Fedora 22 Asterisk Update: Critical TLS Common Name Exploit Fix

* Thu Apr 9 2015 Jeffrey C. Ollie - 13.3.2-1: - The Asterisk Development Team has announced security releases for Certified - Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available - security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11, - 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2. - - These releases are available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ - - The release of these versions resolves the following security vulnerability: - - * AST-2015-003: TLS Certificate Common name NULL byte exploit - - When Asterisk registers to a SIP TLS device and and verifies the server, - Asterisk will accept signed certificates that match a common name other than - the one Asterisk is expecting if the signed certificate has a common name - containing a null byte after the portion of the common name that Asterisk - expected. This potentially allows for a man in the middle attack. - - For more information about the details of this vulnerability, please read - security advisory AST-2015-003, which was released at the same time as this - announcement. - - For a full list of changes in the current releases, please see the ChangeLogs: - - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5 - http://downloads.asterisk.org/pub/telephony/asterisk/releases//ChangeLog-1.8.32.3 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11 - http://downloads.asterisk.org/pub/telephony/asterisk/releases//ChangeLog-11.17.1 - http://downloads.asterisk.org/pub/telephony/asterisk/releases//ChangeLog-12.8.2 - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2 - http://downloads.asterisk.org/pub/telephony/asterisk/releases//ChangeLog-13.3.2 - - The security advisory is available at: - - * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf * Thu Apr 9 2015 Jeffrey C. Ollie - 13.3.1-1: - The Asterisk Development Team has announced the release of Asterisk 13.3.1. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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_session - (Closes issue ASTERISK-24941. Reported by Matt Jordan) - - - * Wed Apr 1 2015 Jeffrey C. Ollie - 13.3.0-1: - The Asterisk Development Team has announced the release of Asterisk 13.3.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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) - - - * Wed Apr 1 2015 Jeffrey C. Ollie - 13.2.0-1: - The Asterisk Development Team has announced the release of Asterisk 13.2.0. - This release is available for immediate download at - http://downloads.asterisk.org/pub/telephony/asterisk/ - - 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) - -