The Asterisk Development Team would like to announce the release of Asterisk 13.15.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.15.0 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 - func_channel: Add ability to get the callid so dialplan has access to it. (Reported by Richard Mudgett) * ASTERISK-26863 - res_pjsip: Add endpoint identification scheme based on a configured SIP header/value (Reported by Matt Jordan) * ASTERISK-17428 - [patch] Allow "Comedian Mail" branding to be removed (Reported by John Covert) Bugs fixed in this release: ----------------------------------- * ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (Reported by Richard Begg) * ASTERISK-26897 - chan_sip: Security vulnerability with client code header (Reported by Alex Villacís Lasso) * ASTERISK-26916 - res_pjsip: Excessive refcount reached on transport ao2 object (Reported by Ross Beer) * ASTERISK-26705 - libasteriskssl.so not found when asterisk is installed for the 1st time (Reported by George Joseph) * ASTERISK-26850 - res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets (Reported by Max Norba) * ASTERISK-26484 - res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument. (Reported by Vinod Dharashive) * ASTERISK-26776 - res_pjsip_pubsub: Crash when generating xpidf content (Reported by Andrew Green) * ASTERISK-26880 - Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled (Reported by Kirsty Tyerman) * ASTERISK-26862 - app_queue: Queue stops calling members with local interface after forwarding in previous call (Reported by Robert Mordec) * ASTERISK-26732 - res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome (Reported by Dan Jenkins) * ASTERISK-26879 - PJSIP external_media_address ignored if no local_net options are provided (Reported by Matt Jordan) * ASTERISK-26867 - autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade). (Reported by Krzysztof Trempala) * 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 - core: Malformed pattern matching extension (various factors) results in crash (Reported by xrobau) * ASTERISK-26865 - chan_iax2: Reload of iax peer results in loss of host address/port (Reported by Richard Begg) * 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 - Document the fact that Asterisk HEP support only works with the PJSIP channel driver (Reported by Olivier Krief) * ASTERISK-26643 - Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk (Reported by Roman Bedros) * ASTERISK-25237 - stasis_cache.c:845 caching_topic_exec: - misleading ERROR message (Reported by Smirnov Aleksey) * ASTERISK-26857 - chan_pjsip: Dialplan function race condition (Reported by Joshua Colp) * ASTERISK-26841 - chan_sip: Call not cancelled after receiving a 422 response (Reported by Jean Aunis - Prescom) * ASTERISK-26822 - pjsip/cli_commands: pjsip show channelstats shows wrong codec (Reported by Kevin Harwell) * ASTERISK-26685 - res_pjsip: Crash when using IPv6 and Transport ws,wss (Reported by Michael Balen) * ASTERISK-24562 - app_voicemail: Cannot set fromstring on a per-mailbox basis (Reported by Mark Scholten) * ASTERISK-26598 - Saynumber is trying to get "and" from "digits/" subfolder (Reported by Jonathan Harris) * ASTERISK-17067 - Long lines in call files cause spurious syntax error (Reported by Dave Olszewski) * ASTERISK-26796 - res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS' (Reported by Jørgen H) * 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-26825 - pjsip.conf.sample: user_agent: still refers to branch 12 (Reported by Tzafrir Cohen) * ASTERISK-26823 - PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist (Reported by Mark Michelson) * ASTERISK-26623 - res_pjsip: Crash when calling PJSIPShowEndpoint (Reported by Jørgen H) * ASTERISK-26808 - res_pjsip_outbound_registration doesn't know about network change events (Reported by George Joseph) * ASTERISK-26313 - chan_sip : Asterisk restart seems to be required for changing encryption option (Reported by benasse) * ASTERISK-26781 - bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio (Reported by Sean Bright) * ASTERISK-26782 - res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication (Reported by Peter Sokolov) * ASTERISK-26812 - [patch] Fix download_externals To Allow The Use Of curl Or wget (Reported by Michael L. Young) * ASTERISK-18271 - Pattern matching with res_config_mysql extensions does not behave as expected (Reported by Charlie Smurthwaite) * ASTERISK-26669 - PJSIP Segfault 13.13.1 (Bundled PJSIP) (Reported by Nic Colledge) * ASTERISK-18731 - [patch] DUNDi weight parameter not processed correctly (Reported by Peter Racz) * ASTERISK-26580 - [patch] Error during LDAP modify action when user unregisters (Reported by Nicholas John Koch) * ASTERISK-26799 - res_pjsip: Using an auth object for inbound and outbound authentication fails. (Reported by Richard Mudgett) * 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 - Function vmauthenticate accesses uninitialized memory (Reported by Filip Jenicek) * ASTERISK-26802 - [patch] Integrity Check Of PJSIP Download Fails (Reported by Michael L. Young) * ASTERISK-15858 - [patch] Fix query with double backslash in string literals and stop log warnings (Reported by Humberto Figuera) * ASTERISK-26057 - res_config_sqlite3 uses incorrect query - unnecessary escape (Reported by Stepan) * ASTERISK-23457 - SQlite3: Realtime queue loading fails after PRAGMA query result (Reported by Scott Griepentrog) * ASTERISK-26794 - http: Crash on Reload Only in ast_tcptls_server_start (Reported by Joshua Elson) * ASTERISK-26714 - Phone default have not ringing on ARM (Reported by Igor Goncharovsky) * ASTERISK-26696 - pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh (Reported by Zach R) * ASTERISK-26756 - res_pjsip_mwi: Asterisk does not terminate MWI subscription (Reported by Carl Fortin) * ASTERISK-26109 - Asterisk fails building with OpenSSL 1.1.0 (Reported by Tzafrir Cohen) * ASTERISK-26723 - VoiceMailPlayMsg not playing messages via realtime (Reported by Ryan Rittgarn) * ASTERISK-18286 - [patch] 'Silence' is truncated in Record() (Reported by var) * ASTERISK-26248 - chan_pjsip: Error when calling PJSIP client with domain specified (Reported by Norbert Varga) * ASTERISK-26788 - core: Protect flags during ast_waitfor (Reported by Joshua Colp) * ASTERISK-26115 - pbx: AMI Originate ignore "failed" extension on call failure (Reported by Nasir Iqbal) * ASTERISK-26785 - configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample (Reported by Torrey Searle) * ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported by nappsoft) * 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 - res_pjsip_session: Add support for overlap dialling (Reported by Richard Begg) * 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-13.15.0 Thank you for your continued support of Asterisk!