Initial import of Asterisk 13. It has been tested to compile
and run, but not a lot of functional testing. This does not have the new PJSIP, which will be coming in a followup commit. This also does not have the patches for compiling with Clang. For upgrading instructions, please see: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13 ----- The Asterisk Development Team is pleased to announce the release of Asterisk 13.0.0. Asterisk 13 is the next major release series of Asterisk. It is a Long Term Support (LTS) release, similar to Asterisk 11. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions For important information regarding upgrading to Asterisk 13, please see the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13 A short list of new features includes: * Asterisk security events are now provided via AMI, allowing end users to monitor their Asterisk system in real time for security related issues. * Both AMI and ARI now allow external systems to control the state of a mailbox. Using AMI actions or ARI resources, external systems can programmatically trigger Message Waiting Indicators (MWI) on subscribed phones. This is of particular use to those who want to build their own VoiceMail application using ARI. * ARI now supports the reception/transmission of out of call text messages using any supported channel driver/protocol stack through ARI. Users receive out of call text messages as JSON events over the ARI websocket connection, and can send out of call text messages using HTTP requests. * The PJSIP stack now supports RFC 4662 Resource Lists, allowing Asterisk to act as a Resource List Server. This includes defining lists of presence state, mailbox state, or lists of presence state/mailbox state; managing subscriptions to lists; and batched delivery of NOTIFY requests to subscribers. * The PJSIP stack can now be used as a means of distributing device state or mailbox state via PUBLISH requests to other Asterisk instances. This is analogous to Asterisk's clustering support using XMPP or Corosync; unlike existing clustering mechanisms, using the PJSIP stack to perform the distribution of state does not rely on another daemon or server to perform the work. And much more! More information about the new features can be found on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation A full list of all new features can also be found in the CHANGES file: http://svnview.digium.com/svn/asterisk/branches/13/CHANGES For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0 Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 13.1.0. The release of Asterisk 13.1.0 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-24554 - AMI/ARI: Generate events on connected line changes (Reported by Matt Jordan) Bugs fixed in this release: ----------------------------------- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0 (Reported by Patrick Laimbock) * ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by Corey Farrell) * ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing leak (Reported by Corey Farrell) * ASTERISK-24430 - missing letter "p" in word response in OriginateResponse event documentation (Reported by Dafi Ni) * ASTERISK-24437 - Review implementation of ast_bridge_impart for leaks and document proper usage (Reported by Scott Griepentrog) * ASTERISK-24453 - manager: acl_change_sub leaks (Reported by Corey Farrell) * ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by Corey Farrell) * ASTERISK-24458 - chan_phone fails to build on big endian systems (Reported by Tzafrir Cohen) * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers (Reported by Olle Johansson) * ASTERISK-24304 - asterisk crashing randomly because of unistim channel (Reported by dhanapathy sathya) * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by Nick Adams) * ASTERISK-24462 - res_pjsip: Stale qualify statistics after disablementation (Reported by Kevin Harwell) * ASTERISK-24465 - audiohooks list leaks reference to formats (Reported by Corey Farrell) * ASTERISK-24466 - app_queue: fix a couple leaks to struct call_queue (Reported by Corey Farrell) * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled (Reported by Corey Farrell) * ASTERISK-24411 - [patch] Status of outbound registration is not changed upon unregistering. (Reported by John Bigelow) * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream leaks (Reported by Corey Farrell) * ASTERISK-24480 - res_http_websockets: Module reference decrease below zero (Reported by Corey Farrell) * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in audiohook callback (Reported by Corey Farrell) * ASTERISK-24487 - configuration: sections should be loadable as template even when not marked (Reported by Scott Griepentrog) * ASTERISK-20127 - [Regression] Config.c config_text_file_load() unescapes semicolons ("\;" -> ";") turning them into comments (corruption) on rewrite of a config file (Reported by George Joseph) * ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload when DNS settings invalid (Reported by Melissa Shepherd) * ASTERISK-24307 - Unintentional memory retention in stringfields (Reported by Etienne Lessard) * ASTERISK-24491 - Memory leak in res_hep (Reported by Zane Conkle) * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes extra calls to ast_module_unref (Reported by Corey Farrell) * ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when waiting for more matching digits. (Reported by Richard Mudgett) * ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to queue caller (Reported by Steve Pitts) * ASTERISK-24504 - chan_console: Fix reference leaks to pvt (Reported by Corey Farrell) * ASTERISK-24250 - [patch] Voicemail with multi-recipients To: header fix (Reported by abelbeck) * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS length exceeds 50 (roughly) national symbols (Reported by Dmitriy Bubnov) * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN revision r227276 (Reported by Xavier Hienne) * ASTERISK-24505 - manager: http connections leak references (Reported by Corey Farrell) * ASTERISK-24502 - Build fails when dev-mode, dont optimize and coverage are enabled (Reported by Corey Farrell) * ASTERISK-24444 - PBX: Crash when generating extension for pattern matching hint (Reported by Leandro Dardini) * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1 (Reported by Gregory Malsack) * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended transfer (Reported by Beppo Mazzucato) * ASTERISK-24501 - ARI: Moving a channel between bridges followed by a hangup can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd (Reported by Matt Jordan) * ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash (Reported by Leon Rowland) * ASTERISK-23651 - Reloading some modules that are loaded already, results in 'No such module' before a successful reload (Reported by Rusty Newton) * ASTERISK-24522 - ConfBridge: delay occurs between kicking all endmarked users when last marked user leaves (Reported by Matt Jordan) * ASTERISK-15242 - transmit_refer leaks sip_refer structures (Reported by David Woolley) * ASTERISK-24508 - pjsip - REFER request from SNOM is rejected with "400 bad request" - DEBUG shows "Received a REFER without a parseable Refer-To" (Reported by Beppo Mazzucato) * ASTERISK-24535 - stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix (Reported by Corey Farrell) * ASTERISK-24471 - Crash - assert_fail in libc in pjmedia_sdp_neg_negotiateofrom /usr/local/lib/libpjmedia.so.2 (Reported by yaron nahum) * ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces in-dialog with invalid target causes crash (Reported by Joshua Colp) * ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial module load (Reported by Matt Jordan) * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs allow blocked addresses through (Reported by Matt Jordan) * ASTERISK-24542 - [patch]Failure showing codecs via 'core show channeltype <tech>' (Reported by snuffy) * ASTERISK-24533 - 2 threads created per chan_sip entry (Reported by xrobau) * ASTERISK-24516 - [patch]Asterisk segfaults when playing back voicemail under high concurrency with an IMAP backend (Reported by David Duncan Ross Palmer) * ASTERISK-24572 - [patch]App_meetme is loaded without its defaults when the configuration file is missing (Reported by Nuno Borges) * ASTERISK-24573 - [patch]Out of sync conversation recording when divided in multiple recordings (Reported by NunowBorges) * ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not reliably transmitted during transfers (Reported by Matt Jordan) * ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip extension to another pjsip extension (Reported by Abhay Gupta) Improvements made in this release: ----------------------------------- * ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR property 'unanswered' (Reported by Matt Jordan) * ASTERISK-24283 - [patch]Microseconds precision in the eventtime column in the cel_odbc module (Reported by Etienne Lessard) * ASTERISK-24530 - [patch] app_record stripping 1/4 second from recordings (Reported by Ben Smithurst) * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded lookups (Reported by Birger "WIMPy" Harzenetter) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0 Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 13.2.0. The release of Asterisk 13.2.0 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: Bugs fixed in this release: ----------------------------------- * ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them all at the same time. (Reported by Richard Mudgett) * ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard (Reported by Kevin Harwell) * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS from JSSIP (Reported by Badalian Vyacheslav) * ASTERISK-24607 - res_pjsip_session: re-INVITE with declined media streams results in 488 (Reported by Matt Jordan) * ASTERISK-24563 - Direct Media calls within private network sometimes get one way audio (Reported by Kevin Harwell) * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core (Reported by Matt Jordan) * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag enabled (Reported by Richard Mudgett) * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is enabled (Reported by Andreas Steinmetz) * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int (Reported by Walter Doekes) * ASTERISK-24536 - AMI redirect with PJSIP fails to move extra channel (Reported by Niklas Larsson) * ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible (Reported by Yaniv Simhi) * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped' (Reported by Rusty Newton) * ASTERISK-24513 - Local channel apparently leaked in off-nominal DTMF attended transfer (Reported by Mark Michelson) * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present on startup (Reported by Richard Kenner) * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment) (Reported by Karsten Wemheuer) * ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall calls to the transferrer. (Reported by Richard Mudgett) * ASTERISK-24376 - res_pjsip_refer: REFER request for remote session attempts to direct channel to external_replaces extension instead of context, without providing for the Referred-To SIP URI (Reported by Matt Jordan) * ASTERISK-24591 - Stasis() side of an ARI originated channel cannot be Redirected (Reported by Kinsey Moore) * ASTERISK-24049 - Asterisk Manager Interface: A number of list type responses aren't using astman_send_listack (Reported by Jonathan Rose) * ASTERISK-24637 - Channel re-enters Stasis() when it should not (Reported by John Bigelow) * ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does not function (Reported by John Kiniston) * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT (Reported by Kristian Høgh) * ASTERISK-20744 - [patch] Security event logging does not work over syslog (Reported by Michael Keuter) * ASTERISK-24665 - Configure check required for pjsip_get_dest_info() (Reported by Mark Michelson) * ASTERISK-23850 - Park Application does not respect Return Context Priority (Reported by Andrew Nagy) * ASTERISK-23991 - [patch]asterisk.pc file contains a small error in the CFlags returned (Reported by Diederik de Groot) * ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown while attempting to publish (Reported by Kevin Harwell) * ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown (Reported by Corey Farrell) * ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails on cross compilation (Reported by abelbeck) * ASTERISK-24624 - Transfer to invalid extension results in hung channel. (Reported by Zane Conkle) * ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf, Incorrect External Addresses is Used in SIP Packets When Responding to INVITE (Reported by David Justl) * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup (Reported by LEI FU) * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects 32-bit packages on 64-bit hosts (Reported by Ben Klang) * ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding to most traffic, potential deadlock (Reported by Jeff Collell) * ASTERISK-24560 - Creating a named ARI bridge twice causes a crash (Reported by Kinsey Moore) * ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported by Matt Jordan) * ASTERISK-24640 - Registration pending stays forever after sip reload (Reported by Max Man) * ASTERISK-24673 - outgoing sip registers cannot be removed or modified without doing restart (or doing module unload chan_sip.so) (Reported by Stefan Engström) * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event (Reported by Gareth Palmer) * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis fails to get app name (Reported by John Bigelow) * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser' (Reported by HZMI8gkCvPpom0tM) * ASTERISK-24693 - Investigate and fix memory leaks in Asterisk (Reported by Kevin Harwell) * ASTERISK-24626 - Voicemail passwords not being stored in ARA (Reported by Paddy Grice) * ASTERISK-24539 - Compile fails on OSX because of sem_timedwait in bridge_channel.c (Reported by George Joseph) * ASTERISK-24544 - Compile fails on OSX Yosemite because of incorrect detection of htonll and ntohll (Reported by George Joseph) * ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX' no longer displays user menus (Reported by Matt Jordan) * ASTERISK-24721 - manager: ModuleLoad action incorrectly reports 'module not found' during a Reload operation (Reported by Matt Jordan) * ASTERISK-24719 - ConfBridge recording channels get stuck when recording started/stopped more than once (Reported by Richard Mudgett) * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported by Kevin Harwell) * ASTERISK-24728 - tcptls: Bad file descriptor error when reloading chan_sip (Reported by Kevin Harwell) * ASTERISK-24729 - Outbound registration not occuring on new registrations after reload. (Reported by Richard Mudgett) * ASTERISK-24676 - Security Vulnerability: URL request injection in libCURL (CVE-2014-8150) (Reported by Matt Jordan) * ASTERISK-24666 - Security Vulnerability: RTP not closed after sip call using unsupported codec (Reported by Y Ateya) * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL versions (Reported by Jared Biel) * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by Stephan Eisvogel) * ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson) * ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response is ever received (Reported by Marco Paland) * ASTERISK-24737 - When agent not logged in, agent status shows unavailable, queue status shows agent invalid (Reported by Richard Mudgett) Improvements made in this release: ----------------------------------- * ASTERISK-24552 - ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes (Reported by Matt Jordan) * ASTERISK-24553 - ARI/AMI: Include language in standard channel snapshot output (Reported by Matt Jordan) * ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by Matt Jordan) * ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for connection-oriented transports. (Reported by Matt Jordan) * ASTERISK-24412 - [patch]Incomplete channel originate/continue handling with ARI (Reported by Nir Simionovich (GreenfieldTech - Israel)) * ASTERISK-24678 - [PATCH] Added atxfer* settings to features.conf.sample (Reported by Niklas Larsson) * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported by cloos) * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by Dan Jenkins) * ASTERISK-24316 - For httpd server, need option to define server name for security purposes (Reported by Andrew Nagy) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0 Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 13.2.1. The release of Asterisk 13.2.1 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: * --- pjsip: resolve compatibility problem with ast_sip_session (Closes issue ASTERISK-24941. 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-13.2.1 Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 13.3.0. The release of Asterisk 13.3.0 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-24703 - ARI: Add the ability to "transfer" (redirect) a channel (Reported by Matt Jordan) * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation (Reported by Dwayne Hubbard) Bugs fixed in this release: ----------------------------------- * ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid string copy (Reported by Yura Kocyuba) * ASTERISK-24748 - res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur (Reported by Joshua Colp) * ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked (Reported by Matt Jordan) * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in res_odbc (Reported by ibercom) * ASTERISK-24479 - Enable REF_DEBUG for module references (Reported by Corey Farrell) * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information (Reported by Matt Jordan) * ASTERISK-24772 - ODBC error in realtime sippeers when device unregisters under MariaDB (Reported by Richard Miller) * ASTERISK-24752 - Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown (Reported by Richard Mudgett) * ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported by Zane Conkle) * ASTERISK-24015 - app_transfer fails with PJSIP channels (Reported by Private Name) * ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk transfer scenario. (Reported by Mark Michelson) * ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by Niklas Larsson) * ASTERISK-24716 - Improve pjsip log messages for presence subscription failure (Reported by Rusty Newton) * ASTERISK-24612 - res_pjsip: No information if a required sorcery wizard is not loaded (Reported by Joshua Colp) * ASTERISK-24768 - res_timing_pthread: file descriptor leak (Reported by Matthias Urlichs) * ASTERISK-24685 - "pjsip show version" CLI command (Reported by Joshua Colp) * ASTERISK-24632 - install_prereq script installs pjproject without IPv6 support (Reported by Rusty Newton) * ASTERISK-24085 - Documentation - We should remove or further document the 'contact' section in pjsip.conf (Reported by Rusty Newton) * ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by JoshE) * ASTERISK-24700 - CRASH: NULL channel is being passed to ast_bridge_transfer_attended() (Reported by Zane Conkle) * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove (Reported by Corey Farrell) * ASTERISK-24799 - [patch] make fails with undefined reference to SSLv3_client_method (Reported by Alexander Traud) * ASTERISK-22670 - Asterisk crashes when processing ISDN AoC Events (Reported by klaus3000) * ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn call (Reported by Marcel Manz) * ASTERISK-24740 - [patch]Segmentation fault on aoc-e event (Reported by Panos Gkikakis) * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime (Reported by Graham Barnett) * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers (Reported by Corey Farrell) * ASTERISK-24796 - Codecs and bucket schema's prevent module unload (Reported by Corey Farrell) * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML (Reported by Ashley Sanders) * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring is invalid (Reported by Rusty Newton) * ASTERISK-24785 - 'Expires' header missing from 200 OK on REGISTER (Reported by Ross Beer) * ASTERISK-24677 - ARI GET variable on channel provides unhelpful response on non-existent variable (Reported by Joshua Colp) * ASTERISK-24797 - bridge_softmix: G.729 codec license held (Reported by Kevin Harwell) * ASTERISK-24812 - ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding (Reported by Matt Jordan) * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill (Reported by JoshE) * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime fail (Reported by Terry Wilson) * ASTERISK-23214 - chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases (Reported by Rusty Newton) * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller (Reported by Makoto Dei) * ASTERISK-18105 - most of asterisk modules are unbuildable in cygwin environment (Reported by feyfre) * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell) * ASTERISK-24751 - Integer values in json payload to ARI cause asterisk to crash (Reported by jeffrey putnam) * ASTERISK-24838 - chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling (Reported by Richard Mudgett) * ASTERISK-24825 - Caller ID not recognized using Centrex/Distinctive dialing (Reported by Richard Mudgett) * ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not HAVE_PJPROJECT (Reported by Stefan Engström) * ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers (Reported by Kevin Harwell) * ASTERISK-24755 - Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge (Reported by John Bigelow) * ASTERISK-24739 - [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules (Reported by Ed Hynan) * ASTERISK-23390 - NewExten Event with application AGI shows up before and after AGI runs (Reported by Benjamin Keith Ford) * ASTERISK-24786 - [patch] - Asterisk terminates when playing a voicemail stored in LDAP (Reported by Graham Barnett) * ASTERISK-24808 - res_config_odbc: Improper escaping of backslashes occurs with MySQL (Reported by Javier Acosta) * ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported by Anatoli) * ASTERISK-20850 - [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality. (Reported by Diederik de Groot) * ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI connection on error (Reported by Dmitriy Serov) * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported by Frank DiGennaro) * ASTERISK-21038 - Bad command completion of "core set debug channel" (Reported by Richard Kenner) * ASTERISK-18708 - func_curl hangs channel under load (Reported by Dave Cabot) * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by Atis Lezdins) * ASTERISK-24876 - Investigate reference leaks from tests/channels/local/local_optimize_away (Reported by Corey Farrell) * ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported by Corey Farrell) * ASTERISK-24817 - init_logger_chain: unreachable code block (Reported by Corey Farrell) * ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by snuffy) * ASTERISK-24879 - [patch]Compilation fails due to 64bit time under OpenBSD (Reported by snuffy) Improvements made in this release: ----------------------------------- * ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes (Reported by Ben Merrills) * ASTERISK-24811 - asterisk-publication sorcery object does not use realtime (Reported by Matt Hoskins) * ASTERISK-24790 - Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context (Reported by Graham Barnett) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0 Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 13.3.1. The release of Asterisk 13.3.1 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: * --- pjsip: resolve compatibility problem with ast_sip_seesion (Closes issue ASTERISK-24941. 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-13.3.1 Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 13.4.0. The release of Asterisk 13.4.0 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-24922 - ARI: Add the ability to intercept hold and raise an event (Reported by Matt Jordan) Bugs fixed in this release: ----------------------------------- * ASTERISK-25112 - Logger: Configuration settings are not reset to default during reload. (Reported by Corey Farrell) * ASTERISK-24944 - main/audiohook.c change prevents G722 call recording (Reported by Ronald Raikes) * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2 or more digits (Reported by Makoto Dei) * ASTERISK-25086 - [patch]PJSIP crashes if endpoint missing in Dial() (Reported by snuffy) * ASTERISK-25089 - res_pjsip_config_wizard: Variable specified in templates aren't being processed correctly (Reported by George Joseph) * ASTERISK-25090 - CLI core show channel truncates cdr variables (Reported by snuffy) * ASTERISK-25085 - [patch]Potential crash after unload of func_periodic_hook or test_message (Reported by Corey Farrell) * ASTERISK-25083 - Message.c: Message channel becomes saturated with frames leading to spammy log messages (Reported by Jonathan Rose) * ASTERISK-25082 - Asterisk deletes message after doing a playback of an INBOX message using ast_vm_play when the Old folder is full for that mailbox. (Reported by Jonathan Rose) * ASTERISK-25041 - [patch]Broken column type checking in res_config_mysql addon (Reported by Alexandre Fournier) * ASTERISK-21893 - Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c (Reported by Alexandr Gordeev) * ASTERISK-25074 - Regression: Recent clang-related change broke cross compiling of Asterisk (Reported by Sebastian Kemper) * ASTERISK-25042 - asterisk.conf options override command-line options. (Reported by Corey Farrell) * ASTERISK-24442 - Outgoing call files don't work properly when set in the future (Reported by tootai) * ASTERISK-25057 - res_pjsip_pubsub: Crash in send_notify due to invalid root pointer in sub_tree (Reported by Matt Jordan) * ASTERISK-24938 - ARI Snoop Channel results in excessive escalating CPU usage (Reported by George Ladoff) * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests. (Reported by Richard Mudgett) * ASTERISK-25003 - Asterisk crashes on attended transfer (using feature) (Reported by Artem Volodin) * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always contain waiting time (Reported by Etienne Lessard) * ASTERISK-25027 - Build System: Many ARI modules are missing dependencies. (Reported by Corey Farrell) * ASTERISK-25061 - pbx_config: Register manager actions with module version of macro. (Reported by Corey Farrell) * ASTERISK-25025 - Periodic crashes (in ast_channel_snapshot_create at stasis_channels.c) with Certified Asterisk 13. (Reported by Chet Stevens) * ASTERISK-25053 - Unit test category /main/presence missing trailing slash. (Reported by Corey Farrell) * ASTERISK-22708 - res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work (Reported by JoshE) * ASTERISK-25054 - Formats interface's cannot be unregistered, needs to hold modules until shutdown. (Reported by Corey Farrell) * ASTERISK-24896 - [patch] Using force black background leads to colours not being reset (Reported by dant) * ASTERISK-25033 - Asterisk 13 (branch head) won't compile without PJSip (Reported by Peter Whisker) * ASTERISK-25028 - Build System: Unneeded defines in asterisk/buildopts.h (Reported by Corey Farrell) * ASTERISK-25048 - Astobj2: Initialization order wrong when both refdebug and AO2_DEBUG are both enabled. (Reported by Corey Farrell) * ASTERISK-19608 - Asterisk-1.8.x starts rejecting calls with cause code 44 after some time. (Reported by Denis Alberto Martinez) * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25037 - res_pjsip_outbound_registration: Potential crash in off-nominal failure case when sending message (Reported by Joshua Colp) * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls (Reported by Steve Davies) * ASTERISK-22790 - check_modem_rate() may return incorrect rate for V.27 (Reported by not here) * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load (Reported by David Brillert) * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400, which is disallowed in res_fax's check_modem_rate (Reported by Matt Jordan) * ASTERISK-24996 - chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR Sections Exist in pjsip.conf (Reported by Ashley Sanders) * ASTERISK-25020 - Mismatched response to outgoing REGISTER request (Reported by Mark Michelson) * ASTERISK-25018 - pjsip show endpoints crashes asterisk when qualified aors present (Reported by Ivan Poddubny) * ASTERISK-24749 - ConfBridge: Wrong language on playing conf-hasjoin and conf-hasleft when played to bridge (Reported by Philippe Bolduc) * ASTERISK-24845 - pjsip send notify not working with Cisco phone (Reported by Carl Fortin) * ASTERISK-25004 - Crash in authenticated reinvite after originated T.38 FAX (Reported by Mark Michelson) * ASTERISK-24999 - PJSIP crashes with malformed contact line (Reported by snuffy) * ASTERISK-24998 - res_corosync: res_corosync tries to load even if res_corosync.conf is missing (Reported by George Joseph) * ASTERISK-24997 - Astobj2: Some callers of __adjust_lock do not pre-check the object (Reported by Corey Farrell) * ASTERISK-24982 - res_pjsip_mwi: Unsolicited MWI NOTIFY only sent on mailbox changes (Reported by Joshua Colp) * ASTERISK-24991 - Check for ao2_alloc failure in __ast_channel_internal_alloc (Reported by Corey Farrell) * ASTERISK-24895 - After hangup on the side of the ISDN network no HangupRequest event comes for the dahdi channel. (Reported by Andrew Zherdin) * ASTERISK-24977 - Contacts that don't use qualify are being marked as unavailable (Reported by George Joseph) * ASTERISK-24774 - Segfault in ast_context_destroy with extensions.ael and extensions.conf (Reported by Corey Farrell) * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have multiple native formats (Reported by Matt Jordan) * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build to Fail (Reported by Ashley Sanders) * ASTERISK-24958 - Forwarding loop detection inhibits certain desirable scenarios (Reported by Mark Michelson) * ASTERISK-24863 - res_pjsip: No endpoint events raised via AMI when contacts cannot be reached/qualified (Reported by Dmitriy Serov) * ASTERISK-24869 - Asterisk segfaults on DAHDI attended transfer due to application (appl) being NULL on unbridged channel (Reported by viniciusfontes) * ASTERISK-24970 - Crash in res_pjsip_pubsub handling of failed notify (Reported by Scott Griepentrog) * ASTERISK-24959 - [patch]CLI command cdr show pgsql status (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-24954 - Git migration: Asterisk version numbers are incompatible with the Test Suite (Reported by Matt Jordan) * ASTERISK-17608 - func_aes.so cannot be loaded if res_crypto / openssl not compiled (Reported by Warren Selby) * ASTERISK-24928 - [patch]t38_udptl_maxdatagram in pjsip.conf not honored (Reported by Juergen Spies) * ASTERISK-24835 - Early Media Not working with Chan SIP and Asterisk 13 (Reported by Andrew Nagy) * ASTERISK-21777 - Asterisk tries to transcode video instead of audio (Reported by Nick Ruggles) * ASTERISK-24380 - core: Native formats are set to h264 with certain audio/video codec configuration, resulting in path translation WARNINGs (Reported by Matt Jordan) * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken into account (Reported by Frederic Van Espen) * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too short (Reported by Y Ateya) * ASTERISK-24935 - res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator. (Reported by Corey Farrell) * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c (Reported by Vadim) * ASTERISK-24933 - T38 fails negotiation (Reported by Jonathan Rose) * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL byte prefix bug (Reported by Matt Jordan) * ASTERISK-21211 - chan_iax2 - unprotected access of iaxs[peer->callno] potentially results in segfault (Reported by Jaco Kroon) * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working (Reported by Christoph Timm) * ASTERISK-24782 - StasisEnd event not present for channel that was swapped out for another after completing attended transfer (Reported by John Bigelow) * ASTERISK-24910 - "timer=no" and "timer=required" settings in pjsip.conf fail (Reported by Ray Crumrine) * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0 (Reported by Jeffrey C. Ollie) * ASTERISK-24914 - Division by zero in file.c when playback of voicemail with video as h264 (Reported by Marcello Ceschia) * ASTERISK-24899 - Parking fall-through behavior different in 13 (Reported by Malcolm Davenport) * ASTERISK-24937 - [patch]res_pjsip_messaging: Messages may be sent out of order (Reported by Mark Michelson) * ASTERISK-24920 - Asterisk handles duplicate SIP requests as if they were each a new request (Reported by Mark Michelson) * ASTERISK-24857 - [patch] "timing test", pjsip incoming/outgoing calls, voicemail prompts and recordings all fail when using the kqueue timer source on FreeBSD 10.x (Reported by Justin T. Gibbs) * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion detection in ast_malloc (Reported by Timo Teräs) * ASTERISK-24142 - CCSS: crash during shutdown due to device lookup in destroyed container (Reported by David Brillert) * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during core restart now (Reported by Peter Katzmann) * ASTERISK-24805 - [patch] - ASAN: Race condition (heap-use-after-free) on asterisk closing (Reported by Badalian Vyacheslav) * ASTERISK-24881 - ast_register_atexit should only be used when absolutely needed (Reported by Corey Farrell) * ASTERISK-24731 - res_pjsip_session cannot be unloaded (Reported by Corey Farrell) * ASTERISK-24864 - app_confbridge: file playback blocks dtmf (Reported by Kevin Harwell) * ASTERISK-14233 - [patch] Buddies are always auto-registered when processing the roster (Reported by Simon Arlott) * ASTERISK-24780 - [patch] - Buddies are always auto-registered when processing the roster (Reported by Simon Arlott) * ASTERISK-24781 - PJSIP: Unnecessary 180 Ringing messages sent with undesireabe consequences. (Reported by Richard Mudgett) Improvements made in this release: ----------------------------------- * ASTERISK-25044 - sorcery: Add ability to insert a new wizard into an object type's list (Reported by George Joseph) * ASTERISK-24892 - Super Awesome Company sound prompts (Reported by Rusty Newton) * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove Hjelm) * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL (Reported by Alexander Traud) * ASTERISK-25045 - vector: Add new capabilities and unit tests (Reported by George Joseph) * ASTERISK-24706 - [patch]add auto-dtmf mode for pjsip (Reported by yaron nahum) * ASTERISK-25051 - Remove unneeded uses of optional_api providers. (Reported by Corey Farrell) * ASTERISK-25040 - pbx: Improve performance of reloads by making hint destruction more performant (Reported by Matt Jordan) * ASTERISK-24917 - [patch] clang compilation warnings (Reported by Diederik de Groot) * ASTERISK-24949 - res_pjsip_outbound_registration: Backport line functionality (Reported by Joshua Colp) * ASTERISK-24965 - cel_pgsql - log_error string references CDR instead of CEL (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-24918 - pjsip: add CLI options to display global and system configuration (Reported by Scott Griepentrog) * ASTERISK-24862 - [patch] Support in-dialog OPTIONS (Reported by yaron nahum) * ASTERISK-24802 - stasis: set a channel variable on websocket disconnect error (Reported by Kevin Harwell) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.4.0 Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 13.5.0. The release of Asterisk 13.5.0 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-25256 - [patch]Post AMI VarSet to empty string events when Asterisk deletes a dialplan variable. (Reported by Richard Mudgett) * ASTERISK-25067 - Sorcery Caching: Implement a new caching module (Reported by Matt Jordan) * ASTERISK-25040 - pbx: Improve performance of reloads by making hint destruction more performant (Reported by Matt Jordan) * ASTERISK-25114 - res_pjsip: Add AMI etents for chan_pjsip contact lifecycle changes (Reported by George Joseph) * ASTERISK-25072 - res_pjsip_outbound_registration: line functionality. Additional check for using the request URI (Reported by Dmitriy Serov) Bugs fixed in this release: ----------------------------------- * ASTERISK-25250 - chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback (Reported by Etienne Lessard) * ASTERISK-25253 - confbridge volume options and other volume controls such as func_volume don't work (Reported by Dmitriy Serov) * ASTERISK-25247 - choppy audio when spying on a g722 channel, chan_sip or chan_pjsip (Reported by hristo) * ASTERISK-24867 - Docs for 'e' option in ResetCDR say to use CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty Newton) * ASTERISK-24853 - Documentation claims chan_sip outbound registrations support WS or WSS as valid transports (not true) (Reported by PSDK) * ASTERISK-25242 - PJSIP: No audio when Asterisk inside NAT and endpoints outside NAT - implement functionality similar to chan_sip 'rtpkeepalive'? (Reported by Mark Michelson) * ASTERISK-25258 - chan_pjsip: Incorrect format switch on received RTP packet (Reported by Joshua Colp) * ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope (Reported by Patric Marschall) * ASTERISK-24934 - [patch]Asterisk manager output does not escape control characters (Reported by warren smith) * ASTERISK-25255 - Missing AMI VarSet events when setting to an empty string. (Reported by Richard Mudgett) * ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to an empty string before Park. (Reported by Richard Mudgett) * ASTERISK-25183 - PJSIP: Crash on NULL channel in chan_pjsip_incoming_response despite previous checks for NULL channel (Reported by Matt Jordan) * ASTERISK-25201 - Crash in PJSIP distributor on already free'd threadpool (Reported by Matt Jordan) * ASTERISK-24782 - StasisEnd event not present for channel that was swapped out for another after completing attended transfer (Reported by John Bigelow) * ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully started when completing attended transfer (Reported by Joshua Colp) * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes (Reported by Rusty Newton) * ASTERISK-22805 - res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP (Reported by Dmitry Burilov) * ASTERISK-24550 - res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake (Reported by Osaulenko Alexander) * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by Badalian Vyacheslav) * ASTERISK-24832 - [patch]DTLS-crashes within openssl (Reported by Stefan Engström) * ASTERISK-25127 - DTLS crashes following "Unable to cancel schedule ID" in dtls_srtp_check_pending (Reported by Dade Brandon) * ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c (Reported by Carl Fortin) * ASTERISK-25115 - Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c (Reported by John Bigelow) * ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early replaces call pickup (Reported by Walter Doekes) * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c (Reported by Walter Doekes) * ASTERISK-25219 - [patch]Source and destination overlap in memcpy in rtp_engine.c (Reported by Walter Doekes) * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS (Reported by Walter Doekes) * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed: Bad file descriptor" (Reported by Barry Chern) * ASTERISK-25165 - Testsuite - Sorcery memory cache leaks (Reported by Corey Farrell) * ASTERISK-25202 - Hints extension state broken between 13.3.2 and 13.4 (Reported by cervajs) * ASTERISK-25196 - res_pjsip_nat: rewrite_contact should not be applied to Contact header when Record-Route headers are present (Reported by Mark Michelson) * ASTERISK-24907 - res_pjsip_outbound_registration: crash during unload if registration attempts are still occuring (Reported by Kevin Harwell) * ASTERISK-25204 - res_pjsip_refer: Duplicated Referred-By or Replaces headers on outbound INVITEs. (Reported by Mark Michelson) * ASTERISK-25171 - Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound. (Reported by Rusty Newton) * ASTERISK-25189 - AMI: Add Linkedid header to standard channel snapshot information. (Reported by Richard Mudgett) * ASTERISK-25172 - Crash in channels/sip/sip blind transfer/caller_refer_only test in ast_format_cap_append_from_cap during ast_request (Reported by Matt Jordan) * ASTERISK-25180 - res_pjsip_mwi: Unsolicited MWI requires reload (Reported by Joshua Colp) * ASTERISK-25182 - [patch] on CLI sip reload, new codecs get appended only (Reported by Alexander Traud) * ASTERISK-25163 - Deadlock in chan_sip between reload of sip peer container and MWI Stasis callback (Reported by Dmitriy Serov) * ASTERISK-25091 - Asterisk REST API - bridge.addChannel crash asterisk when calling channel hangup while adding to bridge (Reported by Ilya Trikoz) * ASTERISK-24900 - Manager event ParkedCallSwap is not documented (Reported by Rusty Newton) * ASTERISK-25162 - func_pjsip_aor: Leak of contact in iterator (Reported by Corey Farrell) * ASTERISK-25158 - res_pjsip: Add option to use AAL2 packing when negotiating g.726 (Reported by Kevin Harwell) * ASTERISK-24344 - CDR_PROP(disable) disables CDR only for first dialed party (Reported by Janusz Karolak) * ASTERISK-24443 - CDR fields (dst, dcontext) empty in transfer call started from Macro (Reported by Arveno Santoro) * ASTERISK-25154 - [patch]fromtag may need to be updatep after successful call dialog match (Reported by Damian Ivereigh) * ASTERISK-25156 - chan_pjsip’s CHAN_START cel event lacks the correct context and exten (Reported by cloos) * ASTERISK-25157 - bridging: Performing a blonde transfer does not result in connected line updates (Reported by Joshua Colp) * ASTERISK-25087 - Asterisk segfault when using Directory application with alias option and specific mailbox configuration (Reported by Chet Stevens) * ASTERISK-24983 - IAX deadlock between hangup and scheduled actions (ex. largrq) (Reported by Y Ateya) * ASTERISK-25096 - [patch]Segfault when registering over websockets with PJSIP (in ast_sockaddr_isnull at /include/asterisk/netsock2.h) (Reported by Josh Kitchens) * ASTERISK-24963 - ASAN: heap-use-after-free with PJSIP and WSS (Reported by Badalian Vyacheslav) * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute but asterisk doesn't detect it. (Reported by ibercom) * ASTERISK-25094 - PBX core: Investigate thread safety issues (Reported by Corey Farrell) * ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark Michelson) * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10} (Reported by Badalian Vyacheslav) * ASTERISK-25137 - endpoint stasis messages are delivered twice (Reported by Vitezslav Novy) * ASTERISK-25116 - res_pjsip: Two PeerStatus AMI messages are sent for every status change (Reported by George Joseph) * ASTERISK-25131 - chan_pjsip: In-dialog authentication not handled. (Reported by Richard Mudgett) * ASTERISK-25100 - asterisk coredump if host has an IPv6 address that end with ::80 (Reported by Mark Petersen) * ASTERISK-25122 - Large SIP packet received via pjsip over websocket crashes Asterisk (Reported by Ivan Poddubny) * ASTERISK-25121 - Stasis: Fix unsafe use of stasis_unsubscribe in modules. (Reported by Corey Farrell) * ASTERISK-24988 - func_talkdetect: Test is bouncing sporadically (Reported by Joshua Colp) * ASTERISK-25105 - res_pjsip: Possible incompatibility between qualify_timeout and pjproject-2.4 (Reported by George Joseph) * ASTERISK-25117 - res_mwi_external_ami: Fix manager action registrations. (Reported by Corey Farrell) New Features made in this release: ----------------------------------- * ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by Joshua Colp) * ASTERISK-25238 - ARI: Support push configuration (Reported by Matt Jordan) * ASTERISK-25173 - ARI: Add the ability to load/reload/unload an Asterisk module (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-13.5.0 Thank you for your continued support of Asterisk! ----- The Asterisk Development Team has announced the release of Asterisk 13.6.0. The release of Asterisk 13.6.0 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-25377 - res_pjsip: Change default "From user" from UUID to something more palatable (Reported by Mark Michelson) * ASTERISK-25252 - ARI: Add the ability to manipulate log channels (Reported by Matt Jordan) Bugs fixed in this release: ----------------------------------- * ASTERISK-25449 - main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny (Reported by Matt Jordan) * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when ICE is not enabled (Reported by Joshua Colp) * ASTERISK-25383 - Core dumps on startup and shutdown with MALLOC_DEBUG enabled (Reported by yaron nahum) * ASTERISK-25423 - Caller gets no Connected line update during call pickup. (Reported by Richard Mudgett) * ASTERISK-25305 - Dynamic logger channels can be added multiple times (Reported by Mark Michelson) * ASTERISK-25418 - On-hold channels redirected out of a bridge appear to still be on hold (Reported by Mark Michelson) * ASTERISK-25384 - Regular Asterisk crashes when using Page application. "user_data is NULL" (Reported by Chet Stevens) * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations (Reported by Elazar Broad) * ASTERISK-25410 - app_record: RECORDED_FILE variable not being populated (Reported by Kevin Harwell) * ASTERISK-25394 - pbx: Incorrect device and presence state when changing hint details (Reported by Joshua Colp) * ASTERISK-25396 - chan_sip: Extremely long callerid name causes invalid SIP (Reported by Walter Doekes) * ASTERISK-25399 - app_queue: AgentComplete event has wrong reason (Reported by Kevin Harwell) * ASTERISK-25185 - Segfault in app_queue on transfer scenarios (Reported by Etienne Lessard) * ASTERISK-25353 - [patch] Transcoding while different in Frame size = Frames lost (Reported by Alexander Traud) * ASTERISK-25325 - ARI PUT reload chan_sip HTTP response 404 (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25390 - default_from_user can crash with certain configuration backends (Reported by Mark Michelson) * ASTERISK-25387 - res_pjsip_nat: Malformed REGISTER request causes NAT'd Contact header to not be rewritten (Reported by Matt Jordan) * ASTERISK-25227 - No audio at in-band announcements in ooh323 channel (Reported by Alexandr Dranchuk) * ASTERISK-25369 - res_parking: ParkAndAnnounce - Inheritable variables aren't applied to the announcer channel (Reported by Jonathan Rose) * ASTERISK-25295 - res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov) * ASTERISK-25381 - res_pjsip: AoRs deleted via ARI (or other mechanism) do not destroy their related contacts (Reported by Matt Jordan) * ASTERISK-25367 - pbx: Long pattern match hints may cause "core show hints" to crash (Reported by Joshua Colp) * ASTERISK-25365 - Persistent subscriptions have extra Content-Length/corrupted messages (Reported by Mark Michelson) * ASTERISK-25362 - Deadlock due to presence state callback (Reported by Mark Michelson) * ASTERISK-25356 - res_pjsip_sdp_rtp: Multiple keepalive scheduled items may exist (Reported by Joshua Colp) * ASTERISK-25355 - sched: ast_sched_del may return prematurely due to spurious wakeup (Reported by Joshua Colp) * ASTERISK-25318 - tests/rest_api/applications/subscribe-endpoint/nominal/resource: Sporadically failing (Reported by Joshua Colp) * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup cause on call pickup (Reported by Joshua Colp) * ASTERISK-25342 - res_pjsip: Repeated usage of pj_gethostip may block (Reported by Joshua Colp) * ASTERISK-25341 - bridge: Hangups may get lost when executing actions (Reported by Joshua Colp) * ASTERISK-25339 - res_pjsip: Empty "auth" sections from non-config backgrounds are interpreted as valid (Reported by Matt Jordan) * ASTERISK-25215 - Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne Gaetz) * ASTERISK-25322 - Crash occurs when using MixMonitor with t() or r() options. (Reported by Richard Mudgett) * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite (Reported by Kevin Harwell) * ASTERISK-25315 - DAHDI channels send shortened duration DTMF tones. (Reported by Richard Mudgett) * ASTERISK-25312 - res_http_websocket: Terminate connection on fatal cases (Reported by Joshua Colp) * ASTERISK-25306 - Persistent subscriptions can save multiple SIP messages at once, leading to potential crashes. (Reported by Mark Michelson) * ASTERISK-25309 - [patch] iLBC 20 advertised (Reported by Alexander Traud) * ASTERISK-25304 - res_pjsip: XML sanitization may write past buffer (Reported by Joshua Colp) * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1 (Reported by Stefan Engström) * ASTERISK-25296 - RTP performance issue with several channel drivers. (Reported by Richard Mudgett) * ASTERISK-25297 - Crashes running channels/pjsip/resolver/srv/failover/in_dialog testsuite tests (Reported by Richard Mudgett) * ASTERISK-25292 - Testuite: tests/apps/bridge/bridge_wait/bridge_wait_e_options fails (Reported by Kevin Harwell) * ASTERISK-25271 - Parking & blind transfer: Transferer channel not hung up if no MOH (Reported by Kevin Harwell) Improvements made in this release: ----------------------------------- * ASTERISK-24870 - ARI: Subscriptions to bridges generally not super useful (Reported by Matt Jordan) * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.6.0 Thank you for your continued support of Asterisk!
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# $NetBSD: Makefile,v 1.1.1.1 2015/12/05 23:29:05 jnemeth Exp $
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# $NetBSD: Makefile,v 1.2 2015/12/05 23:42:44 jnemeth Exp $
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#
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# NOTE: when updating this package, there are two places that sound
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# tarballs need to be checked
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@comment $NetBSD: PLIST,v 1.1.1.1 2015/12/05 23:29:08 jnemeth Exp $
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@comment $NetBSD: PLIST,v 1.2 2015/12/05 23:42:44 jnemeth Exp $
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include/asterisk.h
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include/asterisk/_private.h
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include/asterisk/abstract_jb.h
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$NetBSD: distinfo,v 1.1.1.1 2015/12/05 23:29:08 jnemeth Exp $
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$NetBSD: distinfo,v 1.2 2015/12/05 23:42:44 jnemeth Exp $
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SHA1 (asterisk-13.6.0/asterisk-13.6.0.tar.gz) = 320d3b83f6d4641db76c957738b21a7b8eec938b
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RMD160 (asterisk-13.6.0/asterisk-13.6.0.tar.gz) = c6cdb801c65f3539d16c6db1e10bd75e6807684e
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# $NetBSD: options.mk,v 1.1.1.1 2015/12/05 23:29:09 jnemeth Exp $
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# $NetBSD: options.mk,v 1.2 2015/12/05 23:42:44 jnemeth Exp $
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PKG_OPTIONS_VAR= PKG_OPTIONS.asterisk
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PKG_SUPPORTED_OPTIONS= zaptel x11 unixodbc ilbc webvmail ldap spandsp
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