoint identification scheme based on a configured SIP header/value (Reported by Matt Jordan) - [ASTERISK-17428 <https://issues.asterisk.org/jira/browse/ASTERISK-17428>] - [patch] Allow "Comedian Mail" branding to be removed (Reported by John Covert) *Bugs fixed in this release:* ----------------------------------- - [ASTERISK-26851 <https://issues.asterisk.org/jira/browse/ASTERISK-26851>] - res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (Reported by Richard Begg) - [ASTERISK-26897 <https://issues.asterisk.org/jira/browse/ASTERISK-26897>] - chan_sip: Security vulnerability with client code header (Reported by Alex Villac?s Lasso) - [ASTERISK-26916 <https://issues.asterisk.org/jira/browse/ASTERISK-26916>] - res_pjsip: Excessive refcount reached on transport ao2 object (Reported by Ross Beer) - [ASTERISK-26705 <https://issues.asterisk.org/jira/browse/ASTERISK-26705>] - libasteriskssl.so not found when asterisk is installed for the 1st time (Reported by George Joseph) - [ASTERISK-26850 <https://issues.asterisk.org/jira/browse/ASTERISK-26850>] - res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets (Reported by Max Norba) - [ASTERISK-26484 <https://issues.asterisk.org/jira/browse/ASTERISK-26484>] - res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument. (Reported by Vinod Dharashive) - [ASTERISK-26776 <https://issues.asterisk.org/jira/browse/ASTERISK-26776>] - res_pjsip_pubsub: Crash when generating xpidf content (Reported by Andrew Green) - [ASTERISK-26880 <https://issues.asterisk.org/jira/browse/ASTERISK-26880>] - Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled (Reported by Kirsty Tyerman) - [ASTERISK-26862 <https://issues.asterisk.org/jira/browse/ASTERISK-26862>] - app_queue: Queue stops calling members with local interface after forwarding in previous call (Reported by Robert Mordec) - [ASTERISK-26732 <https://issues.asterisk.org/jira/browse/ASTERISK-26732>] - res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome (Reported by Dan Jenkins) - [ASTERISK-26879 <https://issues.asterisk.org/jira/browse/ASTERISK-26879>] - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) - [ASTERISK-26867 <https://issues.asterisk.org/jira/browse/ASTERISK-26867>] - autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade). (Reported by Krzysztof Trempala) - [ASTERISK-26869 <https://issues.asterisk.org/jira/browse/ASTERISK-26869>] - res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension (Reported by Torrey Searle) - [ASTERISK-26668 <https://issues.asterisk.org/jira/browse/ASTERISK-26668>] - core: Malformed pattern matching extension (various factors) results in crash (Reported by xrobau) - [ASTERISK-26865 <https://issues.asterisk.org/jira/browse/ASTERISK-26865>] - chan_iax2: Reload of iax peer results in loss of host address/port (Reported by Richard Begg) - [ASTERISK-26872 <https://issues.asterisk.org/jira/browse/ASTERISK-26872>] - Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal) (Reported by Matt Jordan) - [ASTERISK-26717 <https://issues.asterisk.org/jira/browse/ASTERISK-26717>] - Document the fact that Asterisk HEP support only works with the PJSIP channel driver (Reported by Olivier Krief) - [ASTERISK-26643 <https://issues.asterisk.org/jira/browse/ASTERISK-26643>] - Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk (Reported by Roman Bedros) - [ASTERISK-25237 <https://issues.asterisk.org/jira/browse/ASTERISK-25237>] - stasis_cache.c:845 caching_topic_exec: - misleading ERROR message (Reported by Smirnov Aleksey) - [ASTERISK-26857 <https://issues.asterisk.org/jira/browse/ASTERISK-26857>] - chan_pjsip: Dialplan function race condition (Reported by Joshua Colp) - [ASTERISK-26841 <https://issues.asterisk.org/jira/browse/ASTERISK-26841>] - chan_sip: Call not cancelled after receiving a 422 response (Reported by Jean Aunis - Prescom) - [ASTERISK-26822 <https://issues.asterisk.org/jira/browse/ASTERISK-26822>] - pjsip/cli_commands: pjsip show channelstats shows wrong codec (Reported by Kevin Harwell) - [ASTERISK-26353 <https://issues.asterisk.org/jira/browse/ASTERISK-26353>] - res_musiconhold: musiconhold seems to think that the general section is a class and issues warning (Reported by Jonathan Harris) - [ASTERISK-26685 <https://issues.asterisk.org/jira/browse/ASTERISK-26685>] - res_pjsip: Crash when using IPv6 and Transport ws,wss (Reported by Michael Balen) - [ASTERISK-24562 <https://issues.asterisk.org/jira/browse/ASTERISK-24562>] - app_voicemail: Cannot set fromstring on a per-mailbox basis (Reported by Mark Scholten) - [ASTERISK-26598 <https://issues.asterisk.org/jira/browse/ASTERISK-26598>] - Saynumber is trying to get "and" from "digits/" subfolder (Reported by Jonathan Harris) - [ASTERISK-17067 <https://issues.asterisk.org/jira/browse/ASTERISK-17067>] - Long lines in call files cause spurious syntax error (Reported by Dave Olszewski) - [ASTERISK-26796 <https://issues.asterisk.org/jira/browse/ASTERISK-26796>] - res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS' (Reported by J?rgen H) - [ASTERISK-25628 <https://issues.asterisk.org/jira/browse/ASTERISK-25628>] - res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging (Reported by Dmitry Wagin) - [ASTERISK-26774 <https://issues.asterisk.org/jira/browse/ASTERISK-26774>] - core: Playback URL fails after some time (Reported by Igor Gamayunov) - [ASTERISK-26825 <https://issues.asterisk.org/jira/browse/ASTERISK-26825>] - pjsip.conf.sample: user_agent: still refers to branch 12 (Reported by Tzafrir Cohen) - [ASTERISK-26823 <https://issues.asterisk.org/jira/browse/ASTERISK-26823>] - PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist (Reported by Mark Michelson) - [ASTERISK-26623 <https://issues.asterisk.org/jira/browse/ASTERISK-26623>] - res_pjsip: Crash when calling PJSIPShowEndpoint (Reported by J?rgen H) - [ASTERISK-26808 <https://issues.asterisk.org/jira/browse/ASTERISK-26808>] - res_pjsip_outbound_registration doesn't know about network change events (Reported by George Joseph) - [ASTERISK-26781 <https://issues.asterisk.org/jira/browse/ASTERISK-26781>] - bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio (Reported by Sean Bright) - [ASTERISK-26782 <https://issues.asterisk.org/jira/browse/ASTERISK-26782>] - res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication (Reported by Peter Sokolov) - [ASTERISK-26812 <https://issues.asterisk.org/jira/browse/ASTERISK-26812>] - [patch] Fix download_externals To Allow The Use Of curl Or wget (Reported by Michael L. Young) - [ASTERISK-18271 <https://issues.asterisk.org/jira/browse/ASTERISK-18271>] - Pattern matching with res_config_mysql extensions does not behave as expected (Reported by Charlie Smurthwaite) - [ASTERISK-26669 <https://issues.asterisk.org/jira/browse/ASTERISK-26669>] - PJSIP Segfault 13.13.1 (Bundled PJSIP) (Reported by Nic Colledge) - [ASTERISK-18731 <https://issues.asterisk.org/jira/browse/ASTERISK-18731>] - [patch] DUNDi weight parameter not processed correctly (Reported by Peter Racz) - [ASTERISK-26799 <https://issues.asterisk.org/jira/browse/ASTERISK-26799>] - res_pjsip: Using an auth object for inbound and outbound authentication fails. (Reported by Richard Mudgett) - [ASTERISK-26738 <https://issues.asterisk.org/jira/browse/ASTERISK-26738>] - Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c (Reported by Michael Maier) - [ASTERISK-25893 <https://issues.asterisk.org/jira/browse/ASTERISK-25893>] - Function vmauthenticate accesses uninitialized memory (Reported by Filip Jenicek) - [ASTERISK-26580 <https://issues.asterisk.org/jira/browse/ASTERISK-26580>] - [patch] Error during LDAP modify action when user unregisters (Reported by Nicholas John Koch) - [ASTERISK-26802 <https://issues.asterisk.org/jira/browse/ASTERISK-26802>] - [patch] Integrity Check Of PJSIP Download Fails (Reported by Michael L. Young) - [ASTERISK-15858 <https://issues.asterisk.org/jira/browse/ASTERISK-15858>] - [patch] Fix query with double backslash in string literals and stop log warnings (Reported by Humberto Figuera) - [ASTERISK-26057 <https://issues.asterisk.org/jira/browse/ASTERISK-26057>] - res_config_sqlite3 uses incorrect query - unnecessary escape (Reported by Stepan) - [ASTERISK-23457 <https://issues.asterisk.org/jira/browse/ASTERISK-23457>] - SQlite3: Realtime queue loading fails after PRAGMA query result (Reported by Scott Griepentrog) - [ASTERISK-26794 <https://issues.asterisk.org/jira/browse/ASTERISK-26794>] - http: Crash on Reload Only in ast_tcptls_server_start (Reported by Joshua Elson) - [ASTERISK-26714 <https://issues.asterisk.org/jira/browse/ASTERISK-26714>] - Phone default have not ringing on ARM (Reported by Igor Goncharovsky) - [ASTERISK-26696 <https://issues.asterisk.org/jira/browse/ASTERISK-26696>] - pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh (Reported by Zach R) - [ASTERISK-26756 <https://issues.asterisk.org/jira/browse/ASTERISK-26756>] - res_pjsip_mwi: Asterisk does not terminate MWI subscription (Reported by Carl Fortin) - [ASTERISK-26109 <https://issues.asterisk.org/jira/browse/ASTERISK-26109>] - Asterisk fails building with OpenSSL 1.1.0 (Reported by Tzafrir Cohen) - [ASTERISK-26723 <https://issues.asterisk.org/jira/browse/ASTERISK-26723>] - VoiceMailPlayMsg not playing messages via realtime (Reported by Ryan Rittgarn) - [ASTERISK-18286 <https://issues.asterisk.org/jira/browse/ASTERISK-18286>] - [patch] 'Silence' is truncated in Record() (Reported by var) - [ASTERISK-26248 <https://issues.asterisk.org/jira/browse/ASTERISK-26248>] - chan_pjsip: Error when calling PJSIP client with domain specified (Reported by Norbert Varga) - [ASTERISK-26788 <https://issues.asterisk.org/jira/browse/ASTERISK-26788>] - core: Protect flags during ast_waitfor (Reported by Joshua Colp) - [ASTERISK-26115 <https://issues.asterisk.org/jira/browse/ASTERISK-26115>] - pbx: AMI Originate ignore "failed" extension on call failure (Reported by Nasir Iqbal) - [ASTERISK-26785 <https://issues.asterisk.org/jira/browse/ASTERISK-26785>] - configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample (Reported by Torrey Searle) - [ASTERISK-26772 <https://issues.asterisk.org/jira/browse/ASTERISK-26772>] - Crash in srv.c on startup with pjsip (Reported by nappsoft) - [ASTERISK-26770 <https://issues.asterisk.org/jira/browse/ASTERISK-26770>] - res_stasis_device_state: Duplicate subscriptions when multiple received at same time (Reported by Joshua Colp) *Improvements made in this release:* ----------------------------------- - [ASTERISK-26864 <https://issues.asterisk.org/jira/browse/ASTERISK-26864>] - res_pjsip_session: Add support for overlap dialling (Reported by Richard Begg) - [ASTERISK-26846 <https://issues.asterisk.org/jira/browse/ASTERISK-26846>] - chan_sip: Add rtcp-mux support (Reported by Sean Bright) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.4.0 *Thank you for your continued support of Asterisk!* -------------- next part -------------- An HTML attachment was scrubbed... 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