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asterisk-voicemail-odbc-13.9.1-1.fc25.1 RPM for x86_64

From Fedora 25 for x86_64 / a

Name: asterisk-voicemail-odbc Distribution: Fedora Project
Version: 13.9.1 Vendor: Fedora Project
Release: 1.fc25.1 Build date: Tue May 17 04:13:47 2016
Group: Applications/Internet Build host: buildvm-27-nfs.phx2.fedoraproject.org
Size: 264400 Source RPM: asterisk-13.9.1-1.fc25.1.src.rpm
Packager: Fedora Project
Url: http://www.asterisk.org/
Summary: Store voicemail in a database using ODBC
Voicemail implementation for Asterisk that uses ODBC to store
voicemail in a database.

Provides

Requires

License

GPLv2

Changelog

* Tue May 17 2016 Jitka Plesnikova <jplesnik@redhat.com> - 13.9.1-1.1
  - Perl 5.24 rebuild
* Sat May 14 2016 Jared Smith <jsmith@fedoraproject.org> - 13.9.1-1
  - Update to upstream 13.9.1 release
  - Use bootstrap.sh instead of calling autoconf tools manually
  - Fix up shifting pjproject submodules
  - Fix up requires on speexdsp-devel for EPEL7 (RHBZ#1310444)
* Tue Feb 16 2016 Jared Smith <jsmith@fedoraproject.org> - 13.7.2-2.1
  - Fix alembic requirement
* Tue Feb 09 2016 Michal Toman <mtoman@fedoraproject.org> - 13.7.2-2
  - Do not use -m32/-m64 on MIPS
* Sun Feb 07 2016 Jared Smith <jsmith@fedoraproject.org> - 13.7.2-1
  - Update to upstream release 13.7.2 to fix ASTERISK-25702
* Fri Feb 05 2016 Jared Smith <jsmith@fedoraproject.org> - 13.7.1-2
  - Create sub-package for alembic scripts
* Thu Feb 04 2016 Jared Smith <jsmith@fedoraproject.org> - 13.7.1-1
  - Update to upstream 13.7.1 release for security fixes
  - Resolves AST-2016-001: BEAST vulnerability in HTTP server
  - Resolves AST-2016-002: File descriptor exhaustion in chan_sip
  - Resolves AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data
  - Full changelog at http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
  - Also build the 'radius' sub-package against freeradius-client-devel, as the
   radiusclient-ng project is dead
* Wed Feb 03 2016 Fedora Release Engineering <releng@fedoraproject.org> - 13.3.2-3.1
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild
* Mon Jan 25 2016 Jared Smith <jsmith@fedoraproject.org> - 13.3.2-3
  - Remove %defattr macro invocations, as they are no longer needed
* Sat Jan 23 2016 Robert Scheck <robert@fedoraproject.org> - 13.3.2-2
  - Rebuild for libical 2.0.0
* Wed Jun 17 2015 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 13.3.2-1.2
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild
* Sat Jun 06 2015 Jitka Plesnikova <jplesnik@redhat.com> - 13.3.2-1.1
  - Perl 5.22 rebuild
* Thu Apr 09 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 09 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 01 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 01 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
* Fri Jan 30 2015 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.1.1-1:
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
  - security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
  - 11.15.1, 12.8.1, and 13.1.1.
  -
  - 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 vulnerabilities:
  -
  - * AST-2015-001: File descriptor leak when incompatible codecs are offered
  -
  -                 Asterisk may be configured to only allow specific audio or
  -                 video codecs to be used when communicating with a
  -                 particular endpoint. When an endpoint sends an SDP offer
  -                 that only lists codecs not allowed by Asterisk, the offer
  -                 is rejected. However, in this case, RTP ports that are
  -                 allocated in the process are not reclaimed.
  -
  -                 This issue only affects the PJSIP channel driver in
  -                 Asterisk. Users of the chan_sip channel driver are not
  -                 affected.
  -
  - * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
  -
  -                 CVE-2014-8150 reported an HTTP request injection
  -                 vulnerability in libcURL. Asterisk uses libcURL in its
  -                 func_curl.so module (the CURL() dialplan function), as well
  -                 as its res_config_curl.so (cURL realtime backend) modules.
  -
  -                 Since Asterisk may be configured to allow for user-supplied
  -                 URLs to be passed to libcURL, it is possible that an
  -                 attacker could use Asterisk as an attack vector to inject
  -                 unauthorized HTTP requests if the version of libcURL
  -                 installed on the Asterisk server is affected by
  -                 CVE-2014-8150.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisory AST-2015-001 and AST-2015-002, which were 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-cert4
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
* Fri Jan 30 2015 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.1.0-1
  - The Asterisk Development Team has announced the release of Asterisk 13.1.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - 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_negotiate from /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 Nuno Borges)
  -  * 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
* Thu Jan 29 2015 Peter Robinson <pbrobinson@fedoraproject.org> 13.0.2-3
  - Add speexdsp as build dep as speex_echo.h has moved - rhbz 1181021
* Thu Jan 15 2015 Tom Callaway <spot@fedoraproject.org> - 13.0.2-2
  - update for lua 5.3
* Wed Dec 10 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.0.2-1
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
  - released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
  -
  - 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-2014-019: Remote Crash Vulnerability in WebSocket Server
  -
  -   When handling a WebSocket frame the res_http_websocket module dynamically
  -   changes the size of the memory used to allow the provided payload to fit. If a
  -   payload length of zero was received the code would incorrectly attempt to
  -   resize to zero. This operation would succeed and end up freeing the memory but
  -   be treated as a failure. When the session was subsequently torn down this
  -   memory would get freed yet again causing a crash.
  -
  - For more information about the details of this vulnerability, please read
  - security advisory AST-2014-019, 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-11.6-cert9
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
  -
  - The security advisory is available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
* Thu Nov 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.0.1-1
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
  - security releases are released as versions 1.8.28-cert3, 11.6-cert8, 1.8.32.1,
  - 11.14.1, 12.7.1, and 13.0.1.
  -
  - 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 vulnerabilities:
  -
  - * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
  -   address families
  -
  -   Many modules in Asterisk that service incoming IP traffic have ACL options
  -   ("permit" and "deny") that can be used to whitelist or blacklist address
  -   ranges. A bug has been discovered where the address family of incoming
  -   packets is only compared to the IP address family of the first entry in the
  -   list of access control rules. If the source IP address for an incoming
  -   packet is not of the same address as the first ACL entry, that packet
  -   bypasses all ACL rules.
  -
  - * AST-2014-018: Permission Escalation through DB dialplan function
  -
  -   The DB dialplan function when executed from an external protocol, such as AMI,
  -   could result in a privilege escalation. Users with a lower class authorization
  -   in AMI can access the internal Asterisk database without the required SYSTEM
  -   class authorization.
  -
  - In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
  - security vulnerability:
  -
  - * AST-2014-014: High call load with ConfBridge can result in resource exhaustion
  -
  -   The ConfBridge application uses an internal bridging API to implement
  -   conference bridges. This internal API uses a state model for channels within
  -   the conference bridge and transitions between states as different things
  -   occur. Unload load it is possible for some state transitions to be delayed
  -   causing the channel to transition from being hung up to waiting for media. As
  -   the channel has been hung up remotely no further media will arrive and the
  -   channel will stay within ConfBridge indefinitely.
  -
  - In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
  - the following security vulnerability:
  -
  - * AST-2014-017: Permission Escalation via ConfBridge dialplan function and
  -                 AMI ConfbridgeStartRecord Action
  -
  -   The CONFBRIDGE dialplan function when executed from an external protocol (such
  -   as AMI) can result in a privilege escalation as certain options within that
  -   function can affect the underlying system. Additionally, the AMI
  -   ConfbridgeStartRecord action has options that would allow modification of the
  -   underlying system, and does not require SYSTEM class authorization in AMI.
  -
  - Finally, the release of 12.7.1 and 13.0.1 resolves the following security
  - vulnerabilities:
  -
  - * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack
  -
  -   The Asterisk module res_pjsip provides the ability to configure ACLs that may
  -   be used to reject SIP requests from various hosts. However, the module
  -   currently fails to create and apply the ACLs defined in its configuration
  -   file on initial module load.
  -
  - * AST-2014-015: Remote crash vulnerability in PJSIP channel driver
  -
  -   The chan_pjsip channel driver uses a queue approach for relating to SIP
  -   sessions. There exists a race condition where actions may be queued to answer
  -   a session or send ringing after a SIP session has been terminated using a
  -   CANCEL request. The code will incorrectly assume that the SIP session is still
  -   active and attempt to send the SIP response. The PJSIP library does not
  -   expect the SIP session to be in the disconnected state when sending the
  -   response and asserts.
  -
  - * AST-2014-016: Remote crash vulnerability in PJSIP channel driver
  -
  -   When handling an INVITE with Replaces message the res_pjsip_refer module
  -   incorrectly assumes that it will be operating on a channel that has just been
  -   created. If the INVITE with Replaces message is sent in-dialog after a session
  -   has been established this assumption will be incorrect. The res_pjsip_refer
  -   module will then hang up a channel that is actually owned by another thread.
  -   When this other thread attempts to use the just hung up channel it will end up
  -   using a freed channel which will likely result in a crash.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
  - AST-2014-016, AST-2014-017, and AST-2014-018, which were 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-cert3
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf
* Thu Nov 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 13.0.0-1
  - The Asterisk Development Team is pleased to announce the release of
  - Asterisk 13.0.0. This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - 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
* Fri Nov 14 2014 Tom Callaway <spot@fedoraproject.org> - 11.13.1-2
  - rebuild for new libsrtp
* Mon Oct 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 11.13.1-1
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
  - security releases are released as versions 1.8.28-cert2, 11.6-cert7, 1.8.31.1,
  - 11.13.1, 12.6.1, and 13.0.0-beta3.
  -
  - 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-2014-011: Asterisk Susceptibility to POODLE Vulnerability
  -
  -   Asterisk is susceptible to the POODLE vulnerability in two ways:
  -   1) The res_jabber and res_xmpp module both use SSLv3 exclusively for their
  -      encrypted connections.
  -   2) The core TLS handling in Asterisk, which is used by the chan_sip channel
  -      driver, Asterisk Manager Interface (AMI), and Asterisk HTTP Server, by
  -      default allow a TLS connection to fallback to SSLv3. This allows for a
  -      MITM to potentially force a connection to fallback to SSLv3, exposing it
  -      to the POODLE vulnerability.
  -
  -   These issues have been resolved in the versions released in conjunction with
  -   this security advisory.
  -
  - For more information about the details of this vulnerability, please read
  - security advisory AST-2014-011, 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-cert2
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert7
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.31.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.13.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.6.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0-beta3
  -
  - The security advisory is available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2014-011.pdf
* Mon Oct 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 11.13.0-1
  - The Asterisk Development Team has announced the release of Asterisk 11.13.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.13.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following are the issues resolved in this release:
  -
  - Bugs fixed in this release:
  - -----------------------------------
  -  * ASTERISK-24032 - Gentoo compilation emits warning:
  -       "_FORTIFY_SOURCE" redefined (Reported by Kilburn)
  -  * ASTERISK-24225 - Dial option z is broken (Reported by
  -       dimitripietro)
  -  * ASTERISK-24178 - [patch]fromdomainport used even if not set
  -       (Reported by Elazar Broad)
  -  * ASTERISK-22252 - res_musiconhold cleanup - REF_DEBUG reload
  -       warnings and ref leaks (Reported by Walter Doekes)
  -  * ASTERISK-23997 - chan_sip: port incorrectly incremented for RTCP
  -       ICE candidates in SDP answer (Reported by Badalian Vyacheslav)
  -  * ASTERISK-24019 - When a Music On Hold stream starts it restarts
  -       at beginning of file. (Reported by Jason Richards)
  -  * ASTERISK-23767 - [patch] Dynamic IAX2 registration stops trying
  -       if ever not able to resolve (Reported by David Herselman)
  -  * ASTERISK-24211 - testsuite: Fix the dial_LS_options test
  -       (Reported by Matt Jordan)
  -  * ASTERISK-24249 - SIP debugs do not stop (Reported by Avinash
  -       Mohod)
  -  * ASTERISK-23577 - res_rtp_asterisk: Crash in
  -       ast_rtp_on_turn_rtp_state when RTP instance is NULL (Reported by
  -       Jay Jideliov)
  -  * ASTERISK-23634 - With TURN Asterisk crashes on multiple (7-10)
  -       concurrent WebRTC (avpg/encryption/icesupport) calls (Reported
  -       by Roman Skvirsky)
  -  * ASTERISK-24301 - Security: Out of call MESSAGE requests
  -       processed via Message channel driver can crash Asterisk
  -       (Reported by Matt Jordan)
  -
  - Improvements made in this release:
  - -----------------------------------
  -  * ASTERISK-24171 - [patch] Provide a manpage for the aelparse
  -       utility (Reported by Jeremy Lainé)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.13.0
* Mon Oct 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 11.12.1-1
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 11.6 and Asterisk 11 and 12. The available security releases are
  - released as versions 11.6-cert6, 11.12.1, and 12.5.1.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - Please note that the release of these versions resolves the following security
  - vulnerability:
  -
  - * AST-2014-010: Remote Crash when Handling Out of Call Message in Certain
  -                 Dialplan Configurations
  -
  - Additionally, the release of Asterisk 12.5.1 resolves the following security
  - vulnerability:
  -
  - * AST-2014-009: Remote Crash Based on Malformed SIP Subscription Requests
  -
  - Note that the crash described in AST-2014-010 can be worked around through
  - dialplan configuration. Given the likelihood of the issue, an advisory was
  - deemed to be warranted.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2014-009 and AST-2014-010, which were 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-11.6-cert6
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.12.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.5.1
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2014-009.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-010.pdf
* Mon Oct 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 11.12.0-1
  - The Asterisk Development Team has announced the release of Asterisk 11.12.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.12.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following are the issues resolved in this release:
  -
  - Bugs fixed in this release:
  - -----------------------------------
  -  * ASTERISK-23911 - URIENCODE/URIDECODE: WARNING about passing an
  -       empty string is a bit over zealous (Reported by Matt Jordan)
  -  * ASTERISK-23985 - PresenceState Action response does not contain
  -       ActionID; duplicates Message Header (Reported by Matt Jordan)
  -  * ASTERISK-23814 - No call started after peer dialed (Reported by
  -       Igor Goncharovsky)
  -  * ASTERISK-24087 - [patch]chan_sip: sip_subscribe_mwi_destroy
  -       should not call sip_destroy (Reported by Corey Farrell)
  -  * ASTERISK-23818 - PBX_Lua: after asterisk startup module is
  -       loaded, but dialplan not available (Reported by Dennis Guse)
  -  * ASTERISK-18345 - [patch] sips connection dropped by asterisk
  -       with a large INVITE (Reported by Stephane Chazelas)
  -  * ASTERISK-23508 - Memory Corruption in
  -       __ast_string_field_ptr_build_va (Reported by Arnd Schmitter)
  -
  - Improvements made in this release:
  - -----------------------------------
  -  * ASTERISK-21178 - Improve documentation for manager command
  -       Getvar, Setvar (Reported by Rusty Newton)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.12.0
* Mon Oct 20 2014 Jeffrey C. Ollie <jeff@ocjtech.us> - 11.11.0-1
  - The Asterisk Development Team has announced the release of Asterisk 11.11.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.11.0 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-22551 - Session timer : UAS (Asterisk) starts counting
  -       at Invite, UAC starts counting at 200 OK. (Reported by i2045)
  -  * ASTERISK-23792 - Mutex left locked in chan_unistim.c (Reported
  -       by Peter Whisker)
  -  * ASTERISK-23582 - [patch]Inconsistent column length in *odbc
  -       (Reported by Walter Doekes)
  -  * ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all
  -       categories but the requested one (Reported by zvision)
  -  * ASTERISK-23035 - ConfBridge with name longer than max (32 chars)
  -       results in several bridges with same conf_name (Reported by
  -       Iñaki Cívico)
  -  * ASTERISK-23824 - ConfBridge: Users cannot be muted via CLI or
  -       AMI when waiting to enter a conference (Reported by Matt Jordan)
  -  * ASTERISK-23683 - #includes - wildcard character in a path more
  -       than one directory deep - results in no config parsing on module
  -       reload (Reported by tootai)
  -  * ASTERISK-23827 - autoservice thread doesn't exit at shutdown
  -       (Reported by Corey Farrell)
  -  * ASTERISK-23609 - Security: AMI action MixMonitor allows
  -       arbitrary programs to be run (Reported by Corey Farrell)
  -  * ASTERISK-23673 - Security: DOS by consuming the number of
  -       allowed HTTP connections. (Reported by Richard Mudgett)
  -  * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite
  -       a DEBUG level of zero (Reported by Rusty Newton)
  -  * ASTERISK-23766 - [patch] Specify timeout for database write in
  -       SQLite (Reported by Igor Goncharovsky)
  -  * ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua
  -       with Lua 5.2 or greater due to addition of goto statement
  -       (Reported by Rusty Newton)
  -  * ASTERISK-23818 - PBX_Lua: after asterisk startup module is
  -       loaded, but dialplan not available (Reported by Dennis Guse)
  -  * ASTERISK-23834 - res_rtp_asterisk debug message gives wrong
  -       length if ICE (Reported by Richard Kenner)
  -  * ASTERISK-23790 - [patch] - SIP From headers longer than 256
  -       characters result in dropped call and 'No closing bracket'
  -       warnings. (Reported by uniken1)
  -  * ASTERISK-23917 - res_http_websocket: Delay in client processing
  -       large streams of data causes disconnect and stuck socket
  -       (Reported by Matt Jordan)
  -  * ASTERISK-23908 - [patch]When using FEC error correction,
  -       asterisk tries considers negative sequence numbers as missing
  -       (Reported by Torrey Searle)
  -  * ASTERISK-23921 - refcounter.py uses excessive ram for large refs
  -       files  (Reported by Corey Farrell)
  -  * ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against
  -       objects that were already freed (Reported by Corey Farrell)
  -  * ASTERISK-23916 - [patch]SIP/SDP fmtp line may include whitespace
  -       between attributes (Reported by Alexander Traud)
  -  * ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite()
  -       (Reported by Steve Davies)
  -  * ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking
  -       PI) in revision 413765 breaks working environments (Reported by
  -       Pavel Troller)
  -
  - Improvements made in this release:
  - -----------------------------------
  -  * ASTERISK-23492 - Add option to safe_asterisk to disable
  -       backgrounding (Reported by Walter Doekes)
  -  * ASTERISK-22961 - [patch] DTLS-SRTP not working with SHA-256
  -       (Reported by Jay Jideliov)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.11.0
* Thu Aug 28 2014 Jitka Plesnikova <jplesnik@redhat.com> - 11.10.2-2.2
  - Perl 5.20 rebuild
* Fri Aug 15 2014 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 11.10.2-2.1
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild
* Thu Jun 19 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.10.2-2:
  - Drop the 389 directory server schema (1061414)
* Thu Jun 19 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.10.2-1:
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security
  - releases are released as versions 1.8.15-cert7, 11.6-cert4, 1.8.28.2, 11.10.2,
  - and 12.3.2.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - These releases resolve security vulnerabilities that were previously fixed in
  - 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1, and 12.3.1. Unfortunately, the fix
  - for AST-2014-007 inadvertently introduced a regression in Asterisk's TCP and TLS
  - handling that prevented Asterisk from sending data over these transports. This
  - regression and the security vulnerabilities have been fixed in the versions
  - specified in this release announcement.
  -
  - The security patches for AST-2014-007 have been updated with the fix for the
  - regression, and are available at http://downloads.asterisk.org/pub/security
  -
  - Please note that the release of these versions resolves the following security
  - vulnerabilities:
  -
  - * AST-2014-005: Remote Crash in PJSIP Channel Driver's Publish/Subscribe
  -                 Framework
  -
  - * AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized
  -                 Shell Access
  -
  - * AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP
  -                 Connections
  -
  - * AST-2014-008: Denial of Service in PJSIP Channel Driver Subscriptions
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2014-005, AST-2014-006, AST-2014-007, and AST-2014-008,
  - which were released with the previous versions that addressed these
  - vulnerabilities.
  -
  - For a full list of changes in the current releases, please see the ChangeLogs:
  -
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.15-cert7
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.2
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert4
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.3.2
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2014-005.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-006.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-007.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-008.pdf
* Thu Jun 19 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.10.1-1:
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security
  - releases are released as versions 1.8.15-cert6, 11.6-cert3, 1.8.28.1, 11.10.1,
  - and 12.3.1.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of these versions resolves the following issue:
  -
  - * AST-2014-007: Denial of Service via Exhaustion of Allowed Concurrent HTTP
  -                 Connections
  -
  -   Establishing a TCP or TLS connection to the configured HTTP or HTTPS port
  -   respectively in http.conf and then not sending or completing a HTTP request
  -   will tie up a HTTP session. By doing this repeatedly until the maximum number
  -   of open HTTP sessions is reached, legitimate requests are blocked.
  -
  - Additionally, the release of 11.6-cert3, 11.10.1, and 12.3.1 resolves the
  - following issue:
  -
  - * AST-2014-006: Permission Escalation via Asterisk Manager User Unauthorized
  -                 Shell Access
  -
  -   Manager users can execute arbitrary shell commands with the MixMonitor manager
  -   action. Asterisk does not require system class authorization for a manager
  -   user to use the MixMonitor action, so any manager user who is permitted to use
  -   manager commands can potentially execute shell commands as the user executing
  -   the Asterisk process.
  -
  - Additionally, the release of 12.3.1 resolves the following issues:
  -
  - * AST-2014-005: Remote Crash in PJSIP Channel Driver's Publish/Subscribe
  -                 Framework
  -
  -   A remotely exploitable crash vulnerability exists in the PJSIP channel
  -   driver's pub/sub framework. If an attempt is made to unsubscribe when not
  -   currently subscribed and the endpoint's âsub_min_expiryâ is set to zero,
  -   Asterisk tries to create an expiration timer with zero seconds, which is not
  -   allowed, so an assertion raised.
  -
  - * AST-2014-008: Denial of Service in PJSIP Channel Driver Subscriptions
  -
  -   When a SIP transaction timeout caused a subscription to be terminated, the
  -   action taken by Asterisk was guaranteed to deadlock the thread on which SIP
  -   requests are serviced. Note that this behavior could only happen on
  -   established subscriptions, meaning that this could only be exploited if an
  -   attacker bypassed authentication and successfully subscribed to a real
  -   resource on the Asterisk server.
  -
  - These issues and their resolutions are described in the security advisories.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2014-005, AST-2014-006, AST-2014-007, and AST-2014-008,
  - which were 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-1.8.15-cert6
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.28.1
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert3
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.10.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.3.1
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2014-005.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-006.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-007.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-008.pdf
* Thu Jun 19 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.10.0-1:
  - The Asterisk Development Team has announced the release of Asterisk 11.10.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.10.0 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-23547 - [patch] app_queue removing callers from queue
  -       when reloading (Reported by Italo Rossi)
  -  * ASTERISK-23559 - app_voicemail fails to load after fix to
  -       dialplan functions (Reported by Corey Farrell)
  -  * ASTERISK-22846 - testsuite: masquerade super test fails on all
  -       branches (still) (Reported by Matt Jordan)
  -  * ASTERISK-23545 - Confbridge talker detection settings
  -       configuration load bug (Reported by John Knott)
  -  * ASTERISK-23546 - CB_ADD_LEN does not do what you'd think
  -       (Reported by Walter Doekes)
  -  * ASTERISK-23620 - Code path in app_stack fails to unlock list
  -       (Reported by Bradley Watkins)
  -  * ASTERISK-23616 - Big memory leak in logger.c (Reported by
  -       ibercom)
  -  * ASTERISK-23576 - Build failure on SmartOS / Illumos / SunOS
  -       (Reported by Sebastian Wiedenroth)
  -  * ASTERISK-23550 - Newer sound sets don't show up in menuselect
  -       (Reported by Rusty Newton)
  -  * ASTERISK-18331 - app_sms failure (Reported by David Woodhouse)
  -  * ASTERISK-19465 - P-Asserted-Identity Privacy (Reported by
  -       Krzysztof Chmielewski)
  -  * ASTERISK-23605 - res_http_websocket: Race condition in shutting
  -       down websocket causes crash (Reported by Matt Jordan)
  -  * ASTERISK-23707 - Realtime Contacts: Apparent mismatch between
  -       PGSQL database state and Asterisk state (Reported by Mark
  -       Michelson)
  -  * ASTERISK-23381 - [patch]ChanSpy- Barge only works on the initial
  -       'spy', if the spied-on channel makes a new call, unable to
  -       barge. (Reported by Robert Moss)
  -  * ASTERISK-23665 - Wrong mime type for codec H263-1998 (h263+)
  -       (Reported by Guillaume Maudoux)
  -  * ASTERISK-23664 - Incorrect H264 specification in SDP. (Reported
  -       by Guillaume Maudoux)
  -  * ASTERISK-22977 - chan_sip+CEL: missing ANSWER and PICKUP event
  -       for INVITE/w/replaces pickup (Reported by Walter Doekes)
  -  * ASTERISK-23709 - Regression in Dahdi/Analog/waitfordialtone
  -       (Reported by Steve Davies)
  -
  - Improvements made in this release:
  - -----------------------------------
  -  * ASTERISK-23649 - [patch]Support for DTLS retransmission
  -       (Reported by NITESH BANSAL)
  -  * ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently
  -       available in a CLI command (Reported by Patrick Laimbock)
  -  * ASTERISK-23754 - [patch] Use var/lib directory for log file
  -       configured in asterisk.conf (Reported by Igor Goncharovsky)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.10.0
* Sat Jun 07 2014 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 11.9.0-2.1
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild
* Thu May 15 2014 Dennis Gilmore <dennis@ausil.us> - 11.9.0-2
  - build against gmime-devel not gmime22-devel
  - do not use -m64 on aarch64
* Wed Apr 23 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.9.0-1:
  - The Asterisk Development Team has announced the release of Asterisk 11.9.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.9.0 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-22790 - check_modem_rate() may return incorrect rate
  -       for V.27 (Reported by Paolo Compagnini)
  -  * ASTERISK-23034 - [patch] manager Originate doesn't abort on
  -       failed format_cap allocation (Reported by Corey Farrell)
  -  * ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in
  -       sip.conf.sample (Reported by Eugene)
  -  * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
  -       minus signs (Reported by Jeremy Lainé)
  -  * ASTERISK-23046 - Custom CDR fields set during a GoSUB called
  -       from app_queue are not inserted (Reported by Denis Pantsyrev)
  -  * ASTERISK-23027 - [patch] Spelling typo "transfered" instead of
  -       "transferred" (Reported by Jeremy Lainé)
  -  * ASTERISK-23008 - Local channels loose CALLERID name when DAHDI
  -       channel connects (Reported by Michael Cargile)
  -  * ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted
  -       request and request queue may differ - fix for locking (Reported
  -       by adomjan)
  -  * ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image
  -       media offer due to invalid or unsupported syntax (Reported by
  -       adomjan)
  -  * ASTERISK-22861 - [patch]Specifying a null time as parameter to
  -       GotoIfTime or ExecIfTime causes segmentation fault (Reported by
  -       Sebastian Murray-Roberts)
  -  * ASTERISK-17837 - extconfig.conf - Maximum Include level (1)
  -       exceeded (Reported by pz)
  -  * ASTERISK-22662 - Documentation fix? - queues.conf says
  -       persistentmembers defaults to yes, it appears to lie (Reported
  -       by Rusty Newton)
  -  * ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot
  -       handle selinux port restrictions (Reported by Corey Farrell)
  -  * ASTERISK-23220 - STACK_PEEK function with no arguments causes
  -       crash/core dump (Reported by James Sharp)
  -  * ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload'
  -       command multiple times on cli_aliases (Reported by Joel Vandal)
  -  * ASTERISK-22757 - segfault in res_clialiases.so on reload when
  -       mapping "module reload" command (Reported by Gareth Blades)
  -  * ASTERISK-17727 - [patch] TLS doesn't get all certificate chain
  -       (Reported by LN)
  -  * ASTERISK-23178 - devicestate.h: device state setting functions
  -       are documented with the wrong return values (Reported by
  -       Jonathan Rose)
  -  * ASTERISK-23232 - LocalBridge AMI Event LocalOptimization value
  -       is opposite to what's expected (Reported by Leon Roy)
  -  * ASTERISK-23098 - [patch]possible null pointer dereference in
  -       format.c (Reported by Marcello Ceschia)
  -  * ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
  -       res_parking.so is not loaded, or if res_parking.conf has no
  -       configuration (Reported by CJ Oster)
  -  * ASTERISK-23069 - Custom CDR variable not recorded when set in
  -       macro called from app_queue (Reported by Bryan Anderson)
  -  * ASTERISK-19499 - ConfBridge MOH is not working for transferee
  -       after attended transfer (Reported by Timo Teräs)
  -  * ASTERISK-23261 - [patch]Output mixup in
  -       ${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
  -  * ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic
  -       payload change in rtp mapping in the 200 OK response (Reported
  -       by NITESH BANSAL)
  -  * ASTERISK-23255 - UUID included for Redhat, but missing for
  -       Debian distros in install_prereq script (Reported by Rusty
  -       Newton)
  -  * ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR
  -       variables for subsequent records (Reported by zvision)
  -  * ASTERISK-23141 - Asterisk crashes on Dial(), in
  -       pbx_find_extension at pbx.c (Reported by Maxim)
  -  * ASTERISK-23336 - Asterisk warning "Don't know how to indicate
  -       condition 33 on ooh323c" on outgoing calls from H323 to SIP peer
  -       (Reported by Alexander Semych)
  -  * 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-23135 - Crash - segfault in ast_channel_hangupcause_set
  -       - probably introduced in 11.7.0 (Reported by OK)
  -  * ASTERISK-23323 - [patch]chan_sip: missing p->owner checks in
  -       handle_response_invite (Reported by Walter Doekes)
  -  * ASTERISK-23406 - [patch]Fix typo in "sip show peer" (Reported by
  -       ibercom)
  -  * ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write
  -       (Reported by Jeremy Lainé)
  -  * ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call
  -       from hold (Reported by Vytis ValentinaviÄius)
  -  * ASTERISK-23104 - Specifying the SetVar AMI without a Channel
  -       cause Asterisk to crash (Reported by Joel Vandal)
  -  * ASTERISK-21930 - [patch]WebRTC over WSS is not working.
  -       (Reported by John)
  -  * ASTERISK-23383 - Wrong sense test on stat return code causes
  -       unchanged config check to break with include files. (Reported by
  -       David Woolley)
  -  * ASTERISK-20149 - Crash when faxing SIP to SIP with strictrtp set
  -       to yes (Reported by Alexandr Gordeev)
  -  * ASTERISK-17523 - Qualify for static realtime peers does not work
  -       (Reported by Maciej Krajewski)
  -  * ASTERISK-21406 - [patch] chan_sip deadlock on monlock between
  -       unload_module and do_monitor (Reported by Corey Farrell)
  -  * ASTERISK-23373 - [patch]Security: Open FD exhaustion with
  -       chan_sip Session-Timers (Reported by Corey Farrell)
  -  * ASTERISK-23340 - Security Vulnerability: stack allocation of
  -       cookie headers in loop allows for unauthenticated remote denial
  -       of service attack (Reported by Matt Jordan)
  -  * ASTERISK-23311 - Manager - MoH Stop Event fails to show up when
  -       leaving Conference (Reported by Benjamin Keith Ford)
  -  * ASTERISK-23420 - [patch]Memory leak in manager_add_filter
  -       function in manager.c (Reported by Etienne Lessard)
  -  * ASTERISK-23488 - Logic error in callerid checksum processing
  -       (Reported by Russ Meyerriecks)
  -  * ASTERISK-23461 - Only first user is muted when joining
  -       confbridge with 'startmuted=yes' (Reported by Chico Manobela)
  -  * ASTERISK-20841 - fromdomain not honored on outbound INVITE
  -       request (Reported by Kelly Goedert)
  -  * ASTERISK-22079 - Segfault: INTERNAL_OBJ (user_data=0x6374652f)
  -       at astobj2.c:120 (Reported by Jamuel Starkey)
  -  * ASTERISK-23509 - [patch]SayNumber for Polish language tries to
  -       play empty files for numbers divisible by 100 (Reported by
  -       zvision)
  -  * ASTERISK-23103 - [patch]Crash in ast_format_cmp, in ao2_find
  -       (Reported by JoshE)
  -  * ASTERISK-23391 - Audit dialplan function usage of channel
  -       variable (Reported by Corey Farrell)
  -  * ASTERISK-23548 - POST to ARI sometimes returns no body on
  -       success (Reported by Scott Griepentrog)
  -  * ASTERISK-23460 - ooh323 channel stuck if call is placed directly
  -       and gatekeeper is not available (Reported by Dmitry Melekhov)
  -
  - Improvements made in this release:
  - -----------------------------------
  -  * ASTERISK-22980 - [patch]Allow building cdr_radius and cel_radius
  -       against libfreeradius-client (Reported by Jeremy Lainé)
  -  * ASTERISK-22661 - Unable to exit ChanSpy if spied channel does
  -       not have a call in progress (Reported by Chris Hillman)
  -  * ASTERISK-23099 - [patch] WSS: enable ast_websocket_read()
  -       function to read the whole available data at first and then wait
  -       for any fragmented packets (Reported by Thava Iyer)
* Tue Mar 11 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.8.1-1:
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security
  - releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1,
  - and 12.1.1.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of these versions resolve the following issues:
  -
  - * AST-2014-001: Stack overflow in HTTP processing of Cookie headers.
  -
  -   Sending a HTTP request that is handled by Asterisk with a large number of
  -   Cookie headers could overflow the stack.
  -
  -   Another vulnerability along similar lines is any HTTP request with a
  -   ridiculous number of headers in the request could exhaust system memory.
  -
  - * AST-2014-002: chan_sip: Exit early on bad session timers request
  -
  -   This change allows chan_sip to avoid creation of the channel and
  -   consumption of associated file descriptors altogether if the inbound
  -   request is going to be rejected anyway.
  -
  - Additionally, the release of 12.1.1 resolves the following issue:
  -
  - * AST-2014-003: res_pjsip: When handling 401/407 responses don't assume a
  -   request will have an endpoint.
  -
  -   This change removes the assumption that an outgoing request will always
  -   have an endpoint and makes the authenticate_qualify option work once again.
  -
  - Finally, a security advisory, AST-2014-004, was released for a vulnerability
  - fixed in Asterisk 12.1.0. Users of Asterisk 12.0.0 are encouraged to upgrade to
  - 12.1.1 to resolve both vulnerabilities.
  -
  - These issues and their resolutions are described in the security advisories.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2014-001, AST-2014-002, AST-2014-003, and AST-2014-004,
  - which were 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-1.8.15-cert5
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.26.1
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.8.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.1.1
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2014-001.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-002.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-003.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-004.pdf
* Tue Mar 04 2014 Jeffrey Ollie <jeff@ocjtech.us> - 11.8.0-1:
  - The Asterisk Development Team has announced the release of Asterisk 11.8.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.8.0 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-22544 - Italian prompt vm-options has advertisement in
  -       it (Reported by Rusty Newton)
  -  * ASTERISK-21383 - STUN Binding Requests Not Being Sent Back from
  -       Asterisk to Chrome (Reported by Shaun Clark)
  -  * ASTERISK-22478 - [patch]Can't use pound(hash) symbol for custom
  -       DTMF menus in ConfBridge (processed as directive) (Reported by
  -       Nicolas Tanski)
  -  * ASTERISK-12117 - chan_sip creates a new local tag (from-tag) for
  -       every register message (Reported by Pawel Pierscionek)
  -  * ASTERISK-20862 - Asterisk min and max member penalties not
  -       honored when set with 0 (Reported by Schmooze Com)
  -  * ASTERISK-22746 - [patch]Crash in chan_dahdi during caller id
  -       read (Reported by Michael Walton)
  -  * ASTERISK-22788 - [patch] main/translate.c: access to variable f
  -       after free in ast_translate() (Reported by Corey Farrell)
  -  * ASTERISK-21242 - Segfault when T.38 re-invite retransmission
  -       receives 200 OK (Reported by Ashley Winters)
  -  * ASTERISK-22590 - BufferOverflow in unpacksms16() when receiving
  -       16 bit multipart SMS with app_sms (Reported by Jan Juergens)
  -  * ASTERISK-22905 - Prevent Asterisk functions that are 'dangerous'
  -       from being executed from external interfaces (Reported by Matt
  -       Jordan)
  -  * ASTERISK-23021 - Typos in code : "avaliable" instead of
  -       "available" (Reported by Jeremy Lainé)
  -  * ASTERISK-22970 - [patch]Documentation fix for QUOTE() (Reported
  -       by Gareth Palmer)
  -  * ASTERISK-21960 - ooh323 channels stuck (Reported by Dmitry
  -       Melekhov)
  -  * ASTERISK-22350 - DUNDI - core dump on shutdown - segfault in
  -       sqlite3_reset from /usr/lib/libsqlite3.so.0 (Reported by Birger
  -       "WIMPy" Harzenetter)
  -  * ASTERISK-22942 - [patch] - Asterisk crashed after
  -       Set(FAXOPT(faxdetect)=t38) (Reported by adomjan)
  -  * ASTERISK-22856 - [patch]SayUnixTime in polish reads minutes
  -       instead of seconds (Reported by Robert Mordec)
  -  * ASTERISK-22854 - [patch] - Deadlock between cel_pgsql unload and
  -       core_event_dispatcher taskprocessor thread (Reported by Etienne
  -       Lessard)
  -  * ASTERISK-22910 - [patch] - REPLACE() calls strcpy on overlapping
  -       memory when <replace-char> is empty (Reported by Gareth Palmer)
  -  * ASTERISK-22871 - cel_pgsql module not loading after "reload" or
  -       "reload cel_pgsql.so" command (Reported by Matteo)
  -  * ASTERISK-23084 - [patch]rasterisk needlessly prints the
  -       AST-2013-007 warning (Reported by Tzafrir Cohen)
  -  * ASTERISK-17138 - [patch] Asterisk not re-registering after it
  -       receives "Forbidden - wrong password on authentication"
  -       (Reported by Rudi)
  -  * ASTERISK-23011 - [patch]configure.ac and pbx_lua don't support
  -       lua 5.2 (Reported by George Joseph)
  -  * ASTERISK-22834 - Parking by blind transfer when lot full orphans
  -       channels (Reported by rsw686)
  -  * ASTERISK-23047 - Orphaned (stuck) channel occurs during a failed
  -       SIP transfer to parking space (Reported by Tommy Thompson)
  -  * ASTERISK-22946 - Local From tag regression with sipgate.de
  -       (Reported by Stephan Eisvogel)
  -  * ASTERISK-23010 - No BYE message sent when sip INVITE is received
  -       (Reported by Ryan Tilton)
  -  * ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
  -       - probably introduced in 11.7.0 (Reported by OK)
  -
  - Improvements made in this release:
  - -----------------------------------
  -  * ASTERISK-22728 - [patch] Improve Understanding Of 'Forcerport'
  -       When Running "sip show peers" (Reported by Michael L. Young)
  -  * ASTERISK-22659 - Make a new core and extra sounds release
  -       (Reported by Rusty Newton)
  -  * ASTERISK-22919 - core show channeltypes slicing  (Reported by
  -       outtolunc)
  -  * ASTERISK-22918 - dahdi show channels slices PRI channel dnid on
  -       output (Reported by outtolunc)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.8.0
* Sat Dec 28 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.7.0-1:
  - The Asterisk Development Team has announced the release of Asterisk 11.7.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.7.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * --- app_confbridge: Can now set the language used for announcements
  -       to the conference.
  -   (Closes issue ASTERISK-19983. Reported by Jonathan White)
  -
  - * --- app_queue: Fix CLI "queue remove member" queue_log entry.
  -   (Closes issue ASTERISK-21826. Reported by Oscar Esteve)
  -
  - * --- chan_sip: Do not increment the SDP version between 183 and 200
  -       responses.
  -   (Closes issue ASTERISK-21204. Reported by NITESH BANSAL)
  -
  - * --- chan_sip: Allow a sip peer to accept both AVP and AVPF calls
  -   (Closes issue ASTERISK-22005. Reported by Torrey Searle)
  -
  - * --- chan_sip: Fix Realtime Peer Update Problem When Un-registering
  -       And Expires Header In 200ok
  -   (Closes issue ASTERISK-22428. Reported by Ben Smithurst)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.7.0
* Sat Dec 28 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.6.1-1:
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security
  - releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4,
  - 10.12.4-digiumphones, and 11.6.1.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of these versions resolve the following issues:
  -
  - * A buffer overflow when receiving odd length 16 bit messages in app_sms. An
  -   infinite loop could occur which would overwrite memory when a message is
  -   received into the unpacksms16() function and the length of the message is an
  -   odd number of bytes.
  -
  - * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk
  -   now marks certain individual dialplan functions as 'dangerous', which will
  -   inhibit their execution from external sources.
  -
  -   A 'dangerous' function is one which results in a privilege escalation. For
  -   example, if one were to read the channel variable SHELL(rm -rf /) Bad
  -   Things(TM) could happen; even if the external source has only read
  -   permissions.
  -
  -   Execution from external sources may be enabled by setting 'live_dangerously'
  -   to 'yes' in the [options] section of asterisk.conf. Although doing so is not
  -   recommended.
  -
  - These issues and their resolutions are described in the security advisories.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2013-006 and AST-2013-007, which were
  - 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-1.8.15-cert4
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.2-cert3
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.24.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.4
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.4-digiumphones
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf
* Sat Dec 28 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.6.0-1:
  - The Asterisk Development Team has announced the release of Asterisk 11.6.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.6.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * --- Confbridge: empty conference not being torn down
  -   (Closes issue ASTERISK-21859. Reported by Chris Gentle)
  -
  - * --- Let Queue wrap up time influence member availability
  -   (Closes issue ASTERISK-22189. Reported by Tony Lewis)
  -
  - * --- Fix a longstanding issue with MFC-R2 configuration that
  -       prevented users
  -   (Closes issue ASTERISK-21117. Reported by Rafael Angulo)
  -
  - * --- chan_iax2: Fix saving the wrong expiry time in astdb.
  -   (Closes issue ASTERISK-22504. Reported by Stefan Wachtler)
  -
  - * --- Fix segfault for certain invalid WebSocket input.
  -   (Closes issue ASTERISK-21825. Reported by Alfred Farrugia)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0
* Mon Oct 21 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.5.1-3:
  - Disable hardened build, as it's apparently causing problems loading modules.
* Thu Aug 29 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.5.1-2:
  - Enable hardened build BZ#954338
  - Significant clean ups
* Thu Aug 29 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.5.1-1:
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases
  - are released as versions 1.8.15-cert2, 11.2-cert2, 1.8.23.1, 10.12.3, 10.12.3-digiumphones,
  - and 11.5.1.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of these versions resolve the following issues:
  -
  - * A remotely exploitable crash vulnerability exists in the SIP channel driver if
  -   an ACK with SDP is received after the channel has been terminated. The
  -   handling code incorrectly assumes that the channel will always be present.
  -
  - * A remotely exploitable crash vulnerability exists in the SIP channel driver if
  -   an invalid SDP is sent in a SIP request that defines media descriptions before
  -   connection information. The handling code incorrectly attempts to reference
  -   the socket address information even though that information has not yet been
  -   set.
  -
  - These issues and their resolutions are described in the security advisories.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2013-004 and AST-2013-005, which were
  - 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-1.8.15-cert3
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.2-cert2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.23.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.3
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.3-digiumphones
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.5.1
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2013-004.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2013-005.pdf
  -
  - The Asterisk Development Team has announced the release of Asterisk 11.5.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.5.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * --- Fix Segfault In app_queue When "persistentmembers" Is Enabled
  -       And Using Realtime
  -   (Closes issue ASTERISK-21738. Reported by JoshE)
  -
  - * --- IAX2: fix race condition with nativebridge transfers.
  -   (Closes issue ASTERISK-21409. Reported by alecdavis)
  -
  - * --- Fix The Payload Being Set On CN Packets And Do Not Set Marker
  -       Bit
  -   (Closes issue ASTERISK-21246. Reported by Peter Katzmann)
  -
  - * --- Fix One-Way Audio With auto_* NAT Settings When SIP Calls
  -       Initiated By PBX
  -   (Closes issue ASTERISK-21374. Reported by Michael L. Young)
  -
  - * --- chan_sip: NOTIFYs for BLF start queuing up and fail to be sent
  -       out after retries fail
  -   (Closes issue ASTERISK-21677. Reported by Dan Martens)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.5.0
* Sat Aug 03 2013 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 11.4.0-2.2
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild
* Wed Jul 17 2013 Petr Pisar <ppisar@redhat.com> - 11.4.0-2.1
  - Perl 5.18 rebuild
* Fri May 24 2013 Rex Dieter <rdieter@fedoraproject.org> 11.4.0-2
  - rebuild (libical)
* Mon May 20 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.4.0-1:
  - The Asterisk Development Team has announced the release of Asterisk 11.4.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.4.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * --- Fix Sorting Order For Parking Lots Stored In Static Realtime
  -   (Closes issue ASTERISK-21035. Reported by Alex Epshteyn)
  -
  - * --- Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On
  -       A Channel
  -   (Closes issue ASTERISK-21294. Reported by daroz)
  -
  - * --- When a session timer expires during a T.38 call, re-invite with
  -       correct SDP
  -   (Closes issue ASTERISK-21232. Reported by Nitesh Bansal)
  -
  - * --- Fix white noise on SRTP decryption
  -   (Closes issue ASTERISK-21323. Reported by andrea)
  -
  - * --- Fix reload skinny with active devices.
  -   (Closes issue ASTERISK-16610. Reported by wedhorn)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.4.0
* Fri May 10 2013 Tom Callaway <spot@fedoraproject.org> - 11.3.0-2:
  - fix build with lua 5.2
* Tue Apr 23 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.3.0-1:
  - The Asterisk Development Team has announced the release of Asterisk 11.3.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.3.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * --- Fix issue where chan_mobile fails to bind to first available
  -       port
  -   (Closes issue ASTERISK-16357. Reported by challado)
  -
  - * --- Fix Queue Log Reporting Every Call COMPLETECALLER With "h"
  -       Extension Present
  -   (Closes issue ASTERISK-20743. Reported by call)
  -
  - * --- Retain XMPP filters across reconnections so external modules
  -       continue to function as expected.
  -   (Closes issue ASTERISK-20916. Reported by kuj)
  -
  - * --- Ensure that a declined media stream is terminated with a '\r\n'
  -   (Closes issue ASTERISK-20908. Reported by Dennis DeDonatis)
  -
  - * --- Fix pjproject compilation in certain circumstances
  -   (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.3.0
* Thu Mar 28 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.2.2-1:
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.15 and Asterisk 1.8, 10, and 11. The available security releases
  - are released as versions 1.8.15-cert2, 1.8.20.2, 10.12.2, 10.12.2-digiumphones,
  - and 11.2.2.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of these versions resolve the following issues:
  -
  - * A possible buffer overflow during H.264 format negotiation. The format
  -   attribute resource for H.264 video performs an unsafe read against a media
  -   attribute when parsing the SDP.
  -
  -   This vulnerability only affected Asterisk 11.
  -
  - * A denial of service exists in Asterisk's HTTP server. AST-2012-014, fixed
  -   in January of this year, contained a fix for Asterisk's HTTP server for a
  -   remotely-triggered crash. While the fix prevented the crash from being
  -   triggered, a denial of service vector still exists with that solution if an
  -   attacker sends one or more HTTP POST requests with very large Content-Length
  -   values.
  -
  -   This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11
  -
  - * A potential username disclosure exists in the SIP channel driver. When
  -   authenticating a SIP request with alwaysauthreject enabled, allowguest
  -   disabled, and autocreatepeer disabled, Asterisk discloses whether a user
  -   exists for INVITE, SUBSCRIBE, and REGISTER transactions in multiple ways.
  -
  -   This vulnerability affects Certified Asterisk 1.8.15, Asterisk 1.8, 10, and 11
  -
  - These issues and their resolutions are described in the security advisories.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2013-001, AST-2013-002, and AST-2013-003, which were
  - 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-1.8.15-cert2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.20.2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.12.2-digiumphones
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.2.2
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2013-001.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2013-002.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2013-003.pdf
* Sun Feb 10 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.2.1-1:
  - The Asterisk Development Team has announced the release of Asterisk 11.2.1.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.2.1 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:
  -
  - * --- Fix astcanary startup problem due to wrong pid value from before
  -       daemon call
  -   (Closes issue ASTERISK-20947. Reported by Jakob Hirsch)
  -
  - * --- Update init.d scripts to handle stderr; readd splash screen for
  -       remote consoles
  -   (Closes issue ASTERISK-20945. Reported by Warren Selby)
  -
  - * --- Reset RTP timestamp; sequence number on SSRC change
  -   (Closes issue ASTERISK-20906. Reported by Eelco Brolman)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.1
* Fri Jan 18 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.2.0-1:
  - The Asterisk Development Team has announced the release of Asterisk 11.2.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.2.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * --- app_meetme: Fix channels lingering when hung up under certain
  -       conditions
  -   (Closes issue ASTERISK-20486. Reported by Michael Cargile)
  -
  - * --- Fix stuck DTMF when bridge is broken.
  -   (Closes issue ASTERISK-20492. Reported by Jeremiah Gowdy)
  -
  - * --- Add missing support for "who hung up" to chan_motif.
  -   (Closes issue ASTERISK-20671. Reported by Matt Jordan)
  -
  - * --- Remove a fixed size limitation for producing SDP and change how
  -       ICE support is disabled by default.
  -   (Closes issue ASTERISK-20643. Reported by coopvr)
  -
  - * --- Fix chan_sip websocket payload handling
  -   (Closes issue ASTERISK-20745. Reported by Iñaki Baz Castillo)
  -
  - * --- Fix pjproject compilation in certain circumstances
  -   (Closes issue ASTERISK-20681. Reported by Dinesh Ramjuttun)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.2.0
* Thu Jan 03 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.1.2-1:
  - The Asterisk Development Team has announced a security release for Asterisk 11,
  - Asterisk 11.1.2. This release addresses the security vulnerabilities reported in
  - AST-2012-014 and AST-2012-015, and replaces the previous version of Asterisk 11
  - released for these security vulnerabilities. The prior release left open a
  - vulnerability in res_xmpp that exists only in Asterisk 11; as such, other
  - versions of Asterisk were resolved correctly by the previous releases.
  -
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of these versions resolve the following two issues:
  -
  - * Stack overflows that occur in some portions of Asterisk that manage a TCP
  -   connection. In SIP, this is exploitable via a remote unauthenticated session;
  -   in XMPP and HTTP connections, this is exploitable via remote authenticated
  -   sessions. The vulnerabilities in SIP and HTTP were corrected in a prior
  -   release of Asterisk; the vulnerability in XMPP is resolved in this release.
  -
  - * A denial of service vulnerability through exploitation of the device state
  -   cache. Anonymous calls had the capability to create devices in Asterisk that
  -   would never be disposed of. Handling the cachability of device states
  -   aggregated via XMPP is handled in this release.
  -
  - These issues and their resolutions are described in the security advisories.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2012-014 and AST-2012-015.
  -
  - For a full list of changes in the current release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.2
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2012-014.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2012-015.pdf
  -
  - Thank you for your continued support of Asterisk - and we apologize for having
  - to do this twice!
* Wed Jan 02 2013 Jeffrey Ollie <jeff@ocjtech.us> - 11.1.1-1:
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.11 and Asterisk 1.8, 10, and 11. The available security releases
  - are released as versions 1.8.11-cert10, 1.8.19.1, 10.11.1, 10.11.1-digiumphones,
  - and 11.1.1.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of these versions resolve the following two issues:
  -
  - * Stack overflows that occur in some portions of Asterisk that manage a TCP
  -   connection. In SIP, this is exploitable via a remote unauthenticated session;
  -   in XMPP and HTTP connections, this is exploitable via remote authenticated
  -   sessions.
  -
  - * A denial of service vulnerability through exploitation of the device state
  -   cache. Anonymous calls had the capability to create devices in Asterisk that
  -   would never be disposed of.
  -
  - These issues and their resolutions are described in the security advisories.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2012-014 and AST-2012-015, which were 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-1.8.11-cert10
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.19.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.11.1-digiumphones
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.1.1
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2012-014.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2012-015.pdf
* Wed Dec 12 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.1.0-1:
  - The Asterisk Development Team has announced the release of Asterisk 11.1.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.1.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * --- Fix execution of 'i' extension due to uninitialized variable.
  -   (Closes issue ASTERISK-20455. Reported by Richard Miller)
  -
  - * --- Prevent resetting of NATted realtime peer address on reload.
  -   (Closes issue ASTERISK-18203. Reported by daren ferreira)
  -
  - * --- Fix ConfBridge crash if no timing module loaded.
  -   (Closes issue ASTERISK-19448. Reported by feyfre)
  -
  - * --- Fix the Park 'r' option when a channel parks itself.
  -   (Closes issue ASTERISK-19382. Reported by James Stocks)
  -
  - * --- Fix an issue where outgoing calls would fail to establish audio
  -       due to ICE negotiation failures.
  -   (Closes issue ASTERISK-20554. Reported by mmichelson)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.1.0
* Fri Dec 07 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.2-1:
  - The Asterisk Development Team has announced the release of Asterisk 11.0.2.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.0.2 resolves an issue reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is the issue resolved in this release:
  -
  - * --- chan_local: Fix local_pvt ref leak in local_devicestate().
  -   (Closes issue ASTERISK-20769. Reported by rmudgett)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.2
* Wed Dec 05 2012 Dan Horák <dan[at]danny.cz> - 11.0.1-3
  - simplify LDFLAGS setting
* Fri Nov 30 2012 Dennis Gilmore <dennis@ausil.us> - 11.0.1-2
  - clean up things to allow building on arm arches
* Mon Nov 05 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.1-1
  - The Asterisk Development Team has announced the release of Asterisk 11.0.1.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.0.1 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:
  -
  - * --- chan_sip: Fix a bug causing SIP reloads to remove all entries
  -       from the registry
  -   (Closes issue ASTERISK-20611. Reported by Alisher)
  -
  - * --- confbridge: Fix a bug which made conferences not record with
  -       AMI/CLI commands
  -   (Closes issue ASTERISK-20601. Reported by Vilius)
  -
  - * --- Fix an issue with res_http_websocket where the chan_sip
  -       WebSocket handler could not be registered.
  -   (Closes issue ASTERISK-20631. Reported by danjenkins)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.1
* Tue Oct 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.0-1:
  - The Asterisk Development Team is pleased to announce the release of
  - Asterisk 11.0.0.  This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - Asterisk 11 is the next major release series of Asterisk.  It is a Long Term
  - Support (LTS) release, similar to Asterisk 1.8.  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 11, please see the
  - Asterisk wiki:
  -
  - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11
  -
  - A short list of new features includes:
  -
  - * A new channel driver named chan_motif has been added which provides support
  -   for Google Talk and Jingle in a single channel driver.  This new channel
  -   driver includes support for both audio and video, RFC2833 DTMF, all codecs
  -   supported by Asterisk, hold, unhold, and ringing notification. It is also
  -   compliant with the current Jingle specification, current Google Jingle
  -   specification, and the original Google Talk protocol.
  -
  - * Support for the WebSocket transport for chan_sip.
  -
  - * SIP peers can now be configured to support negotiation of ICE candidates.
  -
  - * The app_page application now no longer depends on DAHDI or app_meetme. It
  -   has been re-architected to use app_confbridge internally.
  -
  - * Hangup handlers can be attached to channels using the CHANNEL() function.
  -   Hangup handlers will run when the channel is hung up similar to the h
  -   extension; however, unlike an h extension, a hangup handler is associated with
  -   the actual channel and will execute anytime that channel is hung up,
  -   regardless of where it is in the dialplan.
  -
  - * Added pre-dial handlers for the Dial and Follow-Me applications.  Pre-dial
  -   allows you to execute a dialplan subroutine on a channel before a call is
  -   placed but after the application performing a dial action is invoked. This
  -   means that the handlers are executed after the creation of the callee
  -   channels, but before any actions have been taken to actually dial the callee
  -   channels.
  -
  - * Log messages can now be easily associated with a certain call by looking at
  -   a new unique identifier, "Call Id".  Call ids are attached to log messages for
  -   just about any case where it can be determined that the message is related
  -   to a particular call.
  -
  - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
  -   Asterisk. Unlike traditional ACLs defined in specific module configuration
  -   files, Named ACLs can be shared across multiple modules.
  -
  - * The Hangup Cause family of functions and dialplan applications allow for
  -   inspection of the hangup cause codes for each channel involved in a call.
  -   This allows a dialplan writer to determine, for each channel, who hung up and
  -   for what reason(s).
  -
  - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
  -   lets you set some of the configuration options from the general section
  -   of features.conf on a per-channel basis. FEATUREMAP() lets you customize
  -   the key sequence used to activate built-in features, such as blindxfer,
  -   and automon.
  -
  - * Support for DTLS-SRTP in chan_sip.
  -
  - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups
  -   and callgroups to be defined for several channel drivers.
  -
  - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.
  -
  - More information about the new features can be found on the Asterisk wiki:
  -
  - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation
  -
  - A full list of all new features can also be found in the CHANGES file.
  -
  - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES
  -
  - For a full list of changes in the current release, please see the ChangeLog.
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0
* Wed Oct 17 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.0-0.7.rc2:
  - The Asterisk Development Team has announced the second release candidate of
  - Asterisk 11.0.0. This release candidate is available for immediate
  - download at http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 11.0.0-rc2 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 candidate:
  -
  - * --- Fix an issue where outgoing calls would fail to establish audio
  -       due to ICE negotiation failures.
  -   (Closes issue ASTERISK-20554. Reported by mmichelson)
  -
  - * --- Ensure Asterisk fails TCP/TLS SIP calls when certificate
  -       checking fails
  -   (Closes issue ASTERISK-20559. Reported by kmoore)
  -
  - * --- Don't make chan_sip export global symbols.
  -   (Closes issue ASTERISK-20545. Reported by kmoore)
  -
  - For a full list of changes in this release candidate, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.0-rc2
* Tue Oct 09 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.0-0.6.rc1
  - The Asterisk Development Team is pleased to announce the first release candidate
  - of Asterisk 11.0.0.  This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - All interested users of Asterisk are encouraged to participate in the
  - Asterisk 11 testing process.  Please report any issues found to the issue
  - tracker, https://issues.asterisk.org/jira.  It is also very useful to see
  - successful test reports.  Please post those to the asterisk-dev mailing list.
  - All Asterisk users are invited to participate in the #asterisk-testing channel
  - on IRC to work together in testing the many parts of Asterisk.
  -
  - Asterisk 11 is the next major release series of Asterisk.  It will be a Long
  - Term Support (LTS) release, similar to Asterisk 1.8.  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 11, please see the
  - Asterisk wiki:
  -
  - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11
  -
  - A short list of new features includes:
  -
  - * A new channel driver named chan_motif has been added which provides support
  -   for Google Talk and Jingle in a single channel driver.  This new channel
  -   driver includes support for both audio and video, RFC2833 DTMF, all codecs
  -   supported by Asterisk, hold, unhold, and ringing notification. It is also
  -   compliant with the current Jingle specification, current Google Jingle
  -   specification, and the original Google Talk protocol.
  -
  - * Support for the WebSocket transport for chan_sip.
  -
  - * SIP peers can now be configured to support negotiation of ICE candidates.
  -
  - * The app_page application now no longer depends on DAHDI or app_meetme. It
  -   has been re-architected to use app_confbridge internally.
  -
  - * Hangup handlers can be attached to channels using the CHANNEL() function.
  -   Hangup handlers will run when the channel is hung up similar to the h
  -   extension; however, unlike an h extension, a hangup handler is associated with
  -   the actual channel and will execute anytime that channel is hung up,
  -   regardless of where it is in the dialplan.
  -
  - * Added pre-dial handlers for the Dial and Follow-Me applications.  Pre-dial
  -   allows you to execute a dialplan subroutine on a channel before a call is
  -   placed but after the application performing a dial action is invoked. This
  -   means that the handlers are executed after the creation of the callee
  -   channels, but before any actions have been taken to actually dial the callee
  -   channels.
  -
  - * Log messages can now be easily associated with a certain call by looking at
  -   a new unique identifier, "Call Id".  Call ids are attached to log messages for
  -   just about any case where it can be determined that the message is related
  -   to a particular call.
  -
  - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
  -   Asterisk. Unlike traditional ACLs defined in specific module configuration
  -   files, Named ACLs can be shared across multiple modules.
  -
  - * The Hangup Cause family of functions and dialplan applications allow for
  -   inspection of the hangup cause codes for each channel involved in a call.
  -   This allows a dialplan writer to determine, for each channel, who hung up and
  -   for what reason(s).
  -
  - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
  -   lets you set some of the configuration options from the general section
  -   of features.conf on a per-channel basis. FEATUREMAP() lets you customize
  -   the key sequence used to activate built-in features, such as blindxfer,
  -   and automon.
  -
  - * Support for DTLS-SRTP in chan_sip.
  -
  - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups
  -   and callgroups to be defined for several channel drivers.
  -
  - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.
  -
  - More information about the new features can be found on the Asterisk wiki:
  -
  - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation
  -
  - A full list of all new features can also be found in the CHANGES file.
  -
  - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES
  -
  - For a full list of changes in the current release, please see the ChangeLog.
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-rc1
* Wed Sep 26 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.0-0.5.beta2
  - Don't forget format_ilbc module
* Wed Sep 26 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.0-0.4.beta2
  - The Asterisk Development Team is pleased to announce the second beta release of
  - Asterisk 11.0.0.  This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - All interested users of Asterisk are encouraged to participate in the
  - Asterisk 11 testing process.  Please report any issues found to the issue
  - tracker, https://issues.asterisk.org/jira.  It is also very useful to see
  - successful test reports.  Please post those to the asterisk-dev mailing list.
  - All Asterisk users are invited to participate in the #asterisk-testing channel
  - on IRC to work together in testing the many parts of Asterisk.
  -
  - Asterisk 11 is the next major release series of Asterisk.  It will be a Long
  - Term Support (LTS) release, similar to Asterisk 1.8.  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 11, please see the
  - Asterisk wiki:
  -
  - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11
  -
  - A short list of new features includes:
  -
  - * A new channel driver named chan_motif has been added which provides support
  -   for Google Talk and Jingle in a single channel driver.  This new channel
  -   driver includes support for both audio and video, RFC2833 DTMF, all codecs
  -   supported by Asterisk, hold, unhold, and ringing notification. It is also
  -   compliant with the current Jingle specification, current Google Jingle
  -   specification, and the original Google Talk protocol.
  -
  - * Support for the WebSocket transport for chan_sip.
  -
  - * SIP peers can now be configured to support negotiation of ICE candidates.
  -
  - * The app_page application now no longer depends on DAHDI or app_meetme. It
  -   has been re-architected to use app_confbridge internally.
  -
  - * Hangup handlers can be attached to channels using the CHANNEL() function.
  -   Hangup handlers will run when the channel is hung up similar to the h
  -   extension; however, unlike an h extension, a hangup handler is associated with
  -   the actual channel and will execute anytime that channel is hung up,
  -   regardless of where it is in the dialplan.
  -
  - * Added pre-dial handlers for the Dial and Follow-Me applications.  Pre-dial
  -   allows you to execute a dialplan subroutine on a channel before a call is
  -   placed but after the application performing a dial action is invoked. This
  -   means that the handlers are executed after the creation of the callee
  -   channels, but before any actions have been taken to actually dial the callee
  -   channels.
  -
  - * Log messages can now be easily associated with a certain call by looking at
  -   a new unique identifier, "Call Id".  Call ids are attached to log messages for
  -   just about any case where it can be determined that the message is related
  -   to a particular call.
  -
  - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
  -   Asterisk. Unlike traditional ACLs defined in specific module configuration
  -   files, Named ACLs can be shared across multiple modules.
  -
  - * The Hangup Cause family of functions and dialplan applications allow for
  -   inspection of the hangup cause codes for each channel involved in a call.
  -   This allows a dialplan writer to determine, for each channel, who hung up and
  -   for what reason(s).
  -
  - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
  -   lets you set some of the configuration options from the general section
  -   of features.conf on a per-channel basis. FEATUREMAP() lets you customize
  -   the key sequence used to activate built-in features, such as blindxfer,
  -   and automon.
  -
  - * Support for DTLS-SRTP in chan_sip.
  -
  - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups
  -   and callgroups to be defined for several channel drivers.
  -
  - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.
  -
  - More information about the new features can be found on the Asterisk wiki:
  -
  - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation
  -
  - A full list of all new features can also be found in the CHANGES file.
  -
  - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES
  -
  - For a full list of changes in the current release, please see the ChangeLog.
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta2
* Wed Sep 26 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.8.0-1
  - The Asterisk Development Team has announced the release of Asterisk 10.8.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 10.8.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * --- AST-2012-012: Resolve AMI User Unauthorized Shell Access through
  -       ExternalIVR
  -   (Closes issue ASTERISK-20132. Reported by Zubair Ashraf of IBM X-Force Research)
  -
  - * --- AST-2012-013: Resolve ACL rules being ignored during calls by
  -       some IAX2 peers
  -   (Closes issue ASTERISK-20186. Reported by Alan Frisch)
  -
  - * --- Handle extremely out of order RFC 2833 DTMF
  -   (Closes issue ASTERISK-18404. Reported by Stephane Chazelas)
  -
  - * --- Resolve severe memory leak in CEL logging modules.
  -   (Closes issue AST-916. Reported by Thomas Arimont)
  -
  - * --- Only re-create an SRTP session when needed
  -   (Issue ASTERISK-20194. Reported by Nicolo Mazzon)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.8.0
* Tue Sep 04 2012 Dan Horák <dan[at]danny.cz> - 11.0.0-0.3.beta1
  - fix build on s390
* Tue Sep 04 2012 Dan Horák <dan[at]danny.cz> - 10.7.1-2
  - fix build on s390
* Thu Aug 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.7.1-1
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are
  - released as versions 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of Asterisk 1.8.11-cert7, 1.8.15.1, 10.7.1, and 10.7.1-digiumphones
  - resolve the following two issues:
  -
  - * A permission escalation vulnerability in Asterisk Manager Interface.  This
  -   would potentially allow remote authenticated users the ability to execute
  -   commands on the system shell with the privileges of the user running the
  -   Asterisk application.  Please note that the README-SERIOUSLY.bestpractices.txt
  -   file delivered with Asterisk has been updated due to this and other related
  -   vulnerabilities fixed in previous versions of Asterisk.
  -
  - * When an IAX2 call is made using the credentials of a peer defined in a
  -   dynamic Asterisk Realtime Architecture (ARA) backend, the ACL rules for that
  -   peer are not applied to the call attempt. This allows for a remote attacker
  -   who is aware of a peer's credentials to bypass the ACL rules set for that
  -   peer.
  -
  - These issues and their resolutions are described in the security advisories.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2012-012 and AST-2012-013, which were 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-1.8.11-cert7
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.15.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.7.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.7.1-digiumphones
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2012-012.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2012-013.pdf
* Thu Aug 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.7.0-1
  - The Asterisk Development Team has announced the release of Asterisk 10.7.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 10.7.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * --- Fix deadlock potential with ast_set_hangupsource() calls.
  -   (Closes issue ASTERISK-19801. Reported by Alec Davis)
  -
  - * --- Fix request routing issue when outboundproxy is used.
  -   (Closes issue ASTERISK-20008. Reported by Marcus Hunger)
  -
  - * --- Set the Caller ID "tag" on peers even if remote party
  -       information is present.
  -   (Closes issue ASTERISK-19859. Reported by Thomas Arimont)
  -
  - * --- Fix NULL pointer segfault in ast_sockaddr_parse()
  -   (Closes issue ASTERISK-20006. Reported by Michael L. Young)
  -
  - * --- Do not perform install on existing directories
  -   (Closes issue ASTERISK-19492. Reported by Karl Fife)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.7.0
* Thu Aug 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.6.1-1
  - The Asterisk Development Team has announced the release of Asterisk 10.6.1.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 10.6.1 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:
  -
  - * --- Remove a superfluous and dangerous freeing of an SSL_CTX.
  -   (Closes issue ASTERISK-20074. Reported by Trevor Helmsley)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.1
* Thu Aug 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.6.0-1
  - The Asterisk Development Team has announced the release of Asterisk 10.6.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 10.6.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * --- format_mp3: Fix a possible crash in mp3_read().
  -   (Closes issue ASTERISK-19761. Reported by Chris Maciejewsk)
  -
  - * --- Fix local channel chains optimizing themselves out of a call.
  -   (Closes issue ASTERISK-16711. Reported by Alec Davis)
  -
  - * --- Re-add LastMsgsSent value for SIP peers
  -   (Closes issue ASTERISK-17866. Reported by Steve Davies)
  -
  - * --- Prevent sip_pvt refleak when an ast_channel outlasts its
  -       corresponding sip_pvt.
  -   (Closes issue ASTERISK-19425. Reported by David Cunningham)
  -
  - * --- Send more accurate identification information in dialog-info SIP
  -       NOTIFYs.
  -   (Closes issue ASTERISK-16735. Reported by Maciej Krajewski)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.6.0
* Sat Aug 18 2012 Jeffrey Ollie <jeff@ocjtech.us> - 11.0.0-0.2.beta1
  - The Asterisk Development Team is pleased to announce the first beta release of
  - Asterisk 11.0.0.  This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - All interested users of Asterisk are encouraged to participate in the
  - Asterisk 11 testing process.  Please report any issues found to the issue
  - tracker, https://issues.asterisk.org/jira.  It is also very useful to see
  - successful test reports.  Please post those to the asterisk-dev mailing list.
  - All Asterisk users are invited to participate in the #asterisk-testing channel
  - on IRC to work together in testing the many parts of Asterisk.
  -
  - Asterisk 11 is the next major release series of Asterisk.  It will be a Long
  - Term Support (LTS) release, similar to Asterisk 1.8.  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 11, please see the
  - Asterisk wiki:
  -
  - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11
  -
  - A short list of new features includes:
  -
  - * A new channel driver named chan_motif has been added which provides support
  -   for Google Talk and Jingle in a single channel driver.  This new channel
  -   driver includes support for both audio and video, RFC2833 DTMF, all codecs
  -   supported by Asterisk, hold, unhold, and ringing notification. It is also
  -   compliant with the current Jingle specification, current Google Jingle
  -   specification, and the original Google Talk protocol.
  -
  - * Support for the WebSocket transport for chan_sip.
  -
  - * SIP peers can now be configured to support negotiation of ICE candidates.
  -
  - * The app_page application now no longer depends on DAHDI or app_meetme. It
  -   has been re-architected to use app_confbridge internally.
  -
  - * Hangup handlers can be attached to channels using the CHANNEL() function.
  -   Hangup handlers will run when the channel is hung up similar to the h
  -   extension; however, unlike an h extension, a hangup handler is associated with
  -   the actual channel and will execute anytime that channel is hung up,
  -   regardless of where it is in the dialplan.
  -
  - * Added pre-dial handlers for the Dial and Follow-Me applications.  Pre-dial
  -   allows you to execute a dialplan subroutine on a channel before a call is
  -   placed but after the application performing a dial action is invoked. This
  -   means that the handlers are executed after the creation of the caller/callee
  -   channels, but before any actions have been taken to actually dial the callee
  -   channels.
  -
  - * Log messages can now be easily associated with a certain call by looking at
  -   a new unique identifier, "Call Id".  Call ids are attached to log messages for
  -   just about any case where it can be determined that the message is related
  -   to a particular call.
  -
  - * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
  -   Asterisk. Unlike traditional ACLs defined in specific module configuration
  -   files, Named ACLs can be shared across multiple modules.
  -
  - * The Hangup Cause family of functions and dialplan applications allow for
  -   inspection of the hangup cause codes for each channel involved in a call.
  -   This allows a dialplan writer to determine, for each channel, who hung up and
  -   for what reason(s).
  -
  - * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
  -   lets you set some of the configuration options from the general section
  -   of features.conf on a per-channel basis. FEATUREMAP() lets you customize
  -   the key sequence used to activate built-in features, such as blindxfer,
  -   and automon.
  -
  - * Support for named pickupgroups/callgroups, allowing any number of pickupgroups
  -   and callgroups to be defined for several channel drivers.
  -
  - * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.
  -
  - More information about the new features can be found on the Asterisk wiki:
  -
  - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation
  -
  - A full list of all new features can also be found in the CHANGES file.
  -
  - http://svnview.digium.com/svn/asterisk/branches/11/CHANGES
  -
  - For a full list of changes in the current release, please see the ChangeLog.
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0-beta1
* Wed Jul 18 2012 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 10.5.2-1.2
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild
* Mon Jul 09 2012 Petr Pisar <ppisar@redhat.com> - 10.5.2-1.1
  - Perl 5.16 rebuild
* Thu Jul 05 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.5.2-1:
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are
  - released as versions 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of Asterisk 1.8.11-cert4, 1.8.13.1, 10.5.2, and 10.5.2-digiumphones
  - resolve the following two issues:
  -
  - * If Asterisk sends a re-invite and an endpoint responds to the re-invite with
  -   a provisional response but never sends a final response, then the SIP dialog
  -   structure is never freed and the RTP ports for the call are never released. If
  -   an attacker has the ability to place a call, they could create a denial of
  -   service by using all available RTP ports.
  -
  - * If a single voicemail account is manipulated by two parties simultaneously,
  -   a condition can occur where memory is freed twice causing a crash.
  -
  - These issues and their resolution are described in the security advisories.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2012-010 and AST-2012-011, which were 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-1.8.11-cert4
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.13.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.2-digiumphones
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2012-010.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2012-011.pdf
* Thu Jun 28 2012 Petr Pisar <ppisar@redhat.com> - 10.5.1-1.1
  - Perl 5.16 rebuild
* Fri Jun 15 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.5.1-1
  - The Asterisk Development Team has announced a security release for Asterisk 10.
  - This security release is released as version 10.5.1.
  -
  - The release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of Asterisk 10.5.1 resolves the following issue:
  -
  - * A remotely exploitable crash vulnerability was found in the Skinny (SCCP)
  -  Channel driver. When an SCCP client sends an Off Hook message, followed by
  -  a Key Pad Button Message, a structure that was previously set to NULL is
  -  dereferenced.  This allows remote authenticated connections the ability to
  -  cause a crash in the server, denying services to legitimate users.
  -
  - This issue and its resolution is described in the security advisory.
  -
  - For more information about the details of this vulnerability, please read
  - security advisory AST-2012-009, which was released at the same time as this
  - announcement.
  -
  - For a full list of changes in the current releases, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.5.1
  -
  - The security advisory is available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2012-009.pdf
* Fri Jun 15 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.5.0-1
  - The Asterisk Development Team has announced the release of Asterisk 10.5.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 10.5.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * --- Turn off warning message when bind address is set to any.
  -  (Closes issue ASTERISK-19456. Reported by Michael L. Young)
  -
  - * --- Prevent overflow in calculation in ast_tvdiff_ms on 32-bit
  -      machines
  -  (Closes issue ASTERISK-19727. Reported by Ben Klang)
  -
  - * --- Make DAHDISendCallreroutingFacility wait 5 seconds for a reply
  -      before disconnecting the call.
  -  (Closes issue ASTERISK-19708. Reported by mehdi Shirazi)
  -
  - * --- Fix recalled party B feature flags for a failed DTMF atxfer.
  -  (Closes issue ASTERISK-19383. Reported by lgfsantos)
  -
  - * --- Fix DTMF atxfer running h exten after the wrong bridge ends.
  -  (Closes issue ASTERISK-19717. Reported by Mario)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.5.0
* Mon Jun 11 2012 Petr Pisar <ppisar@redhat.com> - 10.4.2-1.1
  - Perl 5.16 rebuild
* Wed May 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.4.2-1
  - The Asterisk Development Team has announced the release of Asterisk 10.4.2.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 10.4.2 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:
  -
  - * --- Resolve crash in subscribing for MWI notifications
  -  (Closes issue ASTERISK-19827. Reported by B. R)
  -
  - * --- Fix crash in ConfBridge when user announcement is played for
  -      more than 2 users
  -  (Closes issue ASTERISK-19899. Reported by Florian Gilcher)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.2
* Wed May 30 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.4.1-1
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.11 and Asterisk 1.8 and 10. The available security releases are
  - released as versions 1.8.11-cert2, 1.8.12.1, and 10.4.1.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of Asterisk 1.8.11-cert2, 1.8.12.1, and 10.4.1 resolve the following
  - two issues:
  -
  - * A remotely exploitable crash vulnerability exists in the IAX2 channel
  -  driver if an established call is placed on hold without a suggested music
  -  class. Asterisk will attempt to use an invalid pointer to the music
  -  on hold class name, potentially causing a crash.
  -
  - * A remotely exploitable crash vulnerability was found in the Skinny (SCCP)
  -  Channel driver. When an SCCP client closes its connection to the server,
  -  a pointer in a structure is set to NULL.  If the client was not in the
  -  on-hook state at the time the connection was closed, this pointer is later
  -  dereferenced. This allows remote authenticated connections the ability to
  -  cause a crash in the server, denying services to legitimate users.
  -
  - These issues and their resolution are described in the security advisories.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2012-007 and AST-2012-008, which were 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-1.8.11-cert2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.12.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.4.1
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2012-007.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2012-008.pdf
* Fri May 04 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.4.0-1
  - The Asterisk Development Team has announced the release of Asterisk 10.4.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 10.4.0 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:
  -
  - * --- Prevent chanspy from binding to zombie channels
  -  (Closes issue ASTERISK-19493. Reported by lvl)
  -
  - * --- Fix Dial m and r options and forked calls generating warnings
  -      for voice frames.
  -  (Closes issue ASTERISK-16901. Reported by Chris Gentle)
  -
  - * --- Remove ISDN hold restriction for non-bridged calls.
  -  (Closes issue ASTERISK-19388. Reported by Birger Harzenetter)
  -
  - * --- Fix copying of CDR(accountcode) to local channels.
  -  (Closes issue ASTERISK-19384. Reported by jamicque)
  -
  - * --- Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors
  -  (Closes issue ASTERISK-19303. Reported by Jon Tsiros)
  -
  - * --- Eliminate double close of file descriptor in manager.c
  -  (Closes issue ASTERISK-18453. Reported by Jaco Kroon)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.4.0
* Tue Apr 24 2012 Jeffrey Ollie <jeff@ocjtech.us> - 10.3.1-1
  - The Asterisk Development Team has announced security releases for Asterisk 1.6.2,
  - 1.8, and 10. The available security releases are released as versions 1.6.2.24,
  - 1.8.11.1, and 10.3.1.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of Asterisk 1.6.2.24, 1.8.11.1, and 10.3.1 resolve the following two
  - issues:
  -
  -  * A permission escalation vulnerability in Asterisk Manager Interface.  This
  -   would potentially allow remote authenticated users the ability to execute
  -   commands on the system shell with the privileges of the user running the
  -   Asterisk application.
  -
  -  * A heap overflow vulnerability in the Skinny Channel driver.  The keypad
  -   button message event failed to check the length of a fixed length buffer
  -   before appending a received digit to the end of that buffer.  A remote
  -   authenticated user could send sufficient keypad button message events that the
  -   buffer would be overrun.
  -
  - In addition, the release of Asterisk 1.8.11.1 and 10.3.1 resolve the following
  - issue:
  -
  -  * A remote crash vulnerability in the SIP channel driver when processing UPDATE
  -   requests.  If a SIP UPDATE request was received indicating a connected line
  -   update after a channel was terminated but before the final destruction of the
  -   associated SIP dialog, Asterisk would attempt a connected line update on a
  -   non-existing channel, causing a crash.
  -
  - These issues and their resolution are described in the security advisories.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2012-004, AST-2012-005, and AST-2012-006, which were
  - 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/asterisk/releases/ChangeLog-1.6.2.24
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.11.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.3.1
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2012-004.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2012-005.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2012-006.pdf
* Thu Mar 29 2012 Russell Bryant <russell@russellbryant.net> - 10.3.0-1
  - Update to 10.3.0
* Fri Mar 16 2012 Russell Bryant <russell@russellbryant.net> - 10.2.1-1
  - Update to 10.2.1 from upstream.
  - Fix remote stack overflow in app_milliwatt.
  - Fix remote stack overflow, including possible code injection, in HTTP digest
    authentication handling.
  - Disable asterisk-corosync package, as it doesn't build right now.
  - Resolves: rhbz#804045, rhbz#804038, rhbz#804042
* Thu Feb 16 2012 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.1.2-2
  - * Add patch extracted from upstream to build with Corosync since
  -   OpenAIS is no longer available.
  - * Add PrivateTmp=true to systemd service file (#782478)
  - * Add some macros to make it easier to build with fewer dependencies
  -   (with corresponding less functionality) (#787389)
  - * Add isa macros in a few places plus a few other changes to make it
  -   easier to cross-compile. (#787779)
* Thu Feb 16 2012 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.1.2-1
  - The Asterisk Development Team has announced the release of Asterisk 10.1.2. This
  - release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 10.1.2 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:
  -
  - * --- Fix SIP INFO DTMF handling for non-numeric codes ---
  -  (Closes issue ASTERISK-19290. Reported by: Ira Emus)
  -
  - * --- Fix crash in ParkAndAnnounce ---
  -  (Closes issue ASTERISK-19311. Reported-by: tootai)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.2
* Thu Feb 16 2012 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.1.1-1
  - The Asterisk Development Team has announced the release of Asterisk 10.1.1. This
  - release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 10.1.1 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * --- Fixes deadlocks occuring in chan_agent ---
  -  (Closes issue ASTERISK-19285. Reported by: Alex Villacis Lasso)
  -
  - * --- Ensure entering T.38 passthrough does not cause an infinite loop ---
  -  (Closes issue ASTERISK-18951. Reported-by: Kristijan Vrban)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.1
* Thu Feb 16 2012 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.1.0-1
  - The Asterisk Development Team is pleased to announce the release of
  - Asterisk 10.1.0. This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 10.1.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * AST-2012-001: prevent crash when an SDP offer
  -  is received with an encrypted video stream when support for video
  -  is disabled and res_srtp is loaded.  (closes issue ASTERISK-19202)
  -  Reported by: Catalin Sanda
  -
  - * Allow playback of formats that don't support seeking.  ast_streamfile
  -  previously did unconditional seeking on files that broke playback of
  -  formats that don't support that functionality.  This patch avoids the
  -  seek that was causing the problem.
  -  (closes issue ASTERISK-18994) Patched by: Timo Teras
  -
  - * Add pjmedia probation concepts to res_rtp_asterisk's learning mode.  In
  -  order to better handle RTP sources with strictrtp enabled (which is the
  -  default setting in 10) using the learning mode to figure out new sources
  -  when they change is handled by checking for a number of consecutive (by
  -  sequence number) packets received to an rtp struct based on a new
  -  configurable value called 'probation'.  Also, during learning mode instead
  -  of liberally accepting all packets received, we now reject packets until a
  -  clear source has been determined.
  -
  - * Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop.  Failing
  -  to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
  -  causes the loop to exit prematurely. This causes a variety of negative side
  -  effects, depending on when the loop exits. This patch handles the frame by
  -  essentially swallowing the frame in the local loop, as the current channel
  -  drivers expect the RTP bridge to handle the frame, and, in the case of the
  -  local bridge loop, no additional action is necessary.
  -  (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
  -  by: Matt Jordan
  -
  - * Fix timing source dependency issues with MOH.  Prior to this patch,
  -  res_musiconhold existed at the same module priority level as the timing
  -  sources that it depends on.  This would cause a problem when music on
  -  hold was reloaded, as the timing source could be changed after
  -  res_musiconhold was processed. This patch adds a new module priority
  -  level, AST_MODPRI_TIMING, that the various timing modules are now loaded
  -  at. This now occurs before loading other resource modules, such
  -  that the timing source is guaranteed to be set prior to resolving
  -  the timing source dependencies.
  -  (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H,
  -  Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
  -  Patched by elguero
  -
  - * Fix RTP reference leak.  If a blind transfer were initiated using a
  -  REFER without a prior reINVITE to place the call on hold, AND if Asterisk
  -  were sending RTCP reports, then there was a reference leak for the
  -  RTP instance of the transferrer.
  -  (closes issue ASTERISK-19192) Reported by: Tyuta Vitali
  -
  - * Fix blind transfers from failing if an 'h' extension
  -  is present.  This prevents the 'h' extension from being run on the
  -  transferee channel when it is transferred via a native transfer
  -  mechanism such as SIP REFER.  (closes issue ASTERISK-19173) Reported
  -  by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
  -  Mark Michelson (license 5049)
  -
  - * Restore call progress code for analog ports. Extracting sig_analog
  -  from chan_dahdi lost call progress detection functionality.  Fix
  -  analog ports from considering a call answered immediately after
  -  dialing has completed if the callprogress option is enabled.
  -  (closes issue ASTERISK-18841)
  -  Reported by: Richard Miller Patched by Richard Miller
  -
  - * Fix regression that 'rtp/rtcp set debup ip' only works when a port
  -  was also specified.
  -  (closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by:
  -  Walter Doekes
  -
  - For a full list of changes in this release candidate, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.0
* Thu Feb 16 2012 Russell Bryant <russellb@fedoraproject.org> - 10.0.0-2
  - Remove asterisk-ais.  OpenAIS was removed from Fedora.
* Thu Jan 12 2012 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 10.0.0-1.1
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild
* Tue Jan 03 2012 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-1
  - Don't build API docs as the build never finishes
* Thu Dec 15 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-1
  - The Asterisk Development Team is proud to announce the release of
  - Asterisk 10.0.0. This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - Asterisk 10 is the next major release series of Asterisk. It will be a
  - Standard support release, similar to Asterisk 1.6.2. For more information about
  - support time lines for Asterisk releases, see the Asterisk versions page:
  -
  -  https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
  -
  - With the release of the Asterisk 10 branch, the preceding '1.' has been removed
  - from the version number per the blog post available at
  -
  -
  - http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/
  -
  - The release of Asterisk 10 would not have been possible without the support and
  - contributions of the community.
  -
  - You can find an overview of the work involved with the 10.0.0 release in the
  - summary:
  -
  - http://svn.asterisk.org/svn/asterisk/tags/10.0.0/asterisk-10.0.0-summary.txt
  -
  - A short list of available features includes:
  -
  - * T.38 gateway functionality has been added to res_fax.
  - * Protocol independent out-of-call messaging support. Text messages not
  -  associated with an active call can now be routed through the Asterisk
  -  dialplan. SIP and XMPP are supported so far.
  - * New highly optimized and customizable ConfBridge application capable of mixing
  -  audio at sample rates ranging from 8kHz-192kHz
  - * Addition of video_mode option in confbridge.conf to provide basic video
  -  conferencing in the ConfBridge() dialplan application.
  - * Support for defining hints has been added to pbx_lua.
  - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
  - * Much, much more!
  -
  - A full list of new features can be found in the CHANGES file.
  -
  -  http://svn.asterisk.org/svn/asterisk/branches/10/CHANGES
  -
  - Also, when upgrading a system between major versions, it is imperative that you
  - read and understand the contents of the UPGRADE.txt file, which is located at:
  -
  -  http://svn.asterisk.org/svn/asterisk/branches/10/UPGRADE.txt
* Fri Dec 09 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.7.rc3
  - The Asterisk Development Team has announced the third release candidate of
  - Asterisk 10.0.0. This release candidate is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 10.0.0-rc3 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release candidate:
  -
  - * Add ASTSBINDIR to the list of configurable paths
  -
  -  This patch also makes astdb2sqlite3 and astcanary use the configured
  -  directory instead of relying on $PATH.
  -
  - * Don't crash on INFO automon request with no channel
  -
  -  AST-2011-014. When automon was enabled in features.conf, it was possible
  -  to crash Asterisk by sending an INFO request if no channel had been
  -  created yet.
  -
  - * Fixed crash from orphaned MWI subscriptions in chan_sip
  -
  -  This patch resolves the issue where MWI subscriptions are orphaned
  -  by subsequent SIP SUBSCRIBE messages.
  -
  - * Fix a change in behavior in 'database show' from 1.8.
  -
  -  In 1.8 and previous versions, one could use any fullword portion of
  -  the key name, including the full key, to obtain the record. Until this
  -  patch, this did not work for the full key.
  -
  - * Default to nat=yes; warn when nat in general and peer differ
  -
  -  AST-2011-013.  It is possible to enumerate SIP usernames when the general and
  -  user/peer nat settings differ in whether to respond to the port a request is
  -  sent from or the port listed for responses in the Via header. In 1.4 and
  -  1.6.2, this would mean if one setting was nat=yes or nat=route and the other
  -  was either nat=no or nat=never. In 1.8 and 10, this would mean when one
  -  was nat=force_rport and the other was nat=no.
  -
  -  In order to address this problem, it was decided to switch the default
  -  behavior to nat=yes/force_rport as it is the most commonly used option
  -  and to strongly discourage setting nat per-peer/user when at all
  -  possible.
  -
  - * Fixed SendMessage stripping extension from To: header in SIP MESSAGE
  -
  -  When using the MessageSend application to send a SIP MESSAGE to a
  -  non-peer, chan_sip stripped off the extension and failed to add it back
  -  to the sip_pvt structure before transmitting. This patch adds the full
  -  URI passed in from the message core to the sip_pvt structure.
  -
  - For a full list of changes in this release candidate, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc3
* Wed Nov 16 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.6.rc2
  - The Asterisk Development Team has announced the second release candidate of
  - Asterisk 10.0.0. This release candidate is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 10.0.0-rc2 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release candidate:
  -
  - * Ensure that a null vmexten does not cause a segfault
  -
  - * Fix issue with ConfBridge participants hanging up during DTMF feature
  -  menu usage getting stuck in conference forever
  -  (closes issue ASTERISK-18829)
  -  Reported by: zvision
  -
  - * Fix app_macro.c MODULEINFO section termination
  -  (closes issue ASTERISK-18848)
  -  Reported by: Tony Mountifield
  -
  - For a full list of changes in this release candidate, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.0.0-rc2
* Fri Nov 11 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.5.rc1
  - The Asterisk Development Team is pleased to announce the first release candidate
  - of Asterisk 10.0.0. This release candidate is available for immediate download
  - at http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - All Asterisk users are encouraged to participate in the Asterisk 10 testing
  - process. Please report any issues found to the issue tracker,
  - https://issues.asterisk.org/jira. It is also very useful to see successful test
  - reports. Please post those to the asterisk-dev mailing list.
  -
  - All Asterisk users are invited to participate in the #asterisk-testing
  - channel on IRC to work together in testing the many parts of Asterisk.
  -
  - Asterisk 10 is the next major release series of Asterisk. It will be a
  - Standard support release, similar to Asterisk 1.6.2. For more
  - information about support time lines for Asterisk releases, see the Asterisk
  - versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
  -
  - A short list of features includes:
  -
  - * T.38 gateway functionality has been added to res_fax.
  - * Protocol independent out-of-call messaging support. Text messages not
  -  associated with an active call can now be routed through the Asterisk
  -  dialplan. SIP and XMPP are supported so far.
  - * New highly optimized and customizable ConfBridge application capable of mixing
  -  audio at sample rates ranging from 8kHz-192kHz
  -  (More information available at
  -   https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 )
  - * Addition of video_mode option in confbridge.conf to provide basic video
  -  conferencing in the ConfBridge() dialplan application.
  - * Support for defining hints has been added to pbx_lua.
  - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
  - * Much, much more!
  -
  - A full list of new features can be found in the CHANGES file.
  -
  - http://svnview.digium.com/svn/asterisk/branches/10/CHANGES
  -
  - For a full list of changes in the current release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-rc1
* Tue Oct 18 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.4.beta2
  - Add patch from upstream SVN to fix AST-2011-012
* Fri Oct 14 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.3.beta2
  - Patch cleanup day
* Thu Sep 29 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.2.beta2
  - The Asterisk Development Team is pleased to announce the second beta release of
  - Asterisk 10.0.0. This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - With the release of the Asterisk 10 branch, the preceding '1.' has been removed
  - from the version number per the blog post available at
  - http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/
  -
  - All interested users of Asterisk are encouraged to participate in the
  - Asterisk 10 testing process. Please report any issues found to the issue
  - tracker, https://issues.asterisk.org/jira. It is also very useful to see
  - successful test reports. Please post those to the asterisk-dev mailing list.
  -
  - All Asterisk users are invited to participate in the #asterisk-testing
  - channel on IRC to work together in testing the many parts of Asterisk.
  -
  - Asterisk 10 is the next major release series of Asterisk. It will be a
  - Standard support release, similar to Asterisk 1.6.2. For more
  - information about support time lines for Asterisk releases, see the Asterisk
  - versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
  -
  - A short list of features includes:
  -
  - * T.38 gateway functionality has been added to res_fax.
  -
  - * Protocol independent out-of-call messaging support. Text messages not
  -  associated with an active call can now be routed through the Asterisk
  -  dialplan. SIP and XMPP are supported so far.
  -
  - * New highly optimized and customizable ConfBridge application capable of mixing
  -  audio at sample rates ranging from 8kHz-192kHz
  -
  - * Addition of video_mode option in confbridge.conf to provide basic video
  -  conferencing in the ConfBridge() dialplan application.
  -
  - * Support for defining hints has been added to pbx_lua.
  -
  - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
  -
  - * Much, much more!
  -
  - A full list of new features can be found in the CHANGES file.
  -
  - http://svnview.digium.com/svn/asterisk/branches/10/CHANGES
  -
  - For a full list of changes in the current release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta2
* Mon Jul 25 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 10.0.0-0.1.beta1
  -
  - The Asterisk Development Team is pleased to announce the first beta release of
  - Asterisk 10.0.0-beta1. This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - With the release of the Asterisk 10 branch, the preceding '1.' has been removed
  - from the version number per the blog post available at
  - http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/
  -
  - All interested users of Asterisk are encouraged to participate in the
  - Asterisk 10 testing process. Please report any issues found to the issue
  - tracker, https://issues.asterisk.org/jira. It is also very useful to see
  - successful test reports. Please post those to the asterisk-dev mailing list.
  -
  - All Asterisk users are invited to participate in the #asterisk-testing
  - channel on IRC to work together in testing the many parts of Asterisk.
  - Additionally users can make use of the RPM and DEB packages now being built for
  - all Asterisk releases. More information available at
  - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
  -
  - Asterisk 10 is the next major release series of Asterisk. It will be a
  - Standard support release, similar to Asterisk 1.6.2. For more
  - information about support time lines for Asterisk releases, see the Asterisk
  - versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
  -
  - A short list of included features includes:
  -
  - * T.38 gateway functionality has been added to res_fax.
  - * Protocol independent out-of-call messaging support.  Text messages not
  -  associated with an active call can now be routed through the Asterisk
  -  dialplan.  SIP and XMPP are supported so far.
  - * New highly optimized and customizable ConfBridge application capable of mixing
  -  audio at sample rates ranging from 8kHz-192kHz
  - * Addition of video_mode option in confbridge.conf to provide basic video
  -  conferencing in the ConfBridge() dialplan application.
  - * Support for defining hints has been added to pbx_lua.
  - * Replacement of Berkeley DB with SQLite for the Asterisk Database (AstDB).
  - * Much, much more!
  -
  - A full list of new features can be found in the CHANGES file.
  -
  - http://svn.digium.com/view/asterisk/branches/10/CHANGES?view=checkout
  -
  - For a full list of changes in the current release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-10.0.0-beta1
* Thu Jul 21 2011 Petr Sabata <contyk@redhat.com> - 1.8.5.0-1.2
  - Perl mass rebuild
* Wed Jul 20 2011 Petr Sabata <contyk@redhat.com> - 1.8.5.0-1.1
  - Perl mass rebuild
* Mon Jul 11 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.5.0-1
  - The Asterisk Development Team announces the release of Asterisk 1.8.5.0. This
  - release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 1.8.5.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * Fix Deadlock with attended transfer of SIP call
  -  (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,
  -  cmaj)
  -
  - * Fixes thread blocking issue in the sip TCP/TLS implementation.
  -  (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,
  -  rossbeer, kowalma, Freddi_Fonet)
  -
  - * Be more tolerant of what URI we accept for call completion PUBLISH requests.
  -  (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)
  -
  - * Fix a nasty chanspy bug which was causing a channel leak every time a spied on
  -  channel made a call.
  -  (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
  -
  - * This patch fixes a bug with MeetMe behavior where the 'P' option for always
  -  prompting for a pin is ignored for the first caller.
  -  (Closes issue #18070. Reported by mav3rick. Patched by bbryant)
  -
  - * Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
  -  the call that the dialplan started an AGI script for is hungup while the AGI
  -  script is in the middle of a command then the AGI script is not notified of
  -  the hangup.
  -  (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
  -
  - * Resolve issue where leaving a voicemail, the MWI message is never sent. The
  -  same thing happens when checking a voicemail and marking it as read.
  -  (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard
  -  Mudgett)
  -
  - * Resolve issue where wait for leader with Music On Hold allows crosstalk
  -  between participants. Parenthesis in the wrong position. Regression from issue
  -  #14365 when expanding conference flags to use 64 bits.
  -  (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5.0
* Thu Jul 07 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.5-0.2
  - Rebuild for net-snmp 5.7
* Fri Jul 01 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.5-0.1.rc1
  - Fix systemd dependencies in EL6 and F15
* Thu Jun 30 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.5-0.1.rc1
  - The Asterisk Development Team has announced the first release candidate of
  - Asterisk 1.8.5. This release candidate is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 1.8.5-rc1 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release candidate:
  -
  - * Fix Deadlock with attended transfer of SIP call
  -  (Closes issue #18837. Reported, patched by alecdavis. Tested by Irontec, ZX81,
  -   cmaj)
  -
  - * Fixes thread blocking issue in the sip TCP/TLS implementation.
  -  (Closes issue #18497. Reported by vois. Patched by dvossel. Tested by vois,
  -   rossbeer, kowalma, Freddi_Fonet)
  -
  - * Be more tolerant of what URI we accept for call completion PUBLISH requests.
  -  (Closes issue #18946. Reported by GeorgeKonopacki. Patched by mmichelson)
  -
  - * Fix a nasty chanspy bug which was causing a channel leak every time a spied on
  -  channel made a call.
  -  (Closes issue #18742. Reported by jkister. Tested by jcovert, jrose)
  -
  - * This patch fixes a bug with MeetMe behavior where the 'P' option for always
  -  prompting for a pin is ignored for the first caller.
  -  (Closes issue #18070. Reported by mav3rick. Patched by bbryant)
  -
  - * Fix issue where Asterisk does not hangup a channel after endpoint hangs up. If
  -  the call that the dialplan started an AGI script for is hungup while the AGI
  -  script is in the middle of a command then the AGI script is not notified of
  -  the hangup.
  -  (Closes issue #17954, #18492. Reported by mn3250, devmod. Patched by rmudgett)
  -
  - * Resolve issue where leaving a voicemail, the MWI message is never sent. The
  -  same thing happens when checking a voicemail and marking it as read.
  -  (Closes issue ASTERISK-18002. Reported by Leif Madsen. Resolved by Richard
  -   Mudgett)
  -
  - * Resolve issue where wait for leader with Music On Hold allows crosstalk
  -  between participants. Parenthesis in the wrong position. Regression from issue
  -  #14365 when expanding conference flags to use 64 bits.
  -  (Closes issue #18418. Reported by MrHanMan. Patched by rmudgett)
  -
  - * Fix timerfd locking issue.
  -  (Closes ASTERISK-17867, ASTERISK-17415. Patched by kobaz)
  -
  - For a full list of changes in this release candidate, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.5-rc1
* Thu Jun 30 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.4.4-2
  - Fedora Directory Server -> 389 Directory Server
* Wed Jun 29 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.4.4-1
  - The Asterisk Development Team has announced the release of Asterisk
  - versions 1.4.41.2, 1.6.2.18.2, and 1.8.4.4, which are security
  - releases.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 resolves the
  - following issue:
  -
  - AST-2011-011: Asterisk may respond differently to SIP requests from an
  - invalid SIP user than it does to a user configured on the system, even
  - when the alwaysauthreject option is set in the configuration. This can
  - leak information about what SIP users are valid on the Asterisk
  - system.
  -
  - For more information about the details of this vulnerability, please
  - read the security advisory AST-2011-011, which was released at the
  - same time as this announcement.
  -
  - For a full list of changes in the current releases, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.41.2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.4
  -
  - Security advisory AST-2011-011 is available at:
  -
  - http://downloads.asterisk.org/pub/security/AST-2011-011.pdf
* Mon Jun 27 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.4.3-3
  - Don't forget stereorize
* Mon Jun 27 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.4.3-2
  - Move /var/run/asterisk to /run/asterisk
  - Add comments to systemd service file on how to mimic safe_asterisk functionality
  - Build more of the optional binaries
  - Install the tmpfiles.d configuration on Fedora 15
* Fri Jun 24 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.4.3-1
  - The Asterisk Development Team has announced the release of Asterisk versions
  - 1.4.41.1, 1.6.2.18.1, and 1.8.4.3, which are security releases.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of Asterisk 1.4.41.1, 1.6.2.18, and 1.8.4.3 resolves several issues
  - as outlined below:
  -
  - * AST-2011-008: If a remote user sends a SIP packet containing a null,
  -  Asterisk assumes available data extends past the null to the
  -  end of the packet when the buffer is actually truncated when
  -  copied.  This causes SIP header parsing to modify data past
  -  the end of the buffer altering unrelated memory structures.
  -  This vulnerability does not affect TCP/TLS connections.
  -  -- Resolved in 1.6.2.18.1 and 1.8.4.3
  -
  - * AST-2011-009: A remote user sending a SIP packet containing a Contact header
  -  with a missing left angle bracket (<) causes Asterisk to
  -  access a null pointer.
  -  -- Resolved in 1.8.4.3
  -
  - * AST-2011-010: A memory address was inadvertently transmitted over the
  -  network via IAX2 via an option control frame and the remote party would try
  -  to access it.
  -  -- Resolved in 1.4.41.1, 1.6.2.18.1, and 1.8.4.3
  -
  - The issues and resolutions are described in the AST-2011-008, AST-2011-009, and
  - AST-2011-010 security advisories.
  -
  - For more information about the details of these vulnerabilities, please read
  - the security advisories AST-2011-008, AST-2011-009, and AST-2011-010, which were
  - released at the same time as this announcement.
  -
  - For a full list of changes in the current releases, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.41.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.18.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.3
  -
  - Security advisories AST-2011-008, AST-2011-009, and AST-2011-010 are available
  - at:
  -
  - http://downloads.asterisk.org/pub/security/AST-2011-008.pdf
  - http://downloads.asterisk.org/pub/security/AST-2011-009.pdf
  - http://downloads.asterisk.org/pub/security/AST-2011-010.pdf
* Tue Jun 21 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.4.2-2
  - Convert to systemd
* Fri Jun 17 2011 Marcela MaÅ¡láÅová <mmaslano@redhat.com> - 1.8.4.2-1.2
  - Perl mass rebuild
* Fri Jun 10 2011 Marcela MaÅ¡láÅová <mmaslano@redhat.com> - 1.8.4.2-1.1
  - Perl 5.14 mass rebuild
* Fri Jun 03 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.4.2-1:
  -
  - The Asterisk Development Team has announced the release of Asterisk
  - version 1.8.4.2, which is a security release for Asterisk 1.8.
  -
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of Asterisk 1.8.4.2 resolves an issue with SIP URI
  - parsing which can lead to a remotely exploitable crash:
  -
  -    Remote Crash Vulnerability in SIP channel driver (AST-2011-007)
  -
  - The issue and resolution is described in the AST-2011-007 security
  - advisory.
  -
  - For more information about the details of this vulnerability, please
  - read the security advisory AST-2011-007, which was released at the
  - same time as this announcement.
  -
  - For a full list of changes in the current release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.4.2
  -
  - Security advisory AST-2011-007 is available at:
  -
  - http://downloads.asterisk.org/pub/security/AST-2011-007.pdf
  -
  - The Asterisk Development Team has announced the release of Asterisk 1.8.4.1.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 1.8.4.1 resolves several issues reported by the
  - community. Without your help this release would not have been possible.
  - Thank you!
  -
  - Below is a list of issues resolved in this release:
  -
  -  * Fix our compliance with RFC 3261 section 18.2.2. (aka Cisco phone fix)
  -   (Closes issue #18951. Reported by jmls. Patched by wdoekes)
  -
  -  * Resolve a change in IPv6 header parsing due to the Cisco phone fix issue.
  -   This issue was found and reported by the Asterisk test suite.
  -   (Closes issue #18951. Patched by mnicholson)
  -
  -  * Resolve potential crash when using SIP TLS support.
  -   (Closes issue #19192. Reported by stknob. Patched by Chainsaw. Tested by
  -    vois, Chainsaw)
  -
  -  * Improve reliability when using SIP TLS.
  -   (Closes issue #19182. Reported by st. Patched by mnicholson)
  -
  -
  - For a full list of changes in this release candidate, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4.1
  
  - The Asterisk Development Team has announced the release of Asterisk 1.8.4. This
  - release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 1.8.4 resolves several issues reported by the community.
  - Without your help this release would not have been possible. Thank you!
  -
  - Below is a sample of the issues resolved in this release:
  -
  -  * Use SSLv23_client_method instead of old SSLv2 only.
  -   (Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell
  -   and chazzam.
  -
  -  * Resolve crash in ast_mutex_init()
  -   (Patched by twilson)
  -
  -  * Resolution of several DTMF based attended transfer issues.
  -   (Closes issue #17999, #17096, #18395, #17273. Reported by iskatel, gelo,
  -   shihchuan, grecco. Patched by rmudgett)
  -
  -   NOTE: Be sure to read the ChangeLog for more information about these changes.
  -
  -  * Resolve deadlocks related to device states in chan_sip
  -   (Closes issue #18310. Reported, patched by one47. Patched by jpeeler)
  -
  -  * Resolve an issue with the Asterisk manager interface leaking memory when
  -   disabled.
  -   (Reported internally by kmorgan. Patched by russellb)
  -
  -  * Support greetingsfolder as documented in voicemail.conf.sample.
  -   (Closes issue #17870. Reported by edhorton. Patched by seanbright)
  -
  -  * Fix channel redirect out of MeetMe() and other issues with channel softhangup
  -   (Closes issue #18585. Reported by oej. Tested by oej, wedhorn, russellb.
  -   Patched by russellb)
  -
  -  * Fix voicemail sequencing for file based storage.
  -   (Closes issue #18498, #18486. Reported by JJCinAZ, bluefox. Patched by
  -   jpeeler)
  -
  -  * Set hangup cause in local_hangup so the proper return code of 486 instead of
  -   503 when using Local channels when the far sides returns a busy. Also affects
  -   CCSS in Asterisk 1.8+.
  -   (Patched by twilson)
  -
  -  * Fix issues with verbose messages not being output to the console.
  -   (Closes issue #18580. Reported by pabelanger. Patched by qwell)
  -
  -  * Fix Deadlock with attended transfer of SIP call
  -   (Closes issue #18837. Reported, patched by alecdavis. Tested by
  -   alecdavid, Irontec, ZX81, cmaj)
  -
  - Includes changes per AST-2011-005 and AST-2011-006
  - For a full list of changes in this release candidate, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.4
  -
  - Information about the security releases are available at:
  -
  - http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
  - http://downloads.asterisk.org/pub/security/AST-2011-006.pdf
* Thu Apr 21 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3.3-1
  - The Asterisk Development Team has announced security releases for Asterisk
  - branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
  - released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two
  - issues:
  -
  - * File Descriptor Resource Exhaustion (AST-2011-005)
  - * Asterisk Manager User Shell Access (AST-2011-006)
  -
  - The issues and resolutions are described in the AST-2011-005 and AST-2011-006
  - security advisories.
  -
  - For more information about the details of these vulnerabilities, please read the
  - security advisories AST-2011-005 and AST-2011-006, which were released at the
  - same time as this announcement.
  -
  - For a full list of changes in the current releases, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.40.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.25
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.3
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.3
  -
  - Security advisory AST-2011-005 and AST-2011-006 are available at:
  -
  - http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
  - http://downloads.asterisk.org/pub/security/AST-2011-006.pdf
* Wed Mar 23 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3.2-2
  - Bump release and rebuild for mysql 5.5.10 soname change.
* Thu Mar 17 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3.2-1
  - The Asterisk Development Team has announced security releases for Asterisk
  - branches 1.6.1, 1.6.2, and 1.8. The available security releases are
  - released as versions 1.6.1.24, 1.6.2.17.2, and 1.8.3.2.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - ** This is a re-release of Asterisk 1.6.1.23, 1.6.2.17.1 and 1.8.3.1 which
  -   contained a bug which caused duplicate manager entries (issue #18987).
  -
  - The releases of Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 resolve two issues:
  -
  -  * Resource exhaustion in Asterisk Manager Interface (AST-2011-003)
  -  * Remote crash vulnerability in TCP/TLS server (AST-2011-004)
  -
  - The issues and resolutions are described in the AST-2011-003 and AST-2011-004
  - security advisories.
  -
  - For more information about the details of these vulnerabilities, please read the
  - security advisories AST-2011-003 and AST-2011-004, which were released at the
  - same time as this announcement.
  -
  - For a full list of changes in the current releases, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.24
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.2
  -
  - Security advisory AST-2011-003 and AST-2011-004 are available at:
  -
  - http://downloads.asterisk.org/pub/security/AST-2011-003.pdf
  - http://downloads.asterisk.org/pub/security/AST-2011-004.pdf
* Thu Mar 17 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3.1-1
  - The Asterisk Development Team has announced security releases for Asterisk
  - branches 1.6.1, 1.6.2, and 1.8. The available security releases are
  - released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The releases of Asterisk 1.6.1.23, 1.6.2.17.1, and 1.8.3.1 resolve two issues:
  -
  -  * Resource exhaustion in Asterisk Manager Interface (AST-2011-003)
  -  * Remote crash vulnerability in TCP/TLS server (AST-2011-004)
  -
  - The issues and resolutions are described in the AST-2011-003 and AST-2011-004
  - security advisories.
  -
  - For more information about the details of these vulnerabilities, please read the
  - security advisories AST-2011-003 and AST-2011-004, which were released at the
  - same time as this announcement.
  -
  - For a full list of changes in the current releases, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.23
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.3.1
  -
  - Security advisory AST-2011-003 and AST-2011-004 are available at:
  -
  - http://downloads.asterisk.org/pub/security/AST-2011-003.pdf
  - http://downloads.asterisk.org/pub/security/AST-2011-004.pdf
* Mon Feb 28 2011 <jeff@ocjtech.us> - 1.8.3-1
  - The Asterisk Development Team has announced the release of Asterisk 1.8.3. This
  - release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 1.8.3 resolves several issues reported by the community
  - and would have not been possible without your participation. Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * Resolve duplicated data in the AstDB when using DIALGROUP()
  -  (Closes issue #18091. Reported by bunny. Patched by tilghman)
  -
  - * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
  -  (Closes issue #18464. Reported, patched by IgorG)
  -
  - * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of
  -  unit tests for the function that does the parsing.
  -  (Closes issue #18350. Reported by gbour. Patched by Marquis)
  -
  - * When using cdr_pgsql the billsec field was not populated correctly on
  -  unanswered calls.
  -  (Closes issue #18406. Reported by joscas. Patched by tilghman)
  -
  - * Resolve memory leak in iCalendar and Exchange calendaring modules.
  -  (Closes issue #18521. Reported, patched by pitel. Tested by cervajs)
  -
  - * This version of Asterisk includes the new Compiler Flags option
  -  BETTER_BACKTRACES which uses libbfd to search for better symbol information
  -  within both the Asterisk binary, as well as loaded modules, to assist when
  -  using inline backtraces to track down problems.
  -  (Patched by tilghman)
  -
  - * Resolve issue where no Music On Hold may be triggered when using
  -  res_timing_dahdi.
  -  (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested
  -  by francesco_r, rfrantik, one47)
  -
  - * Resolve a memory leak when the Asterisk Manager Interface is disabled.
  -  (Reported internally by kmorgan. Patched by russellb)
  -
  - * Reimplemented fax session reservation to reverse the ABI breakage introduced
  -  in r297486.
  -  (Reported internally. Patched by mnicholson)
  -
  - * Fix regression that changed behavior of queues when ringing a queue member.
  -  (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)
  -
  - * Resolve deadlock involving REFER.
  -  (Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.)
  -
  - Additionally, this release has the changes related to security bulletin
  - AST-2011-002 which can be found at
  - http://downloads.asterisk.org/pub/security/AST-2011-002.pdf
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3
* Wed Feb 16 2011 <jeff@ocjtech.us> - 1.8.3-0.7.rc3
  -
  - The Asterisk Development Team has announced the third release candidate of
  - Asterisk 1.8.3. This release candidate is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 1.8.3-rc3 resolves the following issues in addition to
  - those included in 1.8.3-rc1 and 1.8.3-rc2:
  -
  - *  Fix regression that changed behavior of queues when ringing a queue member.
  -   (Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)
  -
  - * Resolve deadlock involving REFER.
  -  (Closes issue #18403. Reported, tested by jthurman. Patched by jpeeler.)
  -
  - For a full list of changes in this release candidate, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc3
* Fri Feb 11 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3-0.6.rc2
  - Bump release to build for F15
* Wed Feb 09 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3-0.5.rc2
  - Remove isa macros
* Wed Feb 09 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3-0.4.rc2
  - Make library dependencies architecture specific
* Mon Feb 07 2011 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 1.8.3-0.3.rc2
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild
* Wed Jan 26 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3-0.2.rc2
  The Asterisk Development Team has announced the second release candidate of
  Asterisk 1.8.3. This release candidate is available for immediate download at
  http://downloads.asterisk.org/pub/telephony/asterisk/
  
  The release of Asterisk 1.8.3-rc2 resolves the following issues in addition to
  those included in 1.8.3-rc1:
  
   * Resolve issue where no Music On Hold may be triggered when using
    res_timing_dahdi.
    (Closes issues #18262. Reported by francesco_r. Patched by cjacobson. Tested
     by francesco_r, rfrantik, one47)
  
   * Resolve a memory leak when the Asterisk Manager Interface is disabled.
    (Reported internally by kmorgan. Patched by russellb)
  
   * Reimplemented fax session reservation to reverse the ABI breakage introduced
    in r297486.
    (Reported internally. Patched by mnicholson)
  
  For a full list of changes in this release candidate, please see the ChangeLog:
  
  http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.3-rc2
* Wed Jan 26 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3-0.1.rc1
  -
  - The Asterisk Development Team has announced the first release candidate of
  - Asterisk 1.8.3. This release candidate is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 1.8.3-rc1 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release candidate:
  -
  -  * Resolve duplicated data in the AstDB when using DIALGROUP()
  -   (Closes issue #18091. Reported by bunny. Patched by tilghman)
  -
  -  * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses.
  -   (Closes issue #18464. Reported, patched by IgorG)
  -
  -  * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of
  -   unit tests for the function that does the parsing.
  -   (Closes issue #18350. Reported by gbour. Patched by Marquis)
  -
  -  * When using cdr_pgsql the billsec field was not populated correctly on
  -   unanswered calls.
  -   (Closes issue #18406. Reported by joscas. Patched by tilghman)
  -
  -  * Resolve memory leak in iCalendar and Exchange calendaring modules.
  -   (Closes issue #18521. Reported, patched by pitel. Tested by cervajs)
  -
  -  * This version of Asterisk includes the new Compiler Flags option
  -   BETTER_BACKTRACES which uses libbfd to search for better symbol information
  -   within both the Asterisk binary, as well as loaded modules, to assist when
  -   using inline backtraces to track down problems.
  -   (Patched by tilghman)
* Wed Jan 26 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.2.3-1
  -
  - The Asterisk Development Team has announced the release of Asterisk 1.8.2.3.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 1.8.2.3 resolves the following issue:
  -
  -  * Reimplemented fax session reservation to reverse the ABI breakage introduced
  -   in r297486.
  -   (Reported by Jeremy Kister on the asterisk-users mailing list. Patched by
  -   mnicholson)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2.3
* Mon Jan 24 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.2.2-2
  - Build with SRTP support
* Mon Jan 24 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.2.2-1
  -
  - The Asterisk Development Team has announced a release for the security issue
  - described in AST-2011-001.
  -
  - Due to a failed merge, Asterisk 1.8.2.1 which should have included the security
  - fix did not. Asterisk 1.8.2.2 contains the the changes which should have been
  - included in Asterisk 1.8.2.1.
  -
  - This releases is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2,
  - 1.8.1.2, and 1.8.2.2 resolve an issue when forming an outgoing SIP request while
  - in pedantic mode, which can cause a stack buffer to be made to overflow if
  - supplied with carefully crafted caller ID information. The issue and resolution
  - are described in the AST-2011-001 security advisory.
  -
  - For more information about the details of this vulnerability, please read the
  - security advisory AST-2011-001, which was released at the same time as this
  - announcement.
  -
  - For a full list of changes in the current release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.2
  -
  - Security advisory AST-2011-001 is available at:
  -
  - http://downloads.asterisk.org/pub/security/AST-2011-001.pdf
* Mon Jan 24 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.2.1-1
  -
  - The Asterisk Development Team has announced security releases for the following
  - versions of Asterisk:
  -
  - * 1.4.38.1
  - * 1.4.39.1
  - * 1.6.1.21
  - * 1.6.2.15.1
  - * 1.6.2.16.1
  - * 1.8.1.2
  - * 1.8.2.1
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The releases of Asterisk 1.4.38.1, 1.4.39.1, 1.6.1.21, 1.6.2.15.1, 1.6.2.16.2,
  - 1.8.1.2, and 1.8.2.1 resolve an issue when forming an outgoing SIP request while
  - in pedantic mode, which can cause a stack buffer to be made to overflow if
  - supplied with carefully crafted caller ID information. The issue and resolution
  - are described in the AST-2011-001 security advisory.
  -
  - For more information about the details of this vulnerability, please read the
  - security advisory AST-2011-001, which was released at the same time as this
  - announcement.
  -
  - For a full list of changes in the current releases, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.38.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.39.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.21
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.15.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.16.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.1.2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.1
  -
  - Security advisory AST-2011-001 is available at:
  -
  - http://downloads.asterisk.org/pub/security/AST-2011-001.pdf
* Mon Jan 24 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.2-1
  -
  - The Asterisk Development Team has announced the release of Asterisk 1.8.2. This
  - release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 1.8.2 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * 'sip notify clear-mwi' needs terminating CRLF.
  -  (Closes issue #18275. Reported, patched by klaus3000)
  -
  - * Patch for deadlock from ordering issue between channel/queue locks in
  -  app_queue (set_queue_variables).
  -  (Closes issue #18031. Reported by rain. Patched by bbryant)
  -
  - * Fix cache of device state changes for multiple servers.
  -  (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested
  -  by russellb)
  -
  - * Resolve issue where channel redirect function (CLI or AMI) hangs up the call
  -  instead of redirecting the call.
  -  (Closes issue #18171. Reported by: SantaFox)
  -  (Closes issue #18185. Reported by: kwemheuer)
  -  (Closes issue #18211. Reported by: zahir_koradia)
  -  (Closes issue #18230. Reported by: vmarrone)
  -  (Closes issue #18299. Reported by: mbrevda)
  -  (Closes issue #18322. Reported by: nerbos)
  -
  - * Fix reloading of peer when a user is requested. Prevent peer reloading from
  -  causing multiple MWI subscriptions to be created when using realtime.
  -  (Closes issue #18342. Reported, patched by nivek.)
  -
  - * Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0
  -  so res_jabber doesn't think there is already an XMPP connection sending
  -  device state. Also clean up CLI commands a bit.
  -  (Closes issue #18272. Reported by klaus3000. Patched by Marquis42)
  -
  - * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of
  -  setting peer->cdr = NULL, set it to not post.
  -  (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)
  -
  - * Fixes issue with outbound google voice calls not working. Thanks to az1234
  -  and nevermind_quack for their input in helping debug the issue.
  -  (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2
* Mon Jan 24 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.1.1-1
  -
  - The Asterisk Development Team has announced the release of Asterisk 1.8.1.1.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 1.8.1.1 resolves two issues reported by the community
  - since the release of Asterisk 1.8.1.
  -
  -  * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of
  -   setting peer->cdr = NULL, set it to not post.
  -   (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)
  -
  -  * Fixes issue with outbound google voice calls not working. Thanks to az1234
  -   and nevermind_quack for their input in helping debug the issue.
  -   (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)
  -
  - For a full list of changes in this release candidate, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1.1
* Mon Jan 24 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.1-1
  -
  - The Asterisk Development Team has announced the release of Asterisk 1.8.1. This
  - release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - The release of Asterisk 1.8.1 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is a sample of the issues resolved in this release:
  -
  - * Fix issue when using directmedia. Asterisk needs to limit the codecs offered
  -   to just the ones that both sides recognize, otherwise they may end up sending
  -   audio that the other side doesn't understand.
  -   (Closes issue #17403. Reported, patched by one47. Tested by one47, falves11)
  -
  - * Resolve issue where Party A in an analog 3-way call would continue to hear
  -   ringback after party C answers.
  -   (Patched by rmudgett)
  -
  - * Fix playback failure when using IAX with the timerfd module.
  -   (Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler)
  -
  - * Fix problem with qualify option packets for realtime peers never stopping.
  -   The option packets not only never stopped, but if a realtime peer was not in
  -   the peer list multiple options dialogs could accumulate over time.
  -   (Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
  -   jpeeler)
  -
  - * Fix issue where it is possible to crash Asterisk by feeding the curl engine
  -   invalid data.
  -   (Closes issue #18161. Reported by wdoekes. Patched by tilghman)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
* Tue Jan 18 2011 Dennis Gilmore <dennis@ausil.us> - 1.8.0-6
  - dont package up the ices bits on el the client doesnt exist for us
* Tue Jan 18 2011 Dennis Gilmore <dennis@ausil.us> - 1.8.0-5
  - dont build the 389 directory server package its not available on rhel6
* Fri Dec 10 2010 Dennis Gilmore <dennis@ausil.us> - 1.8.0-4
  - dont always build AIS modules we dont have the BuildRequires on epel
* Fri Oct 29 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-3
  - Rebuild for new net-snmp.
* Tue Oct 26 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-2
  - Always build AIS modules
* Thu Oct 21 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-1
  - The Asterisk Development Team is proud to announce the release of Asterisk
  - 1.8.0. This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/
  -
  - Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
  - Term Support (LTS) release, similar to Asterisk 1.4. For more information about
  - support time lines for Asterisk releases, see the Asterisk versions page.
  -
  - http://www.asterisk.org/asterisk-versions
  -
  - The release of Asterisk 1.8.0 would not have been possible without the support
  - and contributions of the community. Since Asterisk 1.6.2, we've had over 500
  - reporters, more than 300 testers and greater than 200 developers contributed to
  - this release.
  -
  - You can find a summary of the work involved with the 1.8.0 release in the
  - sumary:
  -
  - http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt
  -
  - A short list of available features includes:
  -
  -     * Secure RTP
  -     * IPv6 Support in the SIP channel driver
  -     * Connected Party Identification Support
  -     * Calendaring Integration
  -     * A new call logging system, Channel Event Logging (CEL)
  -     * Distributed Device State using Jabber/XMPP PubSub
  -     * Call Completion Supplementary Services support
  -     * Advice of Charge support
  -     * Much, much more!
  -
  - A full list of new features can be found in the CHANGES file.
  -
  - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
  -
  - For a full list of changes in the current release candidate, please see the
  - ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0
  -
  - Thank you for your continued support of Asterisk!
* Mon Oct 18 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-0.8.rc5:
  -
  - The release of Asterisk 1.8.0-rc5 was triggered by some last minute platform
  - compatibility IPv6 changes. In addition, the availability of the English sound
  - prompts with Australian accents has been added.
  -
  - A full list of new features can be found in the CHANGES file.
  -
  - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
  -
  - For a full list of changes in the current release candidate, please see the
  - ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc5
  -
  - This release candidate contains fixes since the last release candidate as
  - reported by the community. A sampling of the changes in this release candidate
  - include:
  -
  -  * Additional fixups in chan_gtalk that allow outbound calls to both Google
  -    Talk and Google Voice recipients. Adds new chan_gtalk enhancements externip
  -    and stunaddr.
  -    (Closes issue #13971. Patched by dvossel)
  -
  -  * Resolve manager crash issue.
  -    (Closes issue #17994. Reported by vrban. Patchd by dvossel)
  -
  -  * Documentation updates for sample configuration files.
  -    (Closes issues #18107, #18101. Reported, patched by lathama, lmadsen)
  -
  -  * Resolve issue where faxdetect would only detect the first fax call in
  -    chan_dahdi.
  -    (Closes issue #18116. Reported by seandarcy. Patched by rmudgett)
  -
  -  * Resolve issue where a channel that is setup and torn down *very* quickly may
  -    not have the right call disposition or ${DIALSTATUS}.
  -    (Closes issue #16946. Reported by davidw. Review
  -     https://reviewboard.asterisk.org/r/740/)
  -
  -  * Set TCLASS field of IPv6 header when SIP QoS options are set.
  -    (Closes issue #18099. Reported by jamesnet. Patched by dvossel)
  -
  -  * Resolve issue where Asterisk could crash on shutdown when using SRTP.
  -    (Closes issue #18085. Reported by st. Patched by twilson)
  -
  -  * Fix issue where peers host port would be lost on a SIP reload.
  -    (Closes issue #18135. Reported, tested by lmadsen. Patched by dvossel)
  -
  - A short list of available features includes:
  -
  -   * Secure RTP
  -   * IPv6 Support in the SIP channel driver
  -   * Connected Party Identification Support
  -   * Calendaring Integration
  -   * A new call logging system, Channel Event Logging (CEL)
  -   * Distributed Device State using Jabber/XMPP PubSub
  -   * Call Completion Supplementary Services support
  -   * Advice of Charge support
  -   * Much, much more!
  -
  - A full list of new features can be found in the CHANGES file.
  -
  - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
  -
  - For a full list of changes in the current release candidate, please see the
  - ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc4
* Fri Oct 08 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-0.7.rc3
  - This release candidate contains fixes since the release candidate as reported by
  - the community. A sampling of the changes in this release candidate include:
  -
  -  * Still build chan_sip even if res_crypto cannot be built (use, but not depend)
  -    (Reported by a user on the mailing list. Patched by tilghman)
  -
  -  * Get notifications for call files only when a file is closed, not when created
  -    (Closes issue #17924. Reported by mkeuter. Patched by abeldeck)
  -
  -  * Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk
  -    expects the DTMF to arrive on the RTP stream and not via jingle DTMF
  -    signalling.
  -    (Patched by dvossel. Tested by malcolmd)
  -
  -  * Fixes to allow chan_gtalk to communicate with the Gmail web client.
  -    (Patched by phsultan and dvossel)
  -
  -  * Fix to GET DATA to allow audio to be streamed via an AGI.
  -    (Closes issue #18001. Reported by jamicque. Patched by tilghman)
  -
  -  * Resolve dnsmgr memory corruption in chan_iax2.
  -    (Closes issue #17902. Reported by afried. Patched by russell, dvossel)
  -
  - A short list of available features includes:
  -
  -  * Secure RTP
  -  * IPv6 Support in the SIP channel driver
  -  * Connected Party Identification Support
  -  * Calendaring Integration
  -  * A new call logging system, Channel Event Logging (CEL)
  -  * Distributed Device State using Jabber/XMPP PubSub
  -  * Call Completion Supplementary Services support
  -  * Advice of Charge support
  -  * Much, much more!
  -
  - A full list of new features can be found in the CHANGES file.
  -
  - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
  -
  - For a full list of changes in the current release candidate, please see the
  - ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc3
* Wed Oct 06 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-0.6.rc2
  - This release candidate contains fixes since the last beta release as reported by
  - the community. A sampling of the changes in this release candidate include:
  -
  -  * Add slin16 support for format_wav (new wav16 file extension)
  -    (Closes issue #15029. Reported, patched by andrew. Tested by Qwell)
  -
  -  * Fixes a bug in manager.c where the default configuration values weren't reset
  -    when the manager configuration was reloaded.
  -    (Closes issue #17917. Reported by lmadsen. Patched by bbryant)
  -
  -  * Various fixes for the calendar modules.
  -    (Patched by Jan Kalab.
  -     Reviewboard: https://reviewboard.asterisk.org/r/880/
  -     Closes issue #17877. Review: https://reviewboard.asterisk.org/r/916/
  -     Closes issue #17776. Review: https://reviewboard.asterisk.org/r/921/)
  -
  -  * Add CHANNEL(checkhangup) to check whether a channel is in the process of
  -    being hung up.
  -    (Closes issue #17652. Reported, patched by kobaz)
  -
  -  * Fix a bug with MeetMe where after announcing the amount of time left in a
  -    conference, if music on hold was playing, it doesn't restart.
  -    (Closes issue #17408, Reported, patched by sysreq)
  -
  -  * Fix interoperability problems with session timer behavior in Asterisk.
  -    (Closes issue #17005. Reported by alexcarey. Patched by dvossel)
  -
  -  * Rate limit calls to fsync() to 1 per second after astdb updates. Astdb was
  -    determined to be one of the most significant bottlenecks in SIP registration
  -    processing. This patch improved the speed of an astdb load test by 50000%
  -    (yes, Fifty-Thousand Percent). On this particular load test setup, this
  -    doubled the number of SIP registrations the server could handle.
  -    (Review: https://reviewboard.asterisk.org/r/825/)
  -
  -  * Don't clear the username from a realtime database when a registration
  -    expires. Non-realtime chan_sip does not clear the username from memory when a
  -    registration expiries so realtime probably shouldn't either.
  -    (Closes issue #17551. Reported, patched by: ricardolandim. Patched by
  -     mnicholson)
  -
  -  * Don't hang up a call on an SRTP unprotect failure. Also make it more obvious
  -    when there is an issue en/decrypting.
  -    (Closes issue #17563. Reported by Alexcr. Patched by sfritsch. Tested by
  -     twilson)
  -
  -  * Many more issues. This is a significant upgrade over Asterisk 1.8.0 beta 5!
  -
  - A short list of available features includes:
  -
  -  * Secure RTP
  -  * IPv6 Support in the SIP channel driver
  -  * Connected Party Identification Support
  -  * Calendaring Integration
  -  * A new call logging system, Channel Event Logging (CEL)
  -  * Distributed Device State using Jabber/XMPP PubSub
  -  * Call Completion Supplementary Services support
  -  * Advice of Charge support
  -  * Much, much more!
  -
  - A full list of new features can be found in the CHANGES file.
  -
  - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
  -
  - For a full list of changes in the current release candidate, please see the
  - ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-rc2
* Thu Sep 09 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-0.5.beta5
  - This release contains fixes since the last beta release as reported by the
  - community. A sampling of the changes in this release include:
  -
  -  * Fix issue where TOS is no longer set on RTP packets.
  -    (Closes issue #17890. Reported, patched by elguero)
  -
  -  * Change pedantic default value in chan_sip from 'no' to 'yes'
  -
  -  * Asterisk now dynamically builds the "Supported" header depending on what is
  -    enabled/disabled in sip.conf. Session timers used to always be advertised as
  -    being supported even when they were disabled in the configuration.
  -    (Related to issue #17005. Patched by dvossel)
  -
  -  * Convert MOH to use generic timers.
  -    (Closes issue #17726. Reported by lmadsen. Patched by tilghman)
  -
  -  * Fix SRTP for changing SSRC and multiple a=crypto SDP lines. Adding code to
  -    Asterisk that changed the SSRC during bridges and masquerades broke SRTP
  -    functionality. Also broken was handling the situation where an incoming
  -    INVITE had more than one crypto offer.
  -    (Closes issue #17563. Reported by Alexcr. Patched by twilson)
  -
  - Asterisk 1.8 contains many new features over previous releases of Asterisk.
  - A short list of included features includes:
  -
  -     * Secure RTP
  -     * IPv6 Support in the SIP Channel
  -     * Connected Party Identification Support
  -     * Calendaring Integration
  -     * A new call logging system, Channel Event Logging (CEL)
  -     * Distributed Device State using Jabber/XMPP PubSub
  -     * Call Completion Supplementary Services support
  -     * Advice of Charge support
  -     * Much, much more!
  -
  - A full list of new features can be found in the CHANGES file.
  -
  - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
  -
  - For a full list of changes in the current release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta5
* Tue Aug 24 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-0.4.beta4
  - This release contains fixes since the last beta release as reported by the
  - community. A sampling of the changes in this release include:
  -
  -  * Fix parsing of IPv6 address literals in outboundproxy
  -    (Closes issue #17757. Reported by oej. Patched by sperreault)
  -
  -  * Change the default value for alwaysauthreject in sip.conf to "yes".
  -    (Closes issue #17756. Reported by oej)
  -
  -  * Remove current STUN support from chan_sip.c. This change removes the current
  -    broken/useless STUN support from chan_sip.
  -    (Closes issue #17622. Reported by philipp2.
  -     Review: https://reviewboard.asterisk.org/r/855/)
  -
  -  * PRI CCSS may use a stale dial string for the recall dial string. If an
  -    outgoing call negotiates a different B channel than initially requested, the
  -    saved original dial string was not transferred to the new B channel. CCSS
  -    uses that dial string to generate the recall dial string.
  -    (Patched by rmudgett)
  -
  -  * Split _all_ arguments before parsing them. This fixes multicast RTP paging
  -    using linksys mode.
  -    (Patched by russellb)
  -
  -  * Expand cel_custom.conf.sample. Include the usage of CSV_QUOTE() to ensure
  -    data has valid CSV formatting. Also list the special CEL variables that are
  -    available for use in the mapping. There are also several other CEL fixes in
  -    this release.
  -    (Patched by russellb)
  -
  - Asterisk 1.8 contains many new features over previous releases of Asterisk.
  - A short list of included features includes:
  -
  -     * Secure RTP
  -     * IPv6 Support in the SIP Channel
  -     * Connected Party Identification Support
  -     * Calendaring Integration
  -     * A new call logging system, Channel Event Logging (CEL)
  -     * Distributed Device State using Jabber/XMPP PubSub
  -     * Call Completion Supplementary Services support
  -     * Advice of Charge support
  -     * Much, much more!
  -
  - A full list of new features can be found in the CHANGES file.
  -
  - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
  -
  - For a full list of changes in the current release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta4
* Wed Aug 11 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-0.3.beta3
  -
  - This release contains fixes since the last beta release as reported by the
  - community. A sampling of the changes in this release include:
  -
  -  * Fix a regression where HTTP would always be enabled regardless of setting.
  -    (Closes issue #17708. Reported, patched by pabelanger)
  -
  -  * ACL errors displayed on screen when using dynamic_exclude_static in sip.conf
  -    (Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson)
  -
  -  * Support "channels" in addition to "channel" in chan_dahdi.conf.
  -    (https://reviewboard.asterisk.org/r/804)
  -
  -  * Fix parsing error in sip_sipredirect(). The code was written in a way that
  -    did a bad job of parsing the port out of a URI. Specifically, it would do
  -    badly when dealing with an IPv6 address.
  -    (Closes issue #17661. Reported by oej. Patched by mmichelson)
  -
  -  * Fix inband DTMF detection on outgoing ISDN calls.
  -    (Patched by russellb and rmudgett)
  -
  -  * Fixes issue with translator frame not getting freed. This issue prevented
  -    g729 licenses from being freed up.
  -    (Closes issue #17630. Reported by manvirr. Patched by dvossel)
  -
  -  * Fixed IPv6-related SIP parsing bugs and updated documention.
  -    (Reported by oej. Patched by sperreault)
  -
  -  * Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a
  -    list of a specified item. Matches up with FIELDQTY() and CUT().
  -    (Closes #17713. Reported, patched by gareth. Tested by tilghman)
  -
  - Asterisk 1.8 contains many new features over previous releases of Asterisk.
  - A short list of included features includes:
  -
  -     * Secure RTP
  -     * IPv6 Support
  -     * Connected Party Identification Support
  -     * Calendaring Integration
  -     * A new call logging system, Channel Event Logging (CEL)
  -     * Distributed Device State using Jabber/XMPP PubSub
  -     * Call Completion Supplementary Services support
  -     * Advice of Charge support
  -     * Much, much more!
  -
  - A full list of new features can be found in the CHANGES file.
  -
  - http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=checkout
  -
  - For a full list of changes in the current release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta3
* Mon Aug 02 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-0.2.beta2
  - Rebuild against libpri 1.4.12
* Mon Aug 02 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-0.1.beta2
  - Update to 1.8.0-beta2
  - Disable building chan_misdn until compilation errors are figured out (https://issues.asterisk.org/view.php?id=14333)
  - Start stripping tarballs again because Digium added MP3 code :(
* Sat Jul 31 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.10-1
  -
  - The following are a few of the issues resolved by community developers:
  -
  -  * Allow users to specify a port for DUNDI peers.
  -    (Closes issue #17056. Reported, patched by klaus3000)
  -
  -  * Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is
  -    set.
  -    (Closes issue #16815. Reported, patched by rain)
  -
  -  * If there is realtime configuration, it does not get re-read on reload unless
  -    the config file also changes.
  -    (Closes issue #16982. Reported, patched by dmitri)
  -
  -  * Send AgentComplete manager event for attended transfers.
  -    (Closes issue #16819. Reported, patched by elbriga)
  -
  -  * Correct manager variable 'EventList' case.
  -    (Closes issue #17520. Reported, patched by kobaz)
  -
  - In addition, changes to res_timing_pthread that should make it more stable have
  - also been implemented.
  -
  - For a full list of changes in the current release, please see the
  - ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.10
* Wed Jul 14 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.8-0.3.rc1
  - Add patch to remove requirement on latex2html
* Tue Jun 01 2010 Marcela Maslanova <mmaslano@redhat.com> - 1.6.2.8-0.2.rc1
  - Mass rebuild with perl-5.12.0
* Tue May 04 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.7-1
  -  * Fix building CDR and CEL SQLite3 modules.
  -    (Closes issue #17017. Reported by alephlg. Patched by seanbright)
  -
  -  * Resolve crash in SLAtrunk when the specified trunk doesn't exist.
  -    (Reported in #asterisk-dev by philipp64. Patched by seanbright)
  -
  -  * Include an extra newline after "Aliased CLI command" to get back the prompt.
  -    (Issue #16978. Reported by jw-asterisk. Tested, patched by seanbright)
  -
  -  * Prevent segfault if bad magic number is encountered.
  -    (Issue #17037. Reported, patched by alecdavis)
  -
  -  * Update code to reflect that handle_speechset has 4 arguments.
  -    (Closes issue #17093. Reported, patched by gpatri. Tested by pabelanger,
  -     mmichelson)
  -
  -  * Resolve a deadlock in chan_local.
  -    (Closes issue #16840. Reported, patched by bzing2, russell. Tested by bzing2)
* Mon May 03 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.7-0.2.rc3
  - Update to 1.6.2.7-rc3
* Thu Apr 15 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.7-0.1.rc2
  - Update to 1.6.2.7-rc2
* Fri Mar 12 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.6-1
  - Update to final 1.6.2.6
  -
  - The following are a few of the issues resolved by community developers:
  -
  -  * Make sure to clear red alarm after polarity reversal.
  -    (Closes issue #14163. Reported, patched by jedi98. Tested by mattbrown,
  -     Chainsaw, mikeeccleston)
  -
  -  * Fix problem with duplicate TXREQ packets in chan_iax2
  -    (Closes issue #16904. Reported, patched by rain. Tested by rain, dvossel)
  -
  -  * Fix crash in app_voicemail related to message counting.
  -    (Closes issue #16921. Reported, tested by whardier. Patched by seanbright)
  -
  -  * Overlap receiving: Automatically send CALL PROCEEDING when dialplan starts
  -    (Reported, Patched, and Tested by alecdavis)
  -
  -  * For T.38 reINVITEs treat a 606 the same as a 488.
  -    (Closes issue #16792. Reported, patched by vrban)
  -
  -  * Fix ConfBridge crash when no timing module is loaded.
  -    (Closes issue #16471. Reported, tested by kjotte. Patched, tested by junky)
  -
  - For a full list of changes in this releases, please see the ChangeLog:
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.6
* Mon Mar 08 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.6-0.1.rc2
  - Update to 1.6.2.6-rc2
* Mon Mar 08 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.5-2
  - Add a patch that fixes CLI history when linking against external libedit.
* Thu Feb 25 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.5-1
  - Update to 1.6.2.5
  -
  -         * AST-2010-002: Invalid parsing of ACL rules can compromise security
* Thu Feb 18 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.4-1
  - Update to 1.6.2.4
  -
  -        * AST-2010-002: This security release is intended to raise awareness
  -          of how it is possible to insert malicious strings into dialplans,
  -          and to advise developers to read the best practices documents so
  -          that they may easily avoid these dangers.
* Wed Feb 03 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.2-1
  - Update to 1.6.2.2
  -
  -	* AST-2010-001: An attacker attempting to negotiate T.38 over SIP can
  -	  remotely crash Asterisk by modifying the FaxMaxDatagram field of
  -	  the SDP to contain either a negative or exceptionally large value.
  -	  The same crash occurs when the FaxMaxDatagram field is omitted from
  -	  the SDP as well.
* Fri Jan 15 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.1-1
  - Update to 1.6.2.1 final:
  -
  - * CLI 'queue show' formatting fix.
  -   (Closes issue #16078. Reported by RoadKill. Tested by dvossel. Patched by
  -    ppyy.)
  -
  - * Fix misreverting from 177158.
  -   (Closes issue #15725. Reported, Tested by shanermn. Patched by dimas.)
  -
  - * Fixes subscriptions being lost after 'module reload'.
  -   (Closes issue #16093. Reported by jlaroff. Patched by dvossel.)
  -
  - * app_queue segfaults if realtime field uniqueid is NULL
  -  (Closes issue #16385. Reported, Tested, Patched by haakon.)
  -
  - * Fix to Monitor which previously assumed the file to write to did not contain
  -   pathing.
  -   (Closes issue #16377, #16376. Reported by bcnit. Patched by dant.
* Tue Jan 12 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.1-0.1.rc1
  - Update to 1.6.2.1-rc1
* Sat Dec 19 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-1
  - Released version of 1.6.2.0
* Wed Dec 09 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.16.rc8
  - Update to 1.6.2.0-rc8
* Wed Dec 02 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.15.rc7
  - Update to 1.6.2.0-rc7
* Tue Dec 01 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.14.rc6
  - Change the logrotate and the init scripts so that Asterisk doesn't
    try and write to / or /root
* Thu Nov 19 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.13.rc6
  - Make dependency on uw-imap conditional and some other changes to
    make building on RHEL5 easier.
* Fri Nov 13 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.12.rc6
  - Update to 1.6.2.0-rc6
* Mon Nov 09 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.11.rc5
  - Update to 1.6.2.0-rc5
* Fri Nov 06 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.10.rc4
  - Update to 1.6.2.0-rc4
* Tue Oct 27 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.9.rc3
  - Add patch from upstream to fix how res_http_post forms paths.
* Sat Oct 24 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.8.rc3
  - Add an AST_EXTRA_ARGS option to the init script
  - have the init script to cd to /var/spool/asterisk to prevent annoying message
* Sat Oct 24 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.7.rc3
  - Compile against gmime 2.2 instead of gmime 2.4 because the patch to convert the API calls from 2.2 to 2.4 caused crashes.
* Fri Oct 09 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.6.rc3
  - Require latex2html used in static-http documents
* Wed Oct 07 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.5.rc3
  - Change ownership and permissions on config files to protect them.
* Tue Oct 06 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.4.rc3
  - Update to 1.6.2.0-rc3
* Wed Sep 30 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.3.rc2
  - Merge firmware subpackage back into the main package.
* Wed Sep 30 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.2.rc2
  - Package internal help.
  - Fix up some more paths in the configs so that everything ends up where we want them.
* Wed Sep 30 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.1.rc2
  - Update to 1.6.2.0-rc2
  - We no longer need to strip the tarball as it no longer includes non-free items.
* Wed Sep 09 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1.6-2
  - Enable building of API docs.
  - Depend on version 1.2 or newer of speex
* Sun Sep 06 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1.6-1
  - Update to 1.6.1.6
  - Drop patches that are too troublesome to maintain anymore or have been integrated upstream.
* Tue Sep 01 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.26.rc1
  - Add a patch from Quentin Armitage and rebuld.
* Fri Aug 21 2009 Tomas Mraz <tmraz@redhat.com> - 1.6.1-0.25.rc1
  - rebuilt with new openssl
* Fri Jul 24 2009 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 1.6.1-0.24.rc1
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild
* Thu Mar 05 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.23.rc1
  - Rebuild to pick up new AIS and ODBC deps.
  - Update script that strips out bad content from tarball to do the
    download and to check the GPG signature.
* Mon Feb 23 2009 Fedora Release Engineering <rel-eng@lists.fedoraproject.org> - 1.6.1-0.22.rc1
  - Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild
* Sun Feb 08 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.21.rc1
  - Update to 1.6.1-rc1
  - Add backport of conference bridging that is slated for 1.6.2
  - Add patches to conference bridging that implement CLI apps
* Thu Jan 15 2009 Tomas Mraz <tmraz@redhat.com> - 1.6.1-0.13.beta4
  - rebuild with new openssl
* Sun Jan 04 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.12.beta4
  - Fedora Directory Server compatibility patch/subpackage.
* Sun Jan 04 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.10.beta4
  - Fix up paths. BZ#477238
* Sat Jan 03 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.9.beta4
  - Update patches
* Sat Jan 03 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.8.beta4
  - Update to 1.6.1-beta4
* Tue Dec 09 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.7.beta3
  - Update to 1.6.1-beta3
* Tue Dec 09 2008 Alex Lancaster <alexlan[AT]fedoraproject org> - 1.6.1-0.6.beta2
  - Rebuild for new gmime
* Fri Nov 07 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.5.beta2
  - Add patch to fix missing variable on PPC.
* Fri Nov 07 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.4.beta2
  - Update PPC systems don't have sys/io.h patch.
* Fri Nov 07 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.3.beta2
  - PPC systems don't have sys/io.h
* Fri Nov 07 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.2.beta2
  - Update to 1.6.1 beta 2
* Wed Nov 05 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0.1-3
  - Fix issue with init script giving wrong path to config file.
* Thu Oct 16 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0.1-2
  - Explicitly require dahdi-tools-libs to see if we can get this to build.
* Fri Oct 10 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-1
  - Update to final release.
* Thu Sep 11 2008 - Bastien Nocera <bnocera@redhat.com> - 1.6.0-0.22.beta9
  - Rebuild
* Wed Jul 30 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.21.beta9
  - Replace app_rxfax/app_txfax with app_fax taken from upstream SVN.
* Tue Jul 29 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.20.beta9
  - Bump release and rebuild with new libpri and zaptel.
* Fri Jul 25 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.19.beta9
  - Add patch pulled from upstream SVN that fixes AST-2008-010 and AST-2008-011.
* Fri Jul 25 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.18.beta9
  - Add patch for LDAP extracted from upstream SVN (#442011)
* Wed Jul 02 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.17.beta9
  - Add patch that unbreaks cdr_tds with FreeTDS 0.82.
  - Properly obsolete conference subpackage.
* Thu Jun 12 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.16.beta9
  - Disable building cdr_tds since new FreeTDS in rawhide no longer provides needed library.
* Wed Jun 11 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.15.beta9
  - Bump release and rebuild to fix libtds breakage.
* Mon May 19 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.14.beta9
  - Update to 1.6.0-beta9.
  - Update patches so that they apply cleanly.
  - Temporarily disable app_conference patch as it doesn't compile
  - config/scripts/postgres_cdr.sql has been merged into realtime_pgsql.sql
  - Re-add the asterisk-strip.sh script as a source file.
* Tue Apr 22 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.13.beta8
  - Update to 1.6.0-beta8
  - Contains fixes for AST-2008-006 / CVE-2008-1897
* Wed Apr 02 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.12.beta7.1
  - Return to stripped tarballs since there's more non-free content in
    the Asterisk tarballs than I thought.
* Sun Mar 30 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.11.beta7.1
  - Update to 1.6.0-beta7.1
  - Update patches
  - Back out some changes that were made because beta7 was tagged from
    the wrong branch.
* Fri Mar 28 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.10.beta7
  - Update to 1.6.0-beta7
  - The Asterisk tarball no longer contains the iLBC code, so we can
    directly use the upstream tarball without having to modify it.
  - Get rid of the asterisk-strip.sh script since it's no longer needed.
  - Diable build of codec_ilbc and format_ilbc (these do not contain any
    legally suspect code so they can be included in the tarball but it's
    pointless building them).
  - Update chan_mobile patch to fix API breakages.
  - Add a patch to chan_usbradio to fix API breakages.
* Thu Mar 27 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.9.beta6
  - Add Postgresql schemas from contrib as documentation to the Postgresql subpackage.
* Tue Mar 25 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.8.beta6
  - Update patches.
  - Add patch to compile against external libedit rather than using the
    in-tree version.
  - Add -Werror-implicit-function-declaration to optflags.
  - Get rid of hashtest and hashtest2 binaries that link to unfortified
    versions of *printf functions.  They are compiled with -O0 which
    somehow pulls in the wrong versions.  These programs aren't
    necessary to the operation of the package anyway.
* Wed Mar 19 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.6.beta6
  - Update to 1.6.0-beta6 to fix some security issues.
  -
  - AST-2008-002 details two buffer overflows that were discovered in
  - RTP codec payload type handling.
  -  * http://downloads.digium.com/pub/security/AST-2008-002.pdf
  -  * All users of SIP in Asterisk 1.4 and 1.6 are affected.
  -
  - AST-2008-003 details a vulnerability which allows an attacker to
  - bypass SIP authentication and to make a call into the context
  - specified in the general section of sip.conf.
  -  * http://downloads.digium.com/pub/security/AST-2008-003.pdf
  -  * All users of SIP in Asterisk 1.0, 1.2, 1.4, or 1.6 are affected.
  -
  - AST-2008-004 Logging messages displayed using the ast_verbose
  - logging API call are not displayed as a character string, they are
  - displayed as a format string.
  -  * http://downloads.digium.com/pub/security/AST-2008-004.pdf
  -
  - AST-2008-005 details a problem in the way manager IDs are caculated.
  -  * http://downloads.digium.com/pub/security/AST-2008-005.pdf
* Tue Mar 18 2008 Tom "spot" Callaway <tcallawa@redhat.com> - 1.6.0-0.5.beta5
  - add Requires for versioned perl (libperl.so)
* Wed Mar 05 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.4.beta5
  - Update to 1.6.0-beta5
  - Remove upstreamed patches.
* Mon Mar 03 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.3.beta4
  - Package the directory used to store monitor recordings.
* Tue Feb 26 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.2.beta4
  - Add patch from David Woodhouse that fixes building on PPC64.
* Tue Feb 26 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.1.beta4
  - Update to 1.6.0 beta 4
* Wed Feb 13 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.18-1
  - Update to 1.4.18.
  - Use -march=i486 on i386 builds for atomic operations (GCC 4.3
    compatibility).
  - Use "logger reload" instead of "logger rotate" in logrotate file
    (#432197).
  - Don't explicitly specify a group in in the init script to prevent
    Zaptel breakage (#426629).
  - Split app_ices out to a separate package so that the ices package
    can be required.
  - pbx_kdeconsole has been dropped, don't specifically exclude it from
    the build anymore.
  - Update app_conference patch.
  - Drop upstreamed libcap patch.
* Wed Jan 02 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.17-1
  - Update to 1.4.17 to fix AST-2008-001.
* Fri Dec 28 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.16.2-1
  - Update to 1.4.16.2
* Sat Dec 22 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.16.1-2
  - Bump release and rebuild to fix broken dep on uw-imap.
* Wed Dec 19 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.16.1-1
  - Update to the real 1.4.16.1.
* Wed Dec 19 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.16-2
  - Add patch to bring source up to version 1.4.16.1 which will be
    released shortly to fix some crasher bugs introduced by 1.4.16.
* Tue Dec 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.16-1
  - Update to 1.4.16 to fix security bug.
* Sat Dec 15 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-7
  - Really, really fix the build problems on devel.
* Sat Dec 15 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-6
  - Tweaks to get to build on x86_64
* Wed Dec 12 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-5
  - Exclude PPC64
* Wed Dec 12 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-4
  - Don't build apidocs by default since there's a problem building on x86_64.
* Tue Dec 11 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-3
  - Really get rid of zero length map files.
* Mon Dec 10 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-2
  - Get rid of zero length map files.
  - Shorten descriptions of voicemail subpackages
* Fri Nov 30 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.15-1
  - Update to 1.4.15
* Tue Nov 20 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.14-2
  - Fix license and other rpmlint warnings.
* Mon Nov 19 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.14-1
  - Update to 1.4.14
* Fri Nov 16 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.13-7
  - Add chan_mobile
* Tue Nov 13 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.13-6
  - Don't build cdr_sqlite because sqlite2 has been orphaned.
  - Rebase local patches to latest upstream SVN
  - Update app_conference patch to latest from upstream SVN
  - Apply post-1.4.13 patches from upstream SVN
* Wed Oct 10 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.13-1
  - Update to 1.4.13
* Tue Oct 09 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.12.1-1
  - Update to 1.4.12.1
* Wed Aug 22 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.11-1
  - Update to 1.4.11
* Fri Aug 10 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.10.1-1
  - Update to 1.4.10.1.
* Tue Aug 07 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.10-1
  - Update to 1.4.10 (security update).
* Tue Aug 07 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-7
  - Add a patch that allows alternate extensions to be defined in users.conf
* Mon Aug 06 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-6
  - Update app_conference patch. Enter/leave sounds are now possible.
* Fri Jul 27 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-5
  - Update patches so we don't need to run auto* tools, because autoconf
    2.60 is required and FC-6 and RHEL5 only have autoconf 2.59.
* Thu Jul 26 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-4
  - Don't build app_mp3
* Wed Jul 25 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-3
  - Add app_conference
* Wed Jul 25 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-2
  - Use plain useradd/groupadd rather than the fedora-usermgmt
  - Clean up requirements
  - Clean up build requirements by moving them to package sections
* Tue Jul 24 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.9-1
  - Update to 1.4.9
* Tue Jul 17 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.8-1
  - Update to 1.4.8
  - Drop ixjuser patch.
* Tue Jul 10 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.7.1-1
  - Update to 1.4.7.1
* Mon Jul 09 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.7-1
  - Update to 1.4.7
  - RxFAX/TxFAX applications
* Sun Jul 01 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.6-4
  - It's "sbin", not "bin" silly.
* Sat Jun 30 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.6-3
  - Add patch that lets us change TOS bits even when running non-root
* Fri Jun 29 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.6-2
  - voicemail needs to require /usr/bin/sox and /usr/bin/sendmail
* Fri Jun 29 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.6-1
  - Update to 1.4.6
  - Remove upstreamed patch.
* Thu Jun 21 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-10
  - Build the IMAP and ODBC storage options of voicemail and split
    voicemail out into subpackages.
  - Apply patch so that the system UW IMAP libray can be linked against.
  - Patch modules.conf.sample so that alternal voicemail modules don't
    get loaded simultaneously.
  - Link against system GSM library rather than internal copy.
  - Patch the Makefile so that it doesn't add redundant/wrong compiler
    options.
  - Force building with the standard RPM optimization flags.
  - Install the Asterisk MIB in a location that net-snmp can find it.
  - Only package docs in the main package that are relevant and that
    haven't been packaged by a subpackage.
  - Other minor cleanups.
* Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-9
  - Move sounds
* Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-8
  - Update some more ownership/permissions
* Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-7
  - Fix some permissions.
* Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-6
  - Update init script patch
  - Move pid file to subdir of /var/run
* Mon Jun 18 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-5
  - Update init script patch to run as non-root
* Sun Jun 17 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-4
  - Build modules that depend on FreeTDS.
  - Don't build voicemail with ODBC storage.
* Sun Jun 17 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-3
  - Have the build output the commands executing, rather than covering them up.
* Fri Jun 15 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.5-1
  - Update to 1.4.5
  - Remove upstreamed patch.
* Wed May 09 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.4-2
  - Add a patch to fix CVE-2007-2488/ASA-2007-013
* Fri Apr 27 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.4-1
  - Update to 1.4.4
* Wed Mar 21 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.2-1
  - Update to 1.4.2
* Tue Mar 06 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.1-2
  - Package the IAXy firmware
  - Minor clean-ups in files
* Mon Mar 05 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.1-1
  - Update to 1.4.1
  - Don't build/package codec_zap (zaptel 1.4.0 doesn't support it)
* Fri Dec 15 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.0-6.beta4
  - Update to 1.4.0-beta4
  - Various cleanups.
* Fri Oct 20 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.0-5.beta3
  - Don't package IAXy firmware because of license
  - Don't build app_rpt
  - Don't BR lm_sensors on PPC
  - Better way to prevent download/installation of sound archives
  - Redo tarball to eliminate non-free items
* Thu Oct 19 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.0-4.beta3
  - Remove explicit dependency on glibc-kernheaders.
  - Build jabber modules on PPC
* Wed Oct 18 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.0-3.beta3
  - *Really* update to beta3
  - chan_jingle has been taken out of 1.4
  - Move misplaced binaries to where they should be
* Wed Oct 18 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.0-2.beta3
  - Remove requirement on asterisk-sounds-core until licensing can be
    figured out.
* Wed Oct 18 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.0-1.beta3
  - Update to 1.4.0-beta3
* Sun Oct 15 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.0-0.beta2
  - Update to 1.4.0-beta2
* Tue Jul 25 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.10-1
  - Update to 1.2.10.
* Wed Jun 07 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.9.1
  - Update to 1.2.9.1
* Fri Jun 02 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.8
  - Update to 1.2.8
  - Add misdn.conf to list of configs.
  - Drop chan_bluetooth patch for now...
* Tue May 02 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.7.1-6
  - Zaptel subpackage shouldn't obsolete the sqlite subpackage.
  - Remove mISDN until build issues can be figured out.
* Mon Apr 24 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.7.1-5
  - Build mISDN channel drivers, modelled after spec file from David Woodhouse
* Thu Apr 20 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.7.1-4
  - Update chan_bluetooth patch with some additional information as to
    it's source and comment out more in the configuration file.
* Thu Apr 20 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.7.1-3
  - Add chan_bluetooth
* Wed Apr 19 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.7.1-2
  - Split off more stuff into subpackages.
* Wed Apr 12 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.7-1
  - Update to 1.2.7
* Mon Apr 10 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.6-3
  - Fix detection of libpri on 64 bit arches (taken from Matthias Saou's rpmforge package)
  - Change sqlite subpackage name to sqlite2 (there are sqlite3 modules in development).
* Thu Apr 06 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.6-2
  - Don't build GTK 1.X console since GTK 1.X is being moved out of core...
* Mon Mar 27 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.6-1
  - Update to 1.2.6
* Mon Mar 06 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.5-1
  - Update to 1.2.5.
  - Removed upstreamed MOH patch.
  - Add full urls to the app_(r|t)xfax.c sources.
  - Update spandsp patch.
* Mon Feb 13 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.4-4
  - Actually apply the patch.
* Mon Feb 13 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.4-3
  - Add patch to keep Asterisk from crashing when using MOH inside a MeetMe conference.
* Mon Feb 06 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.4-2
  - BR sqlite2-devel
* Tue Jan 31 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.4-1
  - Update to 1.2.4.
* Wed Jan 25 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.3-4
  - Took some tricks from Asterisk packages by Roy-Magne Mo.
  -   Enable gtk console module.
  -   BR gtk+-devel.
  -   Add logrotate script.
  -   BR sqlite2-devel and new sqlite subpackage.
  -   BR doxygen and graphviz for building duxygen documentation. (But don't build it yet.)
* Wed Jan 25 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.3-3
  - Completely eliminate the "asterisk" user from the spec file.
  - Move more config files to subpackages.
  - Consolidate two patches that patch the init script.
  - BR curl-devel
  - BR alsa-lib-devel
  - alsa, curl, oss subpackages
* Wed Jan 25 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.3-2
  - Do not run as user "asterisk" as that prevents setting of IP TOS (which is bad for quality of service).
  - Add patch for setting TOS separately for SIP and RTP packets.
* Wed Jan 25 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.3-1
  - First version for Fedora Extras.

Files

/usr/lib64/asterisk/modules/app_directory_odbc.so
/usr/lib64/asterisk/modules/app_voicemail_odbc.so


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Fabrice Bellet, Sun Jun 10 17:51:23 2018