# changelog

author: Jared Smith <jsmith@fedoraproject.org> - 13.7.1-1
changelog:
  - Update to upstream 13.7.1 release for security fixes
  - Resolves AST-2016-001: BEAST vulnerability in HTTP server
  - Resolves AST-2016-002: File descriptor exhaustion in chan_sip
  - Resolves AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data
  - Full changelog at http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
  - Also build the 'radius' sub-package against freeradius-client-devel, as the
   radiusclient-ng project is dead


author: Jared Smith <jsmith@fedoraproject.org> - 13.9.1-1
changelog:
  - Update to upstream 13.9.1 release
  - Use bootstrap.sh instead of calling autoconf tools manually
  - Fix up shifting pjproject submodules
  - Fix up requires on speexdsp-devel for EPEL7 (RHBZ#1310444)


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 11.11.0-1
changelog:
  - 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


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 11.12.0-1
changelog:
  - 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


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 11.12.1-1
changelog:
  - 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


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 11.13.0-1
changelog:
  - 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


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 11.13.1-1
changelog:
  - 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


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 13.0.0-1
changelog:
  - The Asterisk Development Team is pleased to announce the release of
  - Asterisk 13.0.0. This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - Asterisk 13 is the next major release series of Asterisk. It is a Long Term
  - Support (LTS) release, similar to Asterisk 11. For more information about
  - support time lines for Asterisk releases, see the Asterisk versions page:
  - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
  -
  - For important information regarding upgrading to Asterisk 13, please see the
  - Asterisk wiki:
  -
  - https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13
  -
  - A short list of new features includes:
  -
  - * Asterisk security events are now provided via AMI, allowing end users to
  -   monitor their Asterisk system in real time for security related issues.
  -
  - * Both AMI and ARI now allow external systems to control the state of a mailbox.
  -   Using AMI actions or ARI resources, external systems can programmatically
  -   trigger Message Waiting Indicators (MWI) on subscribed phones. This is of
  -   particular use to those who want to build their own VoiceMail application
  -   using ARI.
  -
  - * ARI now supports the reception/transmission of out of call text messages using
  -   any supported channel driver/protocol stack through ARI. Users receive out of
  -   call text messages as JSON events over the ARI websocket connection, and can
  -   send out of call text messages using HTTP requests.
  -
  - * The PJSIP stack now supports RFC 4662 Resource Lists, allowing Asterisk to act
  -   as a Resource List Server. This includes defining lists of presence state,
  -   mailbox state, or lists of presence state/mailbox state; managing
  -   subscriptions to lists; and batched delivery of NOTIFY requests to
  -   subscribers.
  -
  - * The PJSIP stack can now be used as a means of distributing device state or
  -   mailbox state via PUBLISH requests to other Asterisk instances. This is
  -   analogous to Asterisk's clustering support using XMPP or Corosync; unlike
  -   existing clustering mechanisms, using the PJSIP stack to perform the
  -   distribution of state does not rely on another daemon or server to perform the
  -   work.
  -
  - And much more!
  -
  - More information about the new features can be found on the Asterisk wiki:
  -
  - https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation
  -
  - A full list of all new features can also be found in the CHANGES file:
  -
  - http://svnview.digium.com/svn/asterisk/branches/13/CHANGES
  -
  - For a full list of changes in the current release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 13.0.1-1
changelog:
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
  - security releases are released as versions 1.8.28-cert3, 11.6-cert8, 1.8.32.1,
  - 11.14.1, 12.7.1, and 13.0.1.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of these versions resolves the following security vulnerabilities:
  -
  - * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
  -   address families
  -
  -   Many modules in Asterisk that service incoming IP traffic have ACL options
  -   ("permit" and "deny") that can be used to whitelist or blacklist address
  -   ranges. A bug has been discovered where the address family of incoming
  -   packets is only compared to the IP address family of the first entry in the
  -   list of access control rules. If the source IP address for an incoming
  -   packet is not of the same address as the first ACL entry, that packet
  -   bypasses all ACL rules.
  -
  - * AST-2014-018: Permission Escalation through DB dialplan function
  -
  -   The DB dialplan function when executed from an external protocol, such as AMI,
  -   could result in a privilege escalation. Users with a lower class authorization
  -   in AMI can access the internal Asterisk database without the required SYSTEM
  -   class authorization.
  -
  - In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
  - security vulnerability:
  -
  - * AST-2014-014: High call load with ConfBridge can result in resource exhaustion
  -
  -   The ConfBridge application uses an internal bridging API to implement
  -   conference bridges. This internal API uses a state model for channels within
  -   the conference bridge and transitions between states as different things
  -   occur. Unload load it is possible for some state transitions to be delayed
  -   causing the channel to transition from being hung up to waiting for media. As
  -   the channel has been hung up remotely no further media will arrive and the
  -   channel will stay within ConfBridge indefinitely.
  -
  - In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
  - the following security vulnerability:
  -
  - * AST-2014-017: Permission Escalation via ConfBridge dialplan function and
  -                 AMI ConfbridgeStartRecord Action
  -
  -   The CONFBRIDGE dialplan function when executed from an external protocol (such
  -   as AMI) can result in a privilege escalation as certain options within that
  -   function can affect the underlying system. Additionally, the AMI
  -   ConfbridgeStartRecord action has options that would allow modification of the
  -   underlying system, and does not require SYSTEM class authorization in AMI.
  -
  - Finally, the release of 12.7.1 and 13.0.1 resolves the following security
  - vulnerabilities:
  -
  - * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack
  -
  -   The Asterisk module res_pjsip provides the ability to configure ACLs that may
  -   be used to reject SIP requests from various hosts. However, the module
  -   currently fails to create and apply the ACLs defined in its configuration
  -   file on initial module load.
  -
  - * AST-2014-015: Remote crash vulnerability in PJSIP channel driver
  -
  -   The chan_pjsip channel driver uses a queue approach for relating to SIP
  -   sessions. There exists a race condition where actions may be queued to answer
  -   a session or send ringing after a SIP session has been terminated using a
  -   CANCEL request. The code will incorrectly assume that the SIP session is still
  -   active and attempt to send the SIP response. The PJSIP library does not
  -   expect the SIP session to be in the disconnected state when sending the
  -   response and asserts.
  -
  - * AST-2014-016: Remote crash vulnerability in PJSIP channel driver
  -
  -   When handling an INVITE with Replaces message the res_pjsip_refer module
  -   incorrectly assumes that it will be operating on a channel that has just been
  -   created. If the INVITE with Replaces message is sent in-dialog after a session
  -   has been established this assumption will be incorrect. The res_pjsip_refer
  -   module will then hang up a channel that is actually owned by another thread.
  -   When this other thread attempts to use the just hung up channel it will end up
  -   using a freed channel which will likely result in a crash.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
  - AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the same
  - time as this announcement.
  -
  - For a full list of changes in the current releases, please see the ChangeLogs:
  -
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 13.0.2-1
changelog:
  - The Asterisk Development Team has announced security releases for Certified
  - Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
  - released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of these versions resolves the following security vulnerability:
  -
  - * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
  -
  -   When handling a WebSocket frame the res_http_websocket module dynamically
  -   changes the size of the memory used to allow the provided payload to fit. If a
  -   payload length of zero was received the code would incorrectly attempt to
  -   resize to zero. This operation would succeed and end up freeing the memory but
  -   be treated as a failure. When the session was subsequently torn down this
  -   memory would get freed yet again causing a crash.
  -
  - For more information about the details of this vulnerability, please read
  - security advisory AST-2014-019, which was released at the same time as this
  - announcement.
  -
  - For a full list of changes in the current releases, please see the ChangeLogs:
  -
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
  -
  - The security advisory is available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf


author: Bruno Wolff III <bruno@wolff.to> - 2.7.0-1
changelog:
  - Move to current upstream release
  - Changlog is available at http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog


author: Francois Aucamp <faucamp@fedoraproject.org> - 1.3-16
changelog:
  - Added patch declaring explicit libm linking dependency (RHBZ #564899)
  - Updated source and URL tags


author: Till Maas <opensource@till.name> - 1.0.3-5
changelog:
  - Rebuilt for Fedora 23 Change
    https://fedoraproject.org/wiki/Changes/Harden_all_packages_with_position-independent_code


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.12-0.3.beta3
changelog:
  -
  - The following are some of the issues resolved in this beta release:
  -
  -   * Prevent a CONNECT message from sending a CONNECT ACKNOWLEDGE in the
  -     wrong state.
  -     (issue #17360. Reported by: shawkris. Patched by rmudgett)
  -
  -   * Made Q.921 delay events to Q.931 if the event could immediately
  -     generate response frames.
  -     (closes issue #17360. Reported by: shawkris. Patched by rmudgett)
  -
  -   * BRI PTMP: Active channels not cleared when the interface goes down.
  -     (closes issue #17865. Reported by: wimpy. Patched by rmudgett)
  -
  -   * Segfault in pri_schedule_del() - ctrl value is invalid.
  -     (closes issue #17522)
  -     (closes issue #18032. Reported by: schmoozecom. Patched by rmudgett)
  -
  -   * Crash when receiving an unknown/unsupported message type.
  -     (closes issue #17968. Reported by: gelo. Patched by rmudgett)
  -
  -   * B410P gets incoming call packets on ISDN but Asterisk doesn't see the
  -     call.
  -     (closes issue #18232. Reported by: lelio. Patched by rmudgett)
  -
  -   * SABME flood on backup D-channel in NFAS configuration.
  -     (closes issue #18255. Reported by: bklang. Patched by rmudgett)
  -
  -   * Asterisk is getting a "No D-channels available!" warning message every
  -     4 seconds.
  -     (closes issue #17270. Reported by: jmls. Patched by rmudgett)
  -
  - This beta release contains several new features, among them:
  -
  - 1.) ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
  - 2.) ETSI Advice Of Charge (AOC) support
  - 3.) ETSI Explicit Call Transfer (ECT) support
  - 4.) ETSI Call Waiting support for ISDN phones
  - 5.) ETSI Malicious Call ID support
  -
  - For a full list of changes in the current release candidate, please see
  - the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.12-beta3


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 1.4.12-1
changelog:
  - The Asterisk Development Team announces the release of libpri version
  - 1.4.12.  This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/libpri/
  -
  - The following are some of the issues resolved in this release:
  -
  -  * Add call transfer exchange of subaddresses support and fix PTMP call
  -    transfer signaling.
  -
  -  * Invalid PTMP redirecting signaling as TE towards NT.
  -
  -  * Add Q931_IE_TIME_DATE to CONNECT message when in network mode.
  -    (issue #18047 (JIRA PRI-114). Reported by: wuwu. Patched by rmudgett)
  -
  -  * Swap of master/slave in pri_enslave() incorrect.
  -    (issue #18769 (JIRA PRI-120). Reported by: jcollie. Patched by jcollie)
  -
  -  * Fix I-frame retransmission quirks.
  -
  -  * Crash if NFAS swaps D channels on a call with an active timer.
  -
  -  * DMS-100 not receiving caller name anymore.
  -    (issue #18822 (JIRA PRI-121). Reported by: cmorford. Patched by rmudgett)
  -
  -  * B channel lost by incoming call in BRI NT PTMP mode.
  -
  -  * Implement the mandatory T312 timer for NT PTMP broadcast SETUP calls.
  -
  - This release contains several new features, among them:
  -
  - 1.) ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
  - 2.) ETSI Advice Of Charge (AOC) support
  - 3.) ETSI Explicit Call Transfer (ECT) support
  - 4.) ETSI Call Waiting support for ISDN phones
  - 5.) ETSI Malicious Call ID support
  - 6.) Add Display IE text handling options.
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.12


author: Jeffrey Ollie <jeff@ocjtech.us> - 1.4.13-1
changelog:
  - The Asterisk Development Team has announced the release of libpri 1.4.13.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/libpri
  -
  - The release of libpri 1.4.13 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:
  -
  - * --- Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8,
  -       and B410P cards.
  -   (Issue AST-598. Reported by Trey Blancher)
  -
  - * --- Implement handling a multi-channel RESTART request.
  -   (Closes issue PRI-93. Reported by Marcin Kowalczyk)
  -
  - * --- Removed MDL/TEI management configuration warning message.
  -   (Closes issue PRI-137. Reported by Bart Coninckx)
  -
  - * --- Allow passing compiler flags (CFLAGS, LDFLAGS)
  -   (Closes issue PRI-144. Reported by Tzafrir Cohen)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.13


author: Jorge A Gallegos <kad@blegh.net> - 1.9.5-0
changelog:
  - Rebuilt with latest stable version from upstream
  - Added Erlang, PAM and JVM plugins
  - Added router-related plugins
  - Added logger plugins


author: Jorge A Gallegos <kad@fedoraproject.org> - 1.9.17-0
changelog:
  - Rebuilt for version 1.9.17
  - Pulling in new documentation from https://github.com/unbit/uwsgi-docs


author: Jorge A Gallegos <kad@blegh.net> - 1.9.17-1
changelog:
  - Copying the version changelog to top-level doc
  - Compile with POSIX capabilities
  - Embed the loggers into the binary itself, no need for an extra package
  - Patching jvm plugin to support arm


author: Jorge A Gallegos <kad@blegh.net> - 1.9.17-2
changelog:
  - Uploaded new sources per spec rework (Jorge Gallegos)
  - Adding more router plugins (Jorge Gallegos)
  - Adding mod_proxy_uwsgi apache module (Jorge Gallegos)
  - Complying with the guidelines for source urls (Jorge Gallegos)
  - The settings in the service file were right before (Jorge Gallegos)
  - Enabling stats log socket, and capabilities (Jorge Gallegos)


author: Jorge A Gallegos <kad@fedoraproject.org> - 1.9.18.2-0
changelog:
  - Breaking up full version in 3 parts (Jorge Gallegos)
  - Update to latest stable 1.9.18.2 (Jorge Gallegos)
  - Forgot to disable debug mode (Jorge Gallegos)


author: Jorge A Gallegos <kad@blegh.net> - 1.9.18.2-1
changelog:
  - The changelog entry must match major + minor (Jorge Gallegos)
  - Adding more ignore entries (Jorge Gallegos)
  - The jvm arm patch has been merged upstream (Jorge Gallegos)
  - Updated license to 'GPLv2 with exceptions' (Jorge Gallegos)
  - Ugh messed up the doc sha (Jorge Gallegos)
  - Adding new sources, bumping up spec to 1.9.18.2 (Jorge Gallegos)


author: Jorge A Gallegos <kad@blegh.net> - 1.9.19-1
changelog:
  - Updating to latest stable, uploading new sources (Jorge Gallegos)
  - Forgot to delete the jvm arm patch file (Jorge Gallegos)


author: Jamie Nguyen <jamielinux@fedoraproject.org> - 1:1.6.1-1
changelog:
  - update to upstream release 1.6.1
  - (#1126891) CVE-2014-3556: SMTP STARTTLS plaintext injection flaw


author: Jamie Nguyen <jamielinux@fedoraproject.org> - 1:1.6.2-2
changelog:
  - create nginx-filesystem subpackage (patch from Remi Collet)
  - create /etc/nginx/default.d as a drop-in directory for configuration files
    for the default server block
  - clean up nginx.conf


author: Jared Smith <jsmiht@fedoraproject.org> - 2.3-2
changelog:
  - Disable video support, and specifically tell it not to use libyuv,
    as the version of libyuv in Fedora is too old
  - Disable ilbc codec support, as it is not needed


author: Peter Lemenkov <lemenkov@gmail.com> - 3.5.7-2
changelog:
  - Kill support for SysV branches (no more love for EL5 and EL6). Well keep
    scriptlet for upgrading from pre-2.8.4 versions for a while.


author: Peter Lemenkov <lemenkov@gmail.com> - 3.5.7-3
changelog:
  - Mark configs as owned by rabbitmq user/group
  - No need to mark tmpfiles-file as config
  - Various config cleanups


author: Peter Lemenkov <lemenkov@gmail.com> - 3.5.7-4
changelog:
  - Remove compatibility triggerun scriptlet
  - Don't wait for slave stop messages forever (patches #2, #3)
  - Drop dependency on syslog.target - this seems to be a leftover
  - Require epmd@0.0.0.0 to run. This is a default value. User should override it
    if necessary. This fixes rhbz #1302368


