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asterisk-voicemail-11.13.1-1.fc20.x86_64.rpm

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• 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 “sub_min_expiry” 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 7 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 Valentinavičius)
  - * 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 4 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 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 3 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 3 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 2 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 7 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 5 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 5 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 9 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 4 2012 Dan Horák <dan[at]danny.cz> - 11.0.0-0.3.beta1

  - fix build on s390

• Tue Sep 4 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 9 2012 Petr Pisar <ppisar@redhat.com> - 10.5.2-1.1

  - Perl 5.16 rebuild

• Thu Jul 5 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 4 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 3 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 9 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 7 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.5-0.2

  - Rebuild for net-snmp 5.7

• Fri Jul 1 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 Mašláňová <mmaslano@redhat.com> - 1.8.4.2-1.2

  - Perl mass rebuild

• Fri Jun 10 2011 Marcela Mašláňová <mmaslano@redhat.com> - 1.8.4.2-1.1

  - Perl 5.14 mass rebuild

• Fri Jun 3 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 9 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3-0.5.rc2

  - Remove isa macros

• Wed Feb 9 2011 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.3-0.4.rc2

  - Make library dependencies architecture specific

• Mon Feb 7 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 8 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 6 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 9 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 2 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.8.0-0.2.beta2

  - Rebuild against libpri 1.4.12

• Mon Aug 2 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 1 2010 Marcela Maslanova <mmaslano@redhat.com> - 1.6.2.8-0.2.rc1

  - Mass rebuild with perl-5.12.0

• Tue May 4 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 3 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 8 2010 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.6-0.1.rc2

  - Update to 1.6.2.6-rc2

• Mon Mar 8 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 3 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 9 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.16.rc8

  - Update to 1.6.2.0-rc8

• Wed Dec 2 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.15.rc7

  - Update to 1.6.2.0-rc7

• Tue Dec 1 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 9 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.11.rc5

  - Update to 1.6.2.0-rc5

• Fri Nov 6 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 9 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.2.0-0.6.rc3

  - Require latex2html used in static-http documents

• Wed Oct 7 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 6 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 9 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 6 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 1 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 5 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 8 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 4 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.12.beta4

  - Fedora Directory Server compatibility patch/subpackage.

• Sun Jan 4 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.10.beta4

  - Fix up paths. BZ#477238

• Sat Jan 3 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.9.beta4

  - Update patches

• Sat Jan 3 2009 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.8.beta4

  - Update to 1.6.1-beta4

• Tue Dec 9 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.7.beta3

  - Update to 1.6.1-beta3

• Tue Dec 9 2008 Alex Lancaster <alexlan[AT]fedoraproject org> - 1.6.1-0.6.beta2

  - Rebuild for new gmime

• Fri Nov 7 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.5.beta2

  - Add patch to fix missing variable on PPC.

• Fri Nov 7 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 7 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.3.beta2

  - PPC systems don't have sys/io.h

• Fri Nov 7 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.1-0.2.beta2

  - Update to 1.6.1 beta 2

• Wed Nov 5 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 2 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 2 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 5 2008 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.6.0-0.4.beta5

  - Update to 1.6.0-beta5
  - Remove upstreamed patches.

• Mon Mar 3 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 2 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 9 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 7 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.10-1

  - Update to 1.4.10 (security update).

• Tue Aug 7 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 6 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 9 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.7-1

  - Update to 1.4.7
  - RxFAX/TxFAX applications

• Sun Jul 1 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 9 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 6 2007 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.1-2

  - Package the IAXy firmware
  - Minor clean-ups in files

• Mon Mar 5 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 7 2006 Jeffrey C. Ollie <jeff@ocjtech.us> - 1.2.9.1

  - Update to 1.2.9.1

• Fri Jun 2 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 2 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 6 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 6 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 6 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.