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Fedora Update Notification
FEDORA-2015-5948
2015-04-10 17:05:34
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Name : asterisk
Product : Fedora 22
Version : 13.3.2
Release : 1.fc22
URL : http://www.asterisk.org/
Summary : The Open Source PBX
Description :
Asterisk is a complete PBX in software. It runs on Linux and provides
all of the features you would expect from a PBX and more. Asterisk
does voice over IP in three protocols, and can interoperate with
almost all standards-based telephony equipment using relatively
inexpensive hardware.
——————————————————————————–
Update Information:
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
——————————————————————————–
ChangeLog:
* Thu Apr 9 2015 Jeffrey C. Ollie <jeff@ocjtech.us> – 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 <jeff@ocjtech.us> – 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)
–
– For a full list of changes in this release, please see the ChangeLog:
–
– http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.1
* Wed Apr 1 2015 Jeffrey C. Ollie <jeff@ocjtech.us> – 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)
–
– For a full list of changes in this release, please see the ChangeLog:
–
– http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0
* Wed Apr 1 2015 Jeffrey C. Ollie <jeff@ocjtech.us> – 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)
–
– For a full list of changes in this release, please see the ChangeLog:
–
– http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0
——————————————————————————–
References:
[ 1 ] Bug #1210225 – CVE-2015-3008 asterisk: TLS Certificate Common name NULL byte exploit
https://bugzilla.redhat.com/show_bug.cgi?id=1210225
——————————————————————————–
This update can be installed with the “yum” update program. Use
su -c ‘yum update asterisk’ at the command line.
For more information, refer to “Managing Software with yum”,
available at http://docs.fedoraproject.org/yum/.
All packages are signed with the Fedora Project GPG key. More details on the
GPG keys used by the Fedora Project can be found at
https://fedoraproject.org/keys
——————————————————————————–
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