summaryrefslogtreecommitdiff
path: root/comms
AgeCommit message (Collapse)AuthorFilesLines
2017-06-21Update to Asterisk 14.5.0: this is mostly a bug fix releases withjnemeth8-107/+88
patches for a number of security issues, several of which do not apply to this package because they relate to PJSIP: AST-2016-009, AST-2016-010, AST-2017-001, AST-2017-002, AST-2017-003, and AST-2017-004. ----- 14.5.0 The Asterisk Development Team would like to announce the release of Asterisk 14.5.0. The release of Asterisk 14.5.0 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: Bugs fixed in this release: ----------------------------------- * ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable (Reported by Stefan Engström) * ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 (Reported by Javier Riveros) * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) * ASTERISK-26998 - res_pjsip_session: INVITE retransmissions could still setup the same call again. (Reported by Richard Mudgett) * ASTERISK-26143 - res_rtp_asterisk: One way audio when transcoding (Reported by Henning Holtschneider) * ASTERISK-26606 - tcptls: Incorrect OpenSSL function call leads to misleading error report (Reported by Bob Ham) * ASTERISK-26983 - Crash in Manager Reload when TLS Config Changes (Reported by Joshua Elson) * ASTERISK-25032 - [patch]cel_odbc sometimes inserts CEL with wrong eventtime (Reported by Etienne Lessard) * ASTERISK-26173 - func_cdr: CDR function does not permit empty values to be assigned (Reported by gkloepfer) * ASTERISK-25506 - [patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages. (Reported by Frederic LE FOLL) * ASTERISK-24529 - Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used (Reported by Corey Farrell) * ASTERISK-26860 - Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null) (Reported by Evers Lab) * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address (Reported by Ksenia) * ASTERISK-26974 - res_pjsip: Deadlock in T.38 framehook (Reported by Richard Mudgett) * ASTERISK-26908 - res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked. (Reported by Richard Mudgett) * ASTERISK-25823 - SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory. (Reported by Andreas Krüger) * ASTERISK-26926 - func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) * ASTERISK-26951 - chan_sip: ACK with SDP does not update a direct media bridge (Reported by Jean Aunis - Prescom) * ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux (Reported by abelbeck) * ASTERISK-26929 - pjsip: Add database tables for RLS (Reported by Joshua Colp) * ASTERISK-26953 - Asterisk crash if hep.conf have some missing parameters (Reported by Joel Vandal) * ASTERISK-26890 - STUN server with non-default-route transport causes INVITE delay (Reported by George Joseph) * ASTERISK-26692 - res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) (Reported by scgm11) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) * ASTERISK-21856 - STUN never works when asterisk started without internet access (Reported by Jeremy Kister) * ASTERISK-20984 - Audible clicks when playing sox encoded au file with STREAM FILE AGI command (Reported by Roman S.) * ASTERISK-26528 - [UBSAN] strings.h:signed integer overflow in ast_str_case_hash (Reported by Badalian Vyacheslav) * ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (Reported by Richard Begg) * ASTERISK-26903 - Listening TCP/TLS sockets stop when temporarily out of open files (Reported by Walter Doekes) * ASTERISK-26928 - pjsip: Add database tables for PUBLISH support (Reported by Joshua Colp) * ASTERISK-26927 - pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). (Reported by Alexander Traud) * ASTERISK-26905 - pjproject_bundled: Merge 3 upstream deadlock patches into bundled (Reported by Ross Beer) * ASTERISK-26897 - chan_sip: Security vulnerability with client code header (Reported by Alex Villacís Lasso) * ASTERISK-25974 - Unused realtime MOH classes not purged on 'moh reload' (Reported by Sébastien Couture) * ASTERISK-26916 - res_pjsip: Excessive refcount reached on transport ao2 object (Reported by Ross Beer) * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers (Reported by Olle Johansson) * ASTERISK-26915 - chan_sip: Session Timers required but refused wrongly. (Reported by Alexander Traud) * ASTERISK-26363 - res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code (Reported by Yaacov Akiba Slama) * ASTERISK-26896 - Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT (Reported by twisted) * ASTERISK-26705 - libasteriskssl.so not found when asterisk is installed for the 1st time (Reported by George Joseph) * ASTERISK-21009 - xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client (Reported by Marcello Ceschia) * ASTERISK-25490 - [patch]SDP crypto tag is validated incorrectly (Reported by Joerg Sonnenberger) * ASTERISK-26086 - res_musiconhold: format option is not documented adequately (Reported by Jens Bürger) * ASTERISK-23996 - No core dumps because of res_musiconhold chdir. (Reported by Walter Doekes) * ASTERISK-24712 - xmpp: starttls problem causes connection spew (Reported by Matthias Urlichs) * ASTERISK-26814 - pjproject_bundled build fails to download pjproject source when using cURL (Reported by Gergely Dömsödi) * ASTERISK-23510 - JABBER_STATUS fails with improper code 7 for unavailable clients (Reported by Anthony Critelli) * ASTERISK-21855 - Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available * ASTERISK-25622 - WARNING for "JABBER: socket read error" should be more specific (Reported by Sean Darcy) * ASTERISK-26515 - rtp_engine: Allocate RTP payloads on a per-session basis (Reported by Joshua Colp) * ASTERISK-26818 - cdr: Problem setting variables in h exten (Reported by scgm11) * ASTERISK-26875 - app_mixmonitor: Recording out of sync when 183 but no RTP (Reported by Aaron An) Improvements made in this release: ----------------------------------- * ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech - Israel)) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.5.0 Thank you for your continued support of Asterisk! ----- 14.4.0 The Asterisk Development Team would like to announce the release of Asterisk 14.4.0. 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 Villacís 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 Jørgen 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 Jørgen 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) *Thank you for your continued support of Asterisk!* ----- 14.3.0 The Asterisk Development Team has announced the release of Asterisk 14.3.0. The release of Asterisk 14.3.0 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: New Features made in this release: ----------------------------------- * ASTERISK-26630 - Make logging PJPROJECT messages a bit easier (Reported by Richard Mudgett) Bugs fixed in this release: ----------------------------------- * ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported by nappsoft) * ASTERISK-26767 - ARI channelvars cause memory leak (Reported by Sébastien Duthil) * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be hung up via ARI (Reported by Tom Pawelek) * ASTERISK-26632 - core: Possibility of a frame "imbalance" leading to stuck channels. (Reported by Mark Michelson) * ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't (Reported by George Joseph) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26704 - res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'. (Reported by Anthony Messina) * ASTERISK-26765 - res_resolver_unbound: FRACK! Excessive ref count trap tripped. (Reported by Richard Mudgett) * ASTERISK-21094 - MixMonitorMute mutes through stream if already slinear (e.g. Originate) (Reported by David Woolley) * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor) (Reported by Dmitriy) * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint (Reported by Ross Beer) * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) * ASTERISK-26754 - build_tools: make_build_h does not handle \ in user name (Reported by Kirill Katsnelson) * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe" (Reported by Kirill Katsnelson) * ASTERISK-26755 - app_queue: Random queues disappear on "core reload queue all" (Reported by Kirill Katsnelson) * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect (Reported by Michael Maier) * ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support for SRV (Reported by Joshua Colp) * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level does not work. (Reported by Richard Mudgett) * ASTERISK-26740 - voicemail API test: uses varlibdir instead of datadir for a sound file (Reported by Tzafrir Cohen) * ASTERISK-26739 - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen) * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem) * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 (Reported by Aaron An) * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud) * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp) * ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson) * ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact Headers Enabled (Reported by JoshE) * ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett) * ASTERISK-26621 - app_queue: Queue application does not ring members with Local interface (Reported by Jonas Kellens) * ASTERISK-26586 - chan_sip: Segfaults upon reload if client with MWI wasn't registered (Reported by Michael Kuron) * ASTERISK-25494 - build: GCC 5.1.x catches some new const, array bounds and missing paren issues (Reported by George Joseph) * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring is invalid (Reported by Rusty Newton) * ASTERISK-25083 - Message.c: Message channel becomes saturated with frames leading to spammy log messages (Reported by Jonathan Rose) * ASTERISK-26653 - pjproject_bundled doesn't verify already downloaded tarballs (Reported by George Joseph) * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed, setting up new calls (Reported by Walter Doekes) * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp line (Reported by Jørgen H) * ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all aors (Reported by George Joseph) * ASTERISK-26647 - Support older DNS style for OpenBSD (Reported by snuffy) * ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_" (Reported by Juris Breicis) * ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems without IPv6 (Reported by Guido Falsi) * ASTERISK-26603 - [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no (Reported by Alexei Gradinari) * ASTERISK-24330 - Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 (Reported by Marek Cervenka) * ASTERISK-26546 - mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2' (Reported by Tzafrir Cohen) * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion) * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron) * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) Improvements made in this release: ----------------------------------- * ASTERISK-23828 - pjsip - Need a command to list active SIP subscriptions (Reported by Rusty Newton) * ASTERISK-26527 - Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec (Reported by Badalian Vyacheslav) * ASTERISK-26624 - res_calendar_caldav: Add support for gmail (Reported by Eduardo Scudeller Libardi) * ASTERISK-26562 - app_controlplayback: Transmit Silence on ControlPlayback pause (Reported by Mikheili Dautashvili) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.3.0 Thank you for your continued support of Asterisk!
2017-06-07Fix build with Perl 5.26.0ryoon4-2/+28
2017-06-05Recursive revbump from lang/perl5 5.26.0ryoon8-14/+16
2017-06-04Update to Asterisk 13.16.0: this is mostly a bugfix release.jnemeth6-54/+55
The Asterisk Development Team would like to announce the release of Asterisk 13.16.0. The release of Asterisk 13.16.0 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: Bugs fixed in this release: ----------------------------------- * ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable (Reported by Stefan Engström) * ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 (Reported by Javier Riveros) * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) * ASTERISK-26998 - res_pjsip_session: INVITE retransmissions could still setup the same call again. (Reported by Richard Mudgett) * ASTERISK-26143 - res_rtp_asterisk: One way audio when transcoding (Reported by Henning Holtschneider) * ASTERISK-26606 - tcptls: Incorrect OpenSSL function call leads to misleading error report (Reported by Bob Ham) * ASTERISK-26983 - Crash in Manager Reload when TLS Config Changes (Reported by Joshua Elson) * ASTERISK-25032 - [patch]cel_odbc sometimes inserts CEL with wrong eventtime (Reported by Etienne Lessard) * ASTERISK-26173 - func_cdr: CDR function does not permit empty values to be assigned (Reported by gkloepfer) * ASTERISK-25506 - [patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages. (Reported by Frederic LE FOLL) * ASTERISK-24529 - Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used (Reported by Corey Farrell) * ASTERISK-26860 - Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null) (Reported by Evers Lab) * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address (Reported by Ksenia) * ASTERISK-26974 - res_pjsip: Deadlock in T.38 framehook (Reported by Richard Mudgett) * ASTERISK-26908 - res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked. (Reported by Richard Mudgett) * ASTERISK-25823 - SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory. (Reported by Andreas Krüger) * ASTERISK-26951 - chan_sip: ACK with SDP does not update a direct media bridge (Reported by Jean Aunis - Prescom) * ASTERISK-26930 - pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux (Reported by abelbeck) * ASTERISK-26926 - func_speex: Crash caused by frame with no datalen (Reported by Richard Kenner) * ASTERISK-26929 - pjsip: Add database tables for RLS (Reported by Joshua Colp) * ASTERISK-26953 - Asterisk crash if hep.conf have some missing parameters (Reported by Joel Vandal) * ASTERISK-26890 - STUN server with non-default-route transport causes INVITE delay (Reported by George Joseph) * ASTERISK-26692 - res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip) (Reported by scgm11) * ASTERISK-26835 - res_rtp_asterisk: Crash when freeing RTCP address string (Reported by Niklas Larsson) * ASTERISK-26853 - res_rtp_asterisk: Crash in pjnath when receiving packet (Reported by Adagio) * ASTERISK-26613 - format_wav: wav16 format read file only by 320 - half of frame (Reported by Vitaly K) * ASTERISK-26169 - format_ogg_vorbis: Memory leak using OGG in MixMonitor (Reported by Ivan Myalkin) * ASTERISK-21856 - STUN never works when asterisk started without internet access (Reported by Jeremy Kister) * ASTERISK-20984 - Audible clicks when playing sox encoded au file with STREAM FILE AGI command (Reported by Roman S.) * ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport (Reported by Richard Begg) * ASTERISK-26903 - Listening TCP/TLS sockets stop when temporarily out of open files (Reported by Walter Doekes) * ASTERISK-26528 - [UBSAN] strings.h:signed integer overflow in ast_str_case_hash (Reported by Badalian Vyacheslav) * ASTERISK-26928 - pjsip: Add database tables for PUBLISH support (Reported by Joshua Colp) * ASTERISK-26927 - pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete(). (Reported by Alexander Traud) * ASTERISK-26905 - pjproject_bundled: Merge 3 upstream deadlock patches into bundled (Reported by Ross Beer) * ASTERISK-26897 - chan_sip: Security vulnerability with client code header (Reported by Alex Villacís Lasso) * ASTERISK-25974 - Unused realtime MOH classes not purged on 'moh reload' (Reported by Sébastien Couture) * ASTERISK-26916 - res_pjsip: Excessive refcount reached on transport ao2 object (Reported by Ross Beer) * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers (Reported by Olle Johansson) * ASTERISK-26915 - chan_sip: Session Timers required but refused wrongly. (Reported by Alexander Traud) * ASTERISK-26363 - res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code (Reported by Yaacov Akiba Slama) * ASTERISK-26896 - Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT (Reported by twisted) * ASTERISK-26705 - libasteriskssl.so not found when asterisk is installed for the 1st time (Reported by George Joseph) * ASTERISK-21009 - xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client (Reported by Marcello Ceschia) * ASTERISK-25490 - [patch]SDP crypto tag is validated incorrectly (Reported by Joerg Sonnenberger) * ASTERISK-24712 - xmpp: starttls problem causes connection spew (Reported by Matthias Urlichs) * ASTERISK-26086 - res_musiconhold: format option is not documented adequately (Reported by Jens Bürger) * ASTERISK-23996 - No core dumps because of res_musiconhold chdir. (Reported by Walter Doekes) * ASTERISK-26814 - pjproject_bundled build fails to download pjproject source when using cURL (Reported by Gergely Dömsödi) * ASTERISK-23510 - JABBER_STATUS fails with improper code 7 for unavailable clients (Reported by Anthony Critelli) * ASTERISK-21855 - Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available (Reported by Jeremy Kister) * ASTERISK-25622 - WARNING for "JABBER: socket read error" should be more specific (Reported by Sean Darcy) * ASTERISK-26818 - cdr: Problem setting variables in h exten (Reported by scgm11) * ASTERISK-26875 - app_mixmonitor: Recording out of sync when 183 but no RTP (Reported by Aaron An) Improvements made in this release: ----------------------------------- * ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26427 - res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip (Reported by Nir Simionovich (GreenfieldTech - Israel)) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.16.0 Thank you for your continued support of Asterisk!
2017-05-29Add fixes for AST-2017-002, AST-2017-003, and AST-2017-004. Notejnemeth2-11/+11
that the first two don't affect pkgsrc as we are using chan_sip not PJSIP. The last only affects users of SCCP, which is Cisco's proprietary protocol. ----- AST-2017-002 A remote crash can be triggered by sending a SIP packet to Asterisk with a specially crafted CSeq header and a Via header with no branch parameter. The issue is that the PJSIP RFC 2543 transaction key generation algorithm does not allocate a large enough buffer. By overrunning the buffer, the memory allocation table becomes corrupted, leading to an eventual crash. This issue is in PJSIP, and so the issue can be fixed without performing an upgrade of Asterisk at all. However, we are releasing a new version of Asterisk with the bundled PJProject updated to include the fix. If you are running Asterisk with chan_sip, this issue does not affect you. ----- AST-2017-003 The multi-part body parser in PJSIP contains a logical error that can make certain multi-part body parts attempt to read memory from outside the allowed boundaries. A specially-crafted packet can trigger these invalid reads and potentially induce a crash. The issue is within the PJSIP project and not in Asterisk. Therefore, the problem can be fixed without upgrading Asterisk. However, we will be releasing a new version of Asterisk where the bundled version of PJSIP has been updated to have the bug patched. If you are using Asterisk with chan_sip, this issue does not affect you. ----- AST-2017-004 A remote memory exhaustion can be triggered by sending an SCCP packet to Asterisk system with chan_skinny enabled that is larger than the length of the SCCP header but smaller than the packet length specified in the header. The loop that reads the rest of the packet doesn't detect that the call to read() returned end-of-file before the expected number of bytes and continues infinitely. The partial data message logging in that tight loop causes Asterisk to exhaust all available memory.
2017-05-13Update to Asterisk 13.15.0. This is mostly a bug fix release with a fewjnemeth5-42/+24
minor enhancements. 13.14.1 was released to fix AST-2017-001. ----- 13.15.0 The Asterisk Development Team would like to announce the release of Asterisk 13.15.0. 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 <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 Villacís 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-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 Jørgen 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-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 Jørgen 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-26313 <https://issues.asterisk.org/jira/browse/ASTERISK-26313>] - chan_sip : Asterisk restart seems to be required for changing encryption option (Reported by benasse) - [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-26580 <https://issues.asterisk.org/jira/browse/ASTERISK-26580>] - [patch] Error during LDAP modify action when user unregisters (Reported by Nicholas John Koch) - [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-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-13.15.0 *Thank you for your continued support of Asterisk!* ----- 13.14.0 The Asterisk Development Team has announced the release of Asterisk 13.14.0. The release of Asterisk 13.14.0 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: New Features made in this release: ----------------------------------- * ASTERISK-26630 - Make logging PJPROJECT messages a bit easier (Reported by Richard Mudgett) Bugs fixed in this release: ----------------------------------- * ASTERISK-26772 - Crash in srv.c on startup with pjsip (Reported by nappsoft) * ASTERISK-26704 - res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'. (Reported by Anthony Messina) * ASTERISK-21094 - MixMonitorMute mutes through stream if already slinear (e.g. Originate) (Reported by David Woolley) * ASTERISK-26716 - ari: Channels with pre-dial handlers cannot be hung up via ARI (Reported by Tom Pawelek) * ASTERISK-26632 - core: Possibility of a frame "imbalance" leading to stuck channels. (Reported by Mark Michelson) * ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't (Reported by George Joseph) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26679 - Crash on invalid contact domain (pjsip aor) (Reported by Dmitriy) * ASTERISK-26699 - res_pjsip: Assertion when sending OPTIONS request to endpoint (Reported by Ross Beer) * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) * ASTERISK-26754 - build_tools: make_build_h does not handle \ in user name (Reported by Kirill Katsnelson) * ASTERISK-26753 - AMI disconnect causes "ast_careful_fwrite: fwrite() returned error: Broken pipe" (Reported by Kirill Katsnelson) * ASTERISK-26755 - app_queue: Random queues disappear on "core reload queue all" (Reported by Kirill Katsnelson) * ASTERISK-26735 - res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect (Reported by Michael Maier) * ASTERISK-26693 - res_pjsip_endpoint_identifier_ip: Add support for SRV (Reported by Joshua Colp) * ASTERISK-26743 - PJPROJECT: Detecting compiled max log level does not work. (Reported by Richard Mudgett) * ASTERISK-26740 - voicemail API test: uses varlibdir instead of datadir for a sound file (Reported by Tzafrir Cohen) * ASTERISK-26739 - voicemail API test: confuses expected and actual values (Reported by Tzafrir Cohen) * ASTERISK-26731 - res_sorcery_memory_cache: memory leak on every sorcery memory cache populate (Reported by Ustinov Artem) * ASTERISK-26710 - [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0 (Reported by Aaron An) * ASTERISK-26672 - Crash when setting remote address on RTP instance (Reported by Richard Mudgett) * ASTERISK-26670 - [patch] Outgoing SIP-URI Dialing via PJSIP (Reported by Alexander Traud) * ASTERISK-26691 - Remember SDP negotiation on SIP_CODEC_INBOUND. (Reported by Alexander Traud) * ASTERISK-26673 - chan_pjsip: Crash when using CHANNEL dialplan function around masquerade (Reported by Joshua Colp) * ASTERISK-26684 - res_pjsip: Various issues with compact SIP headers (Reported by Joshua Elson) * ASTERISK-26655 - [patch]pjsip: Transfers Broken with Compact Headers Enabled (Reported by JoshE) * ASTERISK-26621 - app_queue: Queue application does not ring members with Local interface (Reported by Jonas Kellens) * ASTERISK-26586 - chan_sip: Segfaults upon reload if client with MWI wasn't registered (Reported by Michael Kuron) * ASTERISK-25494 - build: GCC 5.1.x catches some new const, array bounds and missing paren issues (Reported by George Joseph) * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring is invalid (Reported by Rusty Newton) * ASTERISK-25083 - Message.c: Message channel becomes saturated with frames leading to spammy log messages (Reported by Jonathan Rose) * ASTERISK-26653 - pjproject_bundled doesn't verify already downloaded tarballs (Reported by George Joseph) * ASTERISK-26433 - chan_sip: Allows To-tag checks to be bypassed, setting up new calls (Reported by Walter Doekes) * ASTERISK-26579 - codec_opus: Recursiveness when parsing fmtp line (Reported by Jørgen H) * ASTERISK-26644 - PJSIPShowRegistrationsInbound just dumps all aors (Reported by George Joseph) * ASTERISK-26490 - res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_" (Reported by Juris Breicis) * ASTERISK-26617 - res_rtp_asterisk: Can't bind on systems without IPv6 (Reported by Guido Falsi) * ASTERISK-24330 - Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118 (Reported by Marek Cervenka) * ASTERISK-26546 - mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2' (Reported by Tzafrir Cohen) * ASTERISK-26566 - res_rtp_asterisk: RTT miscalculation in RTCP (Reported by Hector Royo Concepcion) * ASTERISK-26604 - chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc. (Reported by Michael Kuron) * ASTERISK-26603 - [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no (Reported by Alexei Gradinari) * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) Improvements made in this release: ----------------------------------- * ASTERISK-23828 - pjsip - Need a command to list active SIP subscriptions (Reported by Rusty Newton) * ASTERISK-26527 - Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec (Reported by Badalian Vyacheslav) * ASTERISK-26624 - res_calendar_caldav: Add support for gmail (Reported by Eduardo Scudeller Libardi) * ASTERISK-26562 - app_controlplayback: Transmit Silence on ControlPlayback pause (Reported by Mikheili Dautashvili) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.14.0 Thank you for your continued support of Asterisk! -----
2017-05-11Update comms/gammu to 1.38.2leot3-9/+9
Changes: 20170328 - 1.38.2 [-] * Improved support for Huawei K3765, E150 and E372. [-] * Fixed decoding of unicode surrogates at message boundary. [+] * Environment variable PHONE_ID for external program. [-] * SMS compatibility with devices following old version of GSM 03.38. [-] * Unicode is now preferred when handling USSD. [+] * Improved decoding of MMS indication SMS. 20170105 - 1.38.1 [-] * Fixed sending SMS to numbers starting with 000. [-] * Fixed parsing of vcalendar files with VALUE=DATE-TIME. [-] * Fixed compatibility with D-Link dwm-157. [-] * Updated list of GSM countries and networks. 20161212 - 1.38.0 [-] * MySQL script for SMSD is compatible with strict mode. [-] * Fixed USSD responses for some AT modems. [-] * Fixed parsing network status for some modems (eg. Quectel UC15). [-] * Fixed handling of emojis and other Unicode chars from supplementary plan. [-] * Fixed compilation with C90 compiler.
2017-05-09Requires termcap.jperkin1-1/+2
2017-05-07Remove patch that has no effect.wiz2-11/+1
2017-04-30Recursive revbump from boost updateryoon5-10/+10
2017-04-22Revbump after icu updateadam14-28/+28
2017-04-18Updated minicom to 2.7.1.wiz2-10/+9
New for version 2.7.1: - CVE-2017-7467: Fix an out of bounds data access that can lead to remote code execution. This issue was found by Solar Designer of Openwall during a security audit of the Virtuozzo 7 product, which contains derived downstream code in its prl-vzvncserver component. The corresponding Virtuozzo 7 fix is: https://src.openvz.org/projects/OVZ/repos/prl-vzvncserver/commits/6d95404e75b98f36b1cc85ee23df99dcf06ca13f Openwall would like to thank the Virtuozzo company for funding the effort.
2017-04-13Update DeforaOS Phone to version 0.5.1khorben3-11/+11
This release brings: - parameter database for mobile data access - additional USSD codes for T-Mobile (Germany) - build fixes
2017-04-04Updated py-colorama to 0.3.7.wiz2-7/+7
0.3.7 * Fix issue #84: check if stream has 'closed' attribute before testing it * Fix issue #74: objects might become None at exit 0.3.6 * Fix issue #81: fix ValueError when a closed stream was used 0.3.5 * Bumping version to re-upload a wheel distribution 0.3.4 * Fix issue #47 and #80 - stream redirection now strips ANSI codes on Linux * Fix issue #53 - strip readline markers * Fix issue #32 - assign orig_stdout and orig_stderr when initialising * Fix issue #57 - Fore.RESET did not reset style of LIGHT_EX colors. Fixed by Andy Neff * Fix issue #51 - add context manager syntax. Thanks to Matt Olsen. * Fix issue #48 - colorama didn't work on Windows when environment variable 'TERM' was set. * Fix issue #54 - fix pylint errors in client code. * Changes to readme and other improvements by Marc Abramowitz and Zearin 0.3.3 * Fix Google Code issue #13 - support changing the console title with OSC escape sequence * Fix Google Code issue #16 - Add support for Windows xterm emulators * Fix Google Code issue #30 - implement \033[nK (clear line) * Fix Google Code issue #49 - no need to adjust for scroll when new position is already relative (CSI n A\B\C\D) * Fix Google Code issue #55 - erase_data fails on Python 3.x * Fix Google Code issue #46 - win32.COORD definition missing * Implement \033[0J and \033[1J (clear screen options) * Fix default ANSI parameters * Fix position after \033[2J (clear screen) * Add command shortcuts: colorama.Cursor, colorama.ansi.set_title, colorama.ansi.clear_line, colorama.ansi.clear_screen * Fix issue #22 - Importing fails for python3 on Windows * Thanks to John Szakmeister for adding support for light colors * Thanks to Charles Merriam for adding documentation to demos
2017-03-31Recursive bump for gpgme update which removed a support library.wiz1-2/+2
2017-02-21Add an upper API version restriction.cherry1-2/+3
The current only user of this buildlink file is asterisk-chan-dongle (which is yet to be committed). With further users, comms/asterisk may need to find a version specific directory as newer versions are imported.
2017-02-17Don't define accept4 locally on new enough NetBSD current.joerg2-1/+22
2017-02-17Add missing includes.joerg3-9/+23
2017-02-12Recursive revbump from fonts/harfbuzzryoon15-30/+30
2017-02-10Add buildlink support.cherry1-0/+12
This will aid subsequent module builds
2017-02-10Um, need bsd.prefs.mk before testing ${OPSYS}.he1-3/+4
2017-02-10Don't enable the inet6 option on the various BSDs, since their stackhe2-4/+10
require separate inet6 and inet sockets, and conserver as of 8.2.1 doesn't do that. Bump PKGREVISION.
2017-02-06Recursive bump for harfbuzz's new graphite2 dependency.wiz15-32/+30
2017-01-19Convert all occurrences (353 by my count) ofagc5-19/+19
MASTER_SITES= site1 \ site2 style continuation lines to be simple repeated MASTER_SITES+= site1 MASTER_SITES+= site2 lines. As previewed on tech-pkg. With thanks to rillig for fixing pkglint accordingly.
2017-01-18Add two patches so that this at least semi-works when the inet6he6-8/+100
option is used: * Use correct sockaddr length when doing getnameinfo() for inet6, so we avoid an early return with "permanent failure" from getnameinfo() * Use temp variables for walking the address lists so that we avoid trying freeaddrinfo(NULL) and getting SEGV This still isn't fully baked and backward compatible: with the inet6 option turned on, on NetBSD the conserver process only opens an inet6 server socket and no longer serves an inet socket (a Linuxism, I suspect), making it troublesome to interoperate with older versions of conserver or installations on hosts without IPv6 connectivity. PKGREVISION bumped.
2017-01-01Revbump after boost updateadam5-8/+10
2017-01-01Add python-3.6 to incompatible versions.wiz2-4/+4
2016-12-12Revert "Specify readline requirement on 30 packages"wiz1-2/+1
Many of these definitely do not depend on readline. So there must be a different underlying problem, and that should be tracked down instead of papering over it.
2016-12-11Update to Asterisk 11.25.1: this fixes AST-2016-009.jnemeth2-12/+11
Asterisk Project Security Advisory - ASTERISK-2016-009 Product Asterisk Summary Nature of Advisory Authentication Bypass Susceptibility Remote unauthenticated sessions Severity Minor Exploits Known No Reported On October 3, 2016 Reported By Walter Doekes Posted On Last Updated On December 8, 2016 Advisory Contact Mmichelson AT digium DOT com CVE Name Description The chan_sip channel driver has a liberal definition for whitespace when attempting to strip the content between a SIP header name and a colon character. Rather than following RFC 3261 and stripping only spaces and horizontal tabs, Asterisk treats any non-printable ASCII character as if it were whitespace. This means that headers such as Contact\x01: will be seen as a valid Contact header. This mostly does not pose a problem until Asterisk is placed in tandem with an authenticating SIP proxy. In such a case, a crafty combination of valid and invalid To headers can cause a proxy to allow an INVITE request into Asterisk without authentication since it believes the request is an in-dialog request. However, because of the bug described above, the request will look like an out-of-dialog request to Asterisk. Asterisk will then process the request as a new call. The result is that Asterisk can process calls from unvetted sources without any authentication. If you do not use a proxy for authentication, then this issue does not affect you. If your proxy is dialog-aware (meaning that the proxy keeps track of what dialogs are currently valid), then this issue does not affect you. If you use chan_pjsip instead of chan_sip, then this issue l does not affect you. Resolution chan_sip has been patched to only treat spaces and horizontal tabs as whitespace following a header name. This allows for Asterisk and authenticating proxies to view requests the same way Affected Versions Product Release Series Asterisk Open Source 11.x All Releases Asterisk Open Source 13.x All Releases Asterisk Open Source 14.x All Releases Certified Asterisk 13.8 All Releases Corrected In Product Release Asterisk Open Source 11.25.1, 13.13.1, 14.2.1 Certified Asterisk 11.6-cert16, 13.8-cert4 Patches SVN URL Revision Links Asterisk Project Security Advisories are posted at http://www.asterisk.org/security This document may be superseded by later versions; if so, the latest version will be posted at http://downloads.digium.com/pub/security/ASTERISK-2016-009.pdf and http://downloads.digium.com/pub/security/ASTERISK-2016-009.html Revision History Date Editor Revisions Made November 28, 2016 Mark Michelson Initial writeup Asterisk Project Security Advisory - ASTERISK-2016-009 Copyright (c) 2016 Digium, Inc. All Rights Reserved. Permission is hereby granted to distribute and publish this advisory in its original, unaltered form.
2016-12-09Update comms/py-gammu to py-gammu-2.7leot2-7/+7
Changes: 2.7 === * Needs Gammu >= 1.37.90 due to API changes. 2.6 === * Fixed error when creating new contact. * Fixed possible testsuite errors.
2016-12-09Update comms/gammu to gammu-1.37.91leot4-15/+14
Changes: 20161023 - 1.37.91 [!] * Changed version of the shared library. [-] * Improved support for ZTE MF100. [-] * Ignore unsolicited +CLCC: reply. [-] * Correctly report when some SMSD SQL backend is not compiled in. [-] * Fix build of MySQL backend on Linux. 20161018 - 1.37.90 [-] * Improved support Huawei K3770. [!] * API changes in some parameter types. [-] * Fixed various Windows compilation issues. [-] * Fixed several resource leaks. [-] * Create outbox SMS atomically in FILES backend. [!] * Removed getlocation command as we no longer fit into their usage policy. [-] * Fixed call diverts on TP-LINK MA260. [+] * Initial support for Oracle database. [!] * Removed unused daemons, pbk and pbk_groups tables from the SMSD schema. [+] * SMSD outbox entries now can have priority set in the database. [+] * Added SIM IMSI to the SMSD status table. [+] * Added CheckNetwork directive. [+] * SMSD attempts to power on radio if disabled. [-] * Fixed processing of AT unsolicited responses in some cases. [-] * Fixed parsing USSD responses from some devices. 20160816 - 1.37.4 [-] * Improved support for Huawei E3131. [-] * Fixed SMS support for MULTIBAND 900E. [-] * Fixed SMS created in text mode. 20160524 - 1.37.3 [-] * Improved support for Huawei E398. [-] * Improved support for Huawei/Vodafone K4505. [-] * Fixed possible crash if SMSD used in library. [-] * Improved support for Huawei E180. 20160413 - 1.37.2 [-] * Fixed compilation of SMSD. 20160413 - 1.37.1 [-] * Properly report errors in HEX encoded strings from SMSD SQL backends. [-] * Configurable SMSD table names. [-] * Improved support for Huawei E303. [-] * Improved support for Vodafone K4511. [-] * Improved support for Telit M2M modules.
2016-12-04Recursive revbump from textproc/icu 58.1ryoon14-25/+28
2016-12-04Specify readline requirement on 30 packagesmarino1-1/+2
Solves: /usr/libexec/binutils225/elf/ld.gold: error: cannot find -lreadline The missing specification is obvious on DragonFly because there's no publically accessible version of readline in base.
2016-12-03Correct the if statement to AND, not OR.sevan1-2/+2
Unbreak builds on FreeBSD & DragonFly BSD
2016-12-03Add dfu-util.sevan1-1/+2
2016-12-03Import dfu-util 0.9sevan4-0/+40
ok wiedi
2016-11-27Update to Asterisk 14.2.0: this is mostly a bugfix release with some minorjnemeth4-16/+22
improvements. pkgsrc change: adapt to new res_resolver_unbound module. The Asterisk Development Team has announced the release of Asterisk 14.2.0. The release of Asterisk 14.2.0 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: Improvements made in this release: ----------------------------------- * ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11) * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11) * ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell) * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan) * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP (Reported by Michael Walton) Bugs fixed in this release: ----------------------------------- * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) * ASTERISK-26520 - codec_opus: Generated fmtp line has no content (Reported by scgm11) * ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. (Reported by Richard Mudgett) * ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. (Reported by Richard Mudgett) * ASTERISK-26556 - manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes (Reported by Michelle Dupuis) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage (Reported by George Joseph) * ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set (Reported by Jason) * ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded. (Reported by Joshua Colp) * ASTERISK-26571 - res_pjsip: Resolution incorrect when explicit IPv6 transport configured (Reported by Joshua Colp) * ASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls (Reported by Daniele Pallastrelli) * ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by Dmitry Melekhov) * ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions (Reported by Matt Jordan) * ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George Joseph) * ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10 (Reported by Jonathan Harris) * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) * ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state (Reported by Joshua Colp) * ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used (Reported by Frankie Chin) * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types. (Reported by Alexander Traud) * ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c (Reported by Matt Krokosz) * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele Giacone) * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey Grachev) * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum (Reported by Joshua Colp) * ASTERISK-26537 - AMI: NewConnectedLine event is not documented (Reported by Etienne Lessard) * ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy (Reported by Badalian Vyacheslav) * ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled. (Reported by Corey Farrell) * ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian Gilmour) * ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls. (Reported by Harley Peters) * ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance (Reported by Joshua Colp) * ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf (Reported by Rusty Newton) * ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph) * ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan) * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported by Andreas Wetzel) * ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations. (Reported by Alexander Traud) * ASTERISK-26482 - [patch] chan_pjsip: segfault on already disconnected session (Reported by Alexei Gradinari) * ASTERISK-26421 - Segmentation Fault with ARI originate into mixing bridge with 43 clients (Reported by Andrew Nagy) * ASTERISK-26444 - 'features show' command in CLI does not return prompt. (Reported by John Kiniston) * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud) * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir Cohen) * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage (Reported by Leandro Dardini) * ASTERISK-26439 - chan_rtp: Crash when originating (Reported by Kayode) * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud) * ASTERISK-26618 - build: Backport addition of librt check to configure.ac (Reported by Kevin Harwell) New Features made in this release: ----------------------------------- * ASTERISK-26595 - ARI: Add the ability to control the source of video in a multi-party mixing bridge (Reported by Matt Jordan) * ASTERISK-26492 - ARI: Add ability to specify channel variables on websocket events (Reported by Mark Michelson) * ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing events (Reported by Matt Jordan) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.2.0 Thank you for your continued support of Asterisk!
2016-11-27Update to Asterisk 13.13.0: this is mainly a bug fix release with somejnemeth3-15/+15
minor improvements. The Asterisk Development Team has announced the release of Asterisk 13.13.0. The release of Asterisk 13.13.0 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: New Features made in this release: ----------------------------------- * ASTERISK-26595 - ARI: Add the ability to control the source of video in a multi-party mixing bridge (Reported by Matt Jordan) * ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing events (Reported by Matt Jordan) Bugs fixed in this release: ----------------------------------- * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported by snuffy) * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid manipulation through agi (Reported by Morten Tryfoss) * ASTERISK-26520 - codec_opus: Generated fmtp line has no content (Reported by scgm11) * ASTERISK-26605 - codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded. (Reported by Richard Mudgett) * ASTERISK-26516 - pjsip: Memory corruption with possible memory leak. (Reported by Richard Mudgett) * ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage (Reported by George Joseph) * ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set (Reported by Jason) * ASTERISK-26575 - testsuite: Need to check PJSIP functionality when res_srtp is not loaded. (Reported by Joshua Colp) * ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by Dmitry Melekhov) * ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11 regressions (Reported by Matt Jordan) * ASTERISK-26412 - build: Prepare for gcc 6.2 (Reported by George Joseph) * ASTERISK-26509 - A few non-critical deprecation warnings when building on Ubuntu 16.10 (Reported by Jonathan Harris) * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) * ASTERISK-26468 - ari: Bridge events stop working after this sequence of ARI calls (Reported by Daniele Pallastrelli) * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic payload types. (Reported by Alexander Traud) * ASTERISK-26549 - app_dial: When PickupChan() is used some channels may have incorrect device state (Reported by Joshua Colp) * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats to maximum (Reported by Joshua Colp) * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele Giacone) * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings" (Reported by Sergey Grachev) * ASTERISK-26537 - AMI: NewConnectedLine event is not documented (Reported by Etienne Lessard) * ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy (Reported by Badalian Vyacheslav) * ASTERISK-26524 - astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled. (Reported by Corey Farrell) * ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian Gilmour) * ASTERISK-26387 - Asterisk segfaults shortly after starting even with no active calls. (Reported by Harley Peters) * ASTERISK-26514 - Super Awesome Company: Don't specify transport in pjsip.conf (Reported by Rusty Newton) * ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance (Reported by Joshua Colp) * ASTERISK-26510 - pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions (Reported by George Joseph) * ASTERISK-22480 - Embedded pjproject: build.mak contains hardcoded full path to version.mak (Reported by Matt Jordan) * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change (Reported by Bill Brigden) * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness (Reported by Andreas Wetzel) * ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations. (Reported by Alexander Traud) * ASTERISK-26421 - Segmentation Fault with ARI originate into mixing bridge with 43 clients (Reported by Andrew Nagy) * ASTERISK-26444 - 'features show' command in CLI does not return prompt. (Reported by John Kiniston) * ASTERISK-26482 - [patch] chan_pjsip: segfault on already disconnected session (Reported by Alexei Gradinari) * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud) * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir Cohen) * ASTERISK-26439 - chan_rtp: Crash when originating (Reported by Kayode) * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage (Reported by Leandro Dardini) * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud) * ASTERISK-26618 - build: Backport addition of librt check to configure.ac (Reported by Kevin Harwell) Improvements made in this release: ----------------------------------- * ASTERISK-25063 - [patch]add X.509 subject alternative name support to Asterisk TLS support (Reported by Maciej Szmigiero) * ASTERISK-26558 - app_queue: add variable to know if the call is not answered after a queue (Reported by scgm11) * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry (Reported by scgm11) * ASTERISK-26538 - codec_opus: Add sample to configs/samples/codecs.conf.sample (Reported by Kevin Harwell) * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands (Reported by Matt Jordan) * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP (Reported by Michael Walton) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.13.0 Thank you for your continued support of Asterisk!
2016-11-27Update to Asterisk 11.25.0: this is a bug fix release.jnemeth2-11/+11
The Asterisk Development Team has announced the release of Asterisk 11.25.0. The release of Asterisk 11.25.0 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: Bugs fixed in this release: ----------------------------------- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module (Reported by Alexander Traud) * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir Cohen) * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage (Reported by Leandro Dardini) * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection triggered. (Reported by Alexander Traud) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0 Thank you for your continued support of Asterisk!
2016-11-24Update doxygen-depend version to 1.8.12 (or add new BUILD_DEPENDS+)mef1-2/+2
2016-11-24Adjust PLIST for doxygen update 1.8.11 to 1.8.12, PKGREVISION++.mef2-5/+5
2016-11-11Update to Asterisk 14.1.2: this is a critical bug fix release.jnemeth2-11/+11
The Asterisk Development Team has announced the release of Asterisk 14.1.2. The release of Asterisk 14.1.2 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.2 Thank you for your continued support of Asterisk!
2016-11-11Update the Asterisk 13.12.2: this is a critical bug fix release.jnemeth2-11/+11
The Asterisk Development Team has announced the release of Asterisk 13.12.2. The release of Asterisk 13.12.2 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression (Reported by Michael Keuter) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.2 Thank you for your continued support of Asterisk!
2016-10-29Update to Asterisk 13.12.1: this is a critical bug fix release.jnemeth2-11/+11
The Asterisk Development Team has announced the release of Asterisk 13.12.1. The release of Asterisk 13.12.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.1 Thank you for your continued support of Asterisk!
2016-10-28Update to Asterisk 14.1.1: this is a critical bug fix release.jnemeth2-11/+11
The Asterisk Development Team has announced the release of Asterisk 14.1.1. The release of Asterisk 14.1.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.1 Thank you for your continued support of Asterisk!
2016-10-28Update to Asterisk 11.24.1: this is a critical bug fix release.jnemeth2-11/+11
The Asterisk Development Team has announced the release of Asterisk 11.24.1. The release of Asterisk 11.24.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following is the issue resolved in this release: Bugs fixed in this release: ----------------------------------- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used (Reported by Doug Lytle) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1 Thank you for your continued support of Asterisk!
2016-10-27Update to Asterisk 14.1.0: this is mostly a bug fix release.jnemeth5-33/+20
The Asterisk Development Team has announced the release of Asterisk 14.1.0. The release of Asterisk 14.1.0 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: New Features made in this release: ----------------------------------- * ASTERISK-26277 - Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment (Reported by Matt Jordan) Bugs fixed in this release: ----------------------------------- * ASTERISK-26477 - pjproject: SEGV during SSL operations (Reported by George Joseph) * ASTERISK-17470 - [patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio. (Reported by effie mouzeli) * ASTERISK-26416 - pjproject-bundled: configure fails to check for all required utilities (Reported by Corey Farrell) * ASTERISK-26466 - core: Be forgiving on external callerid that may be flawed so we don't drop events (Reported by Richard Mudgett) * ASTERISK-26362 - res_config_mysql: Broken after 13.10 (Reported by Carlos Chavez) * ASTERISK-26446 - app_dial: There's no way to override the hangupcause on unanswered channels (Reported by George Joseph) * ASTERISK-26410 - core: Asterisk 14 doesn't show the header in the console or verbose when starting (Reported by Dan Jenkins) * ASTERISK-24311 - Populating database via Alembic fails when using same database for multiple schema sets (Reported by Dafi Ni) * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. (Reported by Alexander Traud) * ASTERISK-26426 - format_ogg_opus: remove from source (Reported by Kevin Harwell) * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::] (Reported by Jacek) * ASTERISK-25468 - Deadlock in chan_sip - core show locks shows do_monitor lock (Reported by Barry Flanagan) * ASTERISK-26397 - manager: PresenceState action crashes Asterisk 14 (Reported by Andrew Nagy) * ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by Joshua Colp) * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed (Reported by Tzafrir Cohen) * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled (Reported by Anthony Messina) * ASTERISK-26391 - Consoles do not display verbose logger messages even when requested. (Reported by Marcelo Terres) * ASTERISK-26263 - SQL error when using realtime and registering extension / inserting into ps_contacts (Reported by Jeppe Ryskov Larsen) * ASTERISK-26365 - rtp: Offer with multiple payloads for same codec is incorrectly handled (Reported by Joshua Colp) * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra) * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten for connectionful protocols (Reported by Joshua Colp) * ASTERISK-26367 - rtp: Timestamps broken when video frame is across multiple RTP packets (Reported by Joshua Colp) * ASTERISK-26375 - res_pjsip_transport_management: Log message states seconds, but time value is milliseconds (Reported by Joshua Colp) * ASTERISK-26364 - res_pjsip: Don't assume a request will have target addresses (Reported by Joshua Colp) * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. (Reported by Richard Mudgett) * ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes unexpected callerid (Reported by Kevin Harwell) * ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed (Reported by Dmitry Melekhov) * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by Etienne Lessard) * ASTERISK-26264 - res_pjsip: Crash when applying ACL from non-existent endpoint (Reported by nappsoft) * ASTERISK-26341 - ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list (Reported by Matt Jordan) * ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up. (Reported by Etienne Lessard) * ASTERISK-26331 - Crash on "core show channeltype Surrogate" in ast_format_cap_get_names (Reported by CGI.NET) * ASTERISK-26269 - res_pjsip: Wrong state for aors without registered contacts after startup (Reported by nappsoft) * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's' extension" (Reported by chris de rock) * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts (Reported by Etienne Lessard) * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard) * ASTERISK-26279 - pjproject-bundled: Fails to compile on Debian 6 (Reported by George Joseph) * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up properly (Reported by Alexander Traud) * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels (Reported by Etienne Lessard) * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code (Reported by David Brillert) * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels. (Reported by Richard Mudgett) * ASTERISK-26288 - followme: fails to reset config items to default values on reload (Reported by Tzafrir Cohen) * ASTERISK-22374 - Finish mapping the sip.conf parameters to res_sip.conf parameters (Reported by Matt Jordan) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include annexb=no attribute. (Reported by Ali Ghavidel) * ASTERISK-25472 - Swagger scripts are not replacing format variable in file brief (Reported by Corey Farrell) * ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it (Reported by József Dudás) * ASTERISK-26305 - Asterisk 14: Two resolver unbound testsuite tests fail (Reported by Richard Mudgett) * ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities not detected in PJProject. (Reported by Alexander Traud) * ASTERISK-25492 - ARI: Path parameters are case sensitive (Reported by Joshua Colp) * ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via chan_pjsip (Reported by Ross Beer) * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names (Reported by Corey Farrell) * ASTERISK-26246 - Security: Privilege escalation by AMI adding dialplan extensions. (Reported by Richard Mudgett) * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed startup. (Reported by Corey Farrell) * ASTERISK-26241 - res_pjsip: When using compact headers, rpid and pai are incorrectly generated (Reported by George Joseph) * ASTERISK-25797 - app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension (Reported by Etienne Lessard) * ASTERISK-26239 - res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname (Reported by George Joseph) * ASTERISK-26238 - res_pjsip: Empty global default_from_user causes crash (Reported by Joshua Colp) * ASTERISK-26268 - alembic: 'auth_username' not in PJSIP 'identify_by' enum (Reported by Joshua Colp) * ASTERISK-26253 - sdp_srtp: libsrtp now a required dependency, shouldn't be (Reported by Ben Merrills) * ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade + indicate (Reported by Ross Beer) * ASTERISK-26183 - alembic: error when using sqlalchemy version 1.1.0b2 (Reported by Kevin Harwell) * ASTERISK-26283 - res_resolver_unbound: fails configure on older Ubuntu and CentOS (Reported by George Joseph) * ASTERISK-26280 - DNS lookups can block channel media paths (Reported by Mark Michelson) * ASTERISK-26278 - asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM. (Reported by Corey Farrell) * ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c (Reported by Richard Mudgett) * ASTERISK-26265 - Errors ignored from some parts of system initialization. (Reported by Corey Farrell) * ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex for get all (Reported by Dmitry) * ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets with IP6 (Reported by Alexander Traud) * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported by Andrew Nagy) * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..." (Reported by Hans van Eijsden) * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard Mudgett) Improvements made in this release: ----------------------------------- * ASTERISK-26409 - codec_opus: Update Asterisk to support the l translation codec. (Reported by Kevin Harwell) * ASTERISK-26289 - Announcer channels in ConfBridges cause inefficiencies (Reported by Mark Michelson) * ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used (Reported by Alexei Gradinari) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.0 Thank you for your continued support of Asterisk!
2016-10-27Update to Asterisk 13.12.0: this is mostly a bug fix release.jnemeth4-20/+20
The Asterisk Development Team has announced the release of Asterisk 13.12.0. The release of Asterisk 13.12.0 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: New Features made in this release: ----------------------------------- * ASTERISK-26277 - Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment (Reported by Matt Jordan) Bugs fixed in this release: ----------------------------------- * ASTERISK-26477 - pjproject: SEGV during SSL operations (Reported by George Joseph) * ASTERISK-17470 - [patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio. (Reported by effie mouzeli) * ASTERISK-26416 - pjproject-bundled: configure fails to check for all required utilities (Reported by Corey Farrell) * ASTERISK-26466 - core: Be forgiving on external callerid that may be flawed so we don't drop events (Reported by Richard Mudgett) * ASTERISK-26362 - res_config_mysql: Broken after 13.10 (Reported by Carlos Chavez) * ASTERISK-26446 - app_dial: There's no way to override the hangupcause on unanswered channels (Reported by George Joseph) * ASTERISK-24311 - Populating database via Alembic fails when using same database for multiple schema sets (Reported by Dafi Ni) * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. (Reported by Alexander Traud) * ASTERISK-26426 - format_ogg_opus: remove from source (Reported by Kevin Harwell) * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::] (Reported by Jacek) * ASTERISK-25468 - Deadlock in chan_sip - core show locks shows do_monitor lock (Reported by Barry Flanagan) * ASTERISK-26397 - manager: PresenceState action crashes Asterisk 14 (Reported by Andrew Nagy) * ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by Joshua Colp) * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed (Reported by Tzafrir Cohen) * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled (Reported by Anthony Messina) * ASTERISK-26263 - SQL error when using realtime and registering extension / inserting into ps_contacts (Reported by Jeppe Ryskov Larsen) * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten for connectionful protocols (Reported by Joshua Colp) * ASTERISK-26367 - rtp: Timestamps broken when video frame is across multiple RTP packets (Reported by Joshua Colp) * ASTERISK-26375 - res_pjsip_transport_management: Log message states seconds, but time value is milliseconds (Reported by Joshua Colp) * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra) * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. (Reported by Richard Mudgett) * ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes unexpected callerid (Reported by Kevin Harwell) * ASTERISK-26349 - 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed (Reported by Dmitry Melekhov) * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by Etienne Lessard) * ASTERISK-26264 - res_pjsip: Crash when applying ACL from non-existent endpoint (Reported by nappsoft) * ASTERISK-26288 - followme: fails to reset config items to default values on reload (Reported by Tzafrir Cohen) * ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up. (Reported by Etienne Lessard) * ASTERISK-26331 - Crash on "core show channeltype Surrogate" in ast_format_cap_get_names (Reported by CGI.NET) * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's' extension" (Reported by chris de rock) * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts (Reported by Etienne Lessard) * ASTERISK-26279 - pjproject-bundled: Fails to compile on Debian 6 (Reported by George Joseph) * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up properly (Reported by Alexander Traud) * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard) * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels (Reported by Etienne Lessard) * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code (Reported by David Brillert) * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels. (Reported by Richard Mudgett) * ASTERISK-26269 - res_pjsip: Wrong state for aors without registered contacts after startup (Reported by nappsoft) * ASTERISK-22374 - Finish mapping the sip.conf parameters to res_sip.conf parameters (Reported by Matt Jordan) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK-25472 - Swagger scripts are not replacing format variable in file brief (Reported by Corey Farrell) * ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include annexb=no attribute. (Reported by Ali Ghavidel) * ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it (Reported by József Dudás) * ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities not detected in PJProject. (Reported by Alexander Traud) * ASTERISK-25492 - ARI: Path parameters are case sensitive (Reported by Joshua Colp) * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names (Reported by Corey Farrell) * ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via chan_pjsip (Reported by Ross Beer) * ASTERISK-26246 - Security: Privilege escalation by AMI adding dialplan extensions. (Reported by Richard Mudgett) * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed startup. (Reported by Corey Farrell) * ASTERISK-26241 - res_pjsip: When using compact headers, rpid and pai are incorrectly generated (Reported by George Joseph) * ASTERISK-26239 - res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname (Reported by George Joseph) * ASTERISK-26238 - res_pjsip: Empty global default_from_user causes crash (Reported by Joshua Colp) * ASTERISK-25797 - app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension (Reported by Etienne Lessard) * ASTERISK-26268 - alembic: 'auth_username' not in PJSIP 'identify_by' enum (Reported by Joshua Colp) * ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade + indicate (Reported by Ross Beer) * ASTERISK-26183 - alembic: error when using sqlalchemy version 1.1.0b2 (Reported by Kevin Harwell) * ASTERISK-26280 - DNS lookups can block channel media paths (Reported by Mark Michelson) * ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c (Reported by Richard Mudgett) * ASTERISK-26265 - Errors ignored from some parts of system initialization. (Reported by Corey Farrell) * ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex for get all (Reported by Dmitry) * ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets with IP6 (Reported by Alexander Traud) * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported by Andrew Nagy) * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..." (Reported by Hans van Eijsden) Improvements made in this release: ----------------------------------- * ASTERISK-26409 - codec_opus: Update Asterisk to support the translation codec. (Reported by Kevin Harwell) * ASTERISK-26289 - Announcer channels in ConfBridges cause inefficiencies (Reported by Mark Michelson) * ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used (Reported by Alexei Gradinari) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.0 Thank you for your continued support of Asterisk!
2016-10-26Update to Asterisk 11.24.0: this is a bug fix release.jnemeth3-12/+49
The Asterisk Development Team has announced the release of Asterisk 11.24.0. The release of Asterisk 11.24.0 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: Bugs fixed in this release: ----------------------------------- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response. (Reported by Alexander Traud) * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::] (Reported by Jacek) * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed (Reported by Tzafrir Cohen) * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron Hamstra) * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs. (Reported by Richard Mudgett) * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by Etienne Lessard) * ASTERISK-26288 - followme: fails to reset config items to default values on reload (Reported by Tzafrir Cohen) * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's' extension" (Reported by chris de rock) * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts (Reported by Etienne Lessard) * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members with Local interface (Reported by Etienne Lessard) * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up properly (Reported by Alexander Traud) * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels (Reported by Etienne Lessard) * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code (Reported by David Brillert) * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels. (Reported by Richard Mudgett) * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have multiple native formats (Reported by Matt Jordan) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change) (Reported by Krzysztof Trempala) * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names (Reported by Corey Farrell) * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed startup. (Reported by Corey Farrell) * ASTERISK-26265 - Errors ignored from some parts of system initialization. (Reported by Corey Farrell) * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported by Andrew Nagy) * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard Mudgett) * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and attended transfer handling (Reported by Ben Smithurst) * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional code. (Reported by Corey Farrell) * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even on lost packets. (Reported by Alexander Traud) * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files (Reported by Tzafrir Cohen) * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by Richard Miller) * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported by Corey Farrell) * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure occurred on RTP instance (Reported by Edwin Vandamme) * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by Alexander Traud) * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when building menuselect (Reported by Jeffrey Walton) * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and bugs (Reported by Alexei Gradinari) * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua Colp) * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George Joseph) Improvements made in this release: ----------------------------------- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by Corey Farrell) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0 Thank you for your continued support of Asterisk!
2016-10-25add and enable asterisk14jnemeth1-1/+2
generated by cgit v1.2.3 (git 2.39.1) at 2024-05-21 11:38:47 +0000