# changelog

author: Jeffrey C. Ollie <jeff@ocjtech.us> - 13.1.0-1
changelog:
  - The Asterisk Development Team has announced the release of Asterisk 13.1.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 13.1.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following are the issues resolved in this release:
  -
  - New Features made in this release:
  - -----------------------------------
  -  * ASTERISK-24554 - AMI/ARI: Generate events on connected line
  -       changes (Reported by Matt Jordan)
  -
  - Bugs fixed in this release:
  - -----------------------------------
  -  * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
  -       against libsrtp-1.5.0 (Reported by Patrick Laimbock)
  -  * ASTERISK-24455 - func_cdr: CDR_PROP leaks payload (Reported by
  -       Corey Farrell)
  -  * ASTERISK-24454 - app_queue: ao2_iterator not destroyed, causing
  -       leak (Reported by Corey Farrell)
  -  * ASTERISK-24430 - missing letter "p" in word response in
  -       OriginateResponse event documentation (Reported by Dafi Ni)
  -  * ASTERISK-24437 - Review implementation of ast_bridge_impart for
  -       leaks and document proper usage (Reported by Scott Griepentrog)
  -  * ASTERISK-24453 - manager: acl_change_sub leaks (Reported by
  -       Corey Farrell)
  -  * ASTERISK-24457 - res_fax: fax gateway frames leak (Reported by
  -       Corey Farrell)
  -  * ASTERISK-24458 - chan_phone fails to build on big endian systems
  -       (Reported by Tzafrir Cohen)
  -  * ASTERISK-21721 - SIP Failed to parse multiple Supported: headers
  -       (Reported by Olle Johansson)
  -  * ASTERISK-24304 - asterisk crashing randomly because of unistim
  -       channel (Reported by dhanapathy sathya)
  -  * ASTERISK-24190 - IMAP voicemail causes segfault (Reported by
  -       Nick Adams)
  -  * ASTERISK-24462 - res_pjsip: Stale qualify statistics after
  -       disablementation (Reported by Kevin Harwell)
  -  * ASTERISK-24465 - audiohooks list leaks reference to formats
  -       (Reported by Corey Farrell)
  -  * ASTERISK-24466 - app_queue: fix a couple leaks to struct
  -       call_queue (Reported by Corey Farrell)
  -  * ASTERISK-24432 - Install refcounter.py when REF_DEBUG is enabled
  -       (Reported by Corey Farrell)
  -  * ASTERISK-24411 - [patch] Status of outbound registration is not
  -       changed upon unregistering. (Reported by John Bigelow)
  -  * ASTERISK-24476 - main/app.c / app_voicemail: ast_writestream
  -       leaks (Reported by Corey Farrell)
  -  * ASTERISK-24480 - res_http_websockets: Module reference decrease
  -       below zero (Reported by Corey Farrell)
  -  * ASTERISK-24482 - func_talkdetect: Fix stasis message leak in
  -       audiohook callback (Reported by Corey Farrell)
  -  * ASTERISK-24487 - configuration: sections should be loadable as
  -       template even when not marked (Reported by Scott Griepentrog)
  -  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
  -       unescapes semicolons ("\;" -> ";") turning them into comments
  -       (corruption) on rewrite of a config file (Reported by George
  -       Joseph)
  -  * ASTERISK-24438 - res_pjsip_multihomed.so blocks Asterisk reload
  -       when DNS settings invalid (Reported by Melissa Shepherd)
  -  * ASTERISK-24307 - Unintentional memory retention in stringfields
  -       (Reported by Etienne Lessard)
  -  * ASTERISK-24491 - Memory leak in res_hep (Reported by Zane
  -       Conkle)
  -  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
  -       extra calls to ast_module_unref (Reported by Corey Farrell)
  -  * ASTERISK-24447 - Bridge DTMF hooks: Audio doesn't pass when
  -       waiting for more matching digits. (Reported by Richard Mudgett)
  -  * ASTERISK-24257 - agent must dial acceptdtmf twice to bridge to
  -       queue caller (Reported by Steve Pitts)
  -  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
  -       (Reported by Corey Farrell)
  -  * ASTERISK-24250 - [patch] Voicemail with multi-recipients To:
  -       header fix (Reported by abelbeck)
  -  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
  -       length exceeds 50 (roughly) national symbols (Reported by
  -       Dmitriy Bubnov)
  -  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
  -       revision r227276 (Reported by Xavier Hienne)
  -  * ASTERISK-24505 - manager: http connections leak references
  -       (Reported by Corey Farrell)
  -  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
  -       coverage are enabled (Reported by Corey Farrell)
  -  * ASTERISK-24444 - PBX: Crash when generating extension for
  -       pattern matching hint (Reported by Leandro Dardini)
  -  * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP
  -       packet to JSON for res_hep_rtcp and report blocks are greater
  -       than 1 (Reported by Gregory Malsack)
  -  * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended
  -       transfer (Reported by Beppo Mazzucato)
  -  * ASTERISK-24501 - ARI: Moving a channel between bridges followed
  -       by a hangup can cause an ARI client to not receive an expected
  -       ChannelLeftBridge event before StasisEnd (Reported by Matt
  -       Jordan)
  -  * ASTERISK-24336 - PJSIP timer_min_se value under 90 causes crash
  -       (Reported by Leon Rowland)
  -  * ASTERISK-23651 - Reloading some modules that are loaded already,
  -       results in 'No such module' before a successful reload (Reported
  -       by Rusty Newton)
  -  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
  -       endmarked users when last marked user leaves (Reported by Matt
  -       Jordan)
  -  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
  -       (Reported by David Woolley)
  -  * ASTERISK-24508 - pjsip - REFER request from SNOM is rejected
  -       with "400 bad request" - DEBUG shows "Received a REFER without a
  -       parseable Refer-To" (Reported by Beppo Mazzucato)
  -  * ASTERISK-24535 - stringfields: Fix regression from fix for
  -       unintentional memory retention and another issue exposed by the
  -       fix (Reported by Corey Farrell)
  -  * ASTERISK-24471 - Crash - assert_fail in libc in
  -       pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
  -       (Reported by yaron nahum)
  -  * ASTERISK-24528 - res_pjsip_refer: Sending INVITE with Replaces
  -       in-dialog with invalid target causes crash (Reported by Joshua
  -       Colp)
  -  * ASTERISK-24531 - res_pjsip_acl: ACLs not applied on initial
  -       module load (Reported by Matt Jordan)
  -  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
  -       allow blocked addresses through (Reported by Matt Jordan)
  -  * ASTERISK-24542 - [patch]Failure showing codecs via 'core show
  -       channeltype <tech>' (Reported by snuffy)
  -  * ASTERISK-24533 - 2 threads created per chan_sip entry (Reported
  -       by xrobau)
  -  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
  -       voicemail under high concurrency with an IMAP backend (Reported
  -       by David Duncan Ross Palmer)
  -  * ASTERISK-24572 - [patch]App_meetme is loaded without its
  -       defaults when the configuration file is missing (Reported by
  -       Nuno Borges)
  -  * ASTERISK-24573 - [patch]Out of sync conversation recording when
  -       divided in multiple recordings (Reported by Nuno Borges)
  -  * ASTERISK-24537 - Stasis: StasisStart/StasisEnd events are not
  -       reliably transmitted during transfers (Reported by Matt Jordan)
  -  * ASTERISK-24556 - Asterisk 13 core dumps when calling from pjsip
  -       extension to another pjsip extension  (Reported by Abhay Gupta)
  -
  - Improvements made in this release:
  - -----------------------------------
  -  * ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR
  -       property 'unanswered' (Reported by Matt Jordan)
  -  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
  -       column in the cel_odbc module (Reported by Etienne Lessard)
  -  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
  -       recordings (Reported by Ben Smithurst)
  -  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
  -       lookups (Reported by Birger "WIMPy" Harzenetter)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 13.1.1-1:
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.cert-4, 1.8.32.2, 11.6-cert10,
  - 11.15.1, 12.8.1, and 13.1.1.
  -
  - These releases are available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases
  -
  - The release of these versions resolves the following security vulnerabilities:
  -
  - * AST-2015-001: File descriptor leak when incompatible codecs are offered
  -
  -                 Asterisk may be configured to only allow specific audio or
  -                 video codecs to be used when communicating with a
  -                 particular endpoint. When an endpoint sends an SDP offer
  -                 that only lists codecs not allowed by Asterisk, the offer
  -                 is rejected. However, in this case, RTP ports that are
  -                 allocated in the process are not reclaimed.
  -
  -                 This issue only affects the PJSIP channel driver in
  -                 Asterisk. Users of the chan_sip channel driver are not
  -                 affected.
  -
  - * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
  -
  -                 CVE-2014-8150 reported an HTTP request injection
  -                 vulnerability in libcURL. Asterisk uses libcURL in its
  -                 func_curl.so module (the CURL() dialplan function), as well
  -                 as its res_config_curl.so (cURL realtime backend) modules.
  -
  -                 Since Asterisk may be configured to allow for user-supplied
  -                 URLs to be passed to libcURL, it is possible that an
  -                 attacker could use Asterisk as an attack vector to inject
  -                 unauthorized HTTP requests if the version of libcURL
  -                 installed on the Asterisk server is affected by
  -                 CVE-2014-8150.
  -
  - For more information about the details of these vulnerabilities, please read
  - security advisory AST-2015-001 and AST-2015-002, which were released at the same
  - time as this announcement.
  -
  - For a full list of changes in the current releases, please see the ChangeLogs:
  -
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
  - http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
  - http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
  -
  - The security advisories are available at:
  -
  -  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
  -  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 13.2.0-1:
changelog:
  - The Asterisk Development Team has announced the release of Asterisk 13.2.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 13.2.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following are the issues resolved in this release:
  -
  - Bugs fixed in this release:
  - -----------------------------------
  -  * ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them
  -       all at the same time. (Reported by Richard Mudgett)
  -  * ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow
  -       when using non-default sorcery wizard (Reported by Kevin
  -       Harwell)
  -  * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
  -       from JSSIP (Reported by Badalian Vyacheslav)
  -  * ASTERISK-24607 - res_pjsip_session: re-INVITE with declined
  -       media streams results in 488 (Reported by Matt Jordan)
  -  * ASTERISK-24563 - Direct Media calls within private network
  -       sometimes get one way audio (Reported by Kevin Harwell)
  -  * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to
  -       race condition in accessing codec in stored ast_frame and codec
  -       core (Reported by Matt Jordan)
  -  * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
  -       enabled (Reported by Richard Mudgett)
  -  * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
  -       enabled (Reported by Andreas Steinmetz)
  -  * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
  -       casts char to unsigned int (Reported by Walter Doekes)
  -  * ASTERISK-24536 - AMI redirect with PJSIP fails to move extra
  -       channel (Reported by Niklas Larsson)
  -  * ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is
  -       chosen for RTP compatible channels when the DTMF mode is not
  -       compatible (Reported by Yaniv Simhi)
  -  * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
  -       level - 'Remote address is null, most likely RTP has been
  -       stopped' (Reported by Rusty Newton)
  -  * ASTERISK-24513 - Local channel apparently leaked in off-nominal
  -       DTMF attended transfer (Reported by Mark Michelson)
  -  * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
  -       on startup (Reported by Richard Kenner)
  -  * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
  -       destination when 'sendrpid=yes' (in proxy environment) (Reported
  -       by Karsten Wemheuer)
  -  * ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall
  -       calls to the transferrer. (Reported by Richard Mudgett)
  -  * ASTERISK-24376 - res_pjsip_refer: REFER request for remote
  -       session attempts to direct channel to external_replaces
  -       extension instead of context, without providing for the
  -       Referred-To SIP URI (Reported by Matt Jordan)
  -  * ASTERISK-24591 - Stasis() side of an ARI originated channel
  -       cannot be Redirected (Reported by Kinsey Moore)
  -  * ASTERISK-24049 - Asterisk Manager Interface: A number of list
  -       type responses aren't using astman_send_listack (Reported by
  -       Jonathan Rose)
  -  * ASTERISK-24637 - Channel re-enters Stasis() when it should not
  -       (Reported by John Bigelow)
  -  * ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does
  -       not function (Reported by John Kiniston)
  -  * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
  -       (Reported by Kristian Høgh)
  -  * ASTERISK-20744 - [patch] Security event logging does not work
  -       over syslog (Reported by Michael Keuter)
  -  * ASTERISK-24665 - Configure check required for
  -       pjsip_get_dest_info() (Reported by Mark Michelson)
  -  * ASTERISK-23850 - Park Application does not respect Return
  -       Context Priority (Reported by Andrew Nagy)
  -  * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
  -       in the CFlags returned (Reported by Diederik de Groot)
  -  * ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown
  -       while attempting to publish (Reported by Kevin Harwell)
  -  * ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown
  -       (Reported by Corey Farrell)
  -  * ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails
  -       on cross compilation (Reported by abelbeck)
  -  * ASTERISK-24624 - Transfer to invalid extension results in hung
  -       channel. (Reported by Zane Conkle)
  -  * ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf,
  -       Incorrect External Addresses is Used in SIP Packets When
  -       Responding to INVITE (Reported by David Justl)
  -  * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
  -       voicemail is not deleted after review, hangup (Reported by LEI
  -       FU)
  -  * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
  -       32-bit packages on 64-bit hosts (Reported by Ben Klang)
  -  * ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding
  -       to most traffic, potential deadlock (Reported by Jeff Collell)
  -  * ASTERISK-24560 - Creating a named ARI bridge twice causes a
  -       crash (Reported by Kinsey Moore)
  -  * ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when
  -       MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported
  -       by Matt Jordan)
  -  * ASTERISK-24640 - Registration pending stays forever after sip
  -       reload (Reported by Max Man)
  -  * ASTERISK-24673 - outgoing sip registers cannot be removed or
  -       modified without doing restart (or doing module unload
  -       chan_sip.so) (Reported by Stefan Engström)
  -  * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
  -       m() option does not queue an MWI event (Reported by Gareth
  -       Palmer)
  -  * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
  -       fails to get app name (Reported by John Bigelow)
  -  * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
  -       column comparison for 'defaultuser' (Reported by
  -       HZMI8gkCvPpom0tM)
  -  * ASTERISK-24693 - Investigate and fix memory leaks in Asterisk
  -       (Reported by Kevin Harwell)
  -  * ASTERISK-24626 - Voicemail passwords not being stored in ARA
  -       (Reported by Paddy Grice)
  -  * ASTERISK-24539 - Compile fails on OSX because of sem_timedwait
  -       in bridge_channel.c (Reported by George Joseph)
  -  * ASTERISK-24544 - Compile fails on OSX Yosemite because of
  -       incorrect detection of htonll and ntohll (Reported by George
  -       Joseph)
  -  * ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX'
  -       no longer displays user menus (Reported by Matt Jordan)
  -  * ASTERISK-24721 - manager: ModuleLoad action incorrectly reports
  -       'module not found' during a Reload operation (Reported by Matt
  -       Jordan)
  -  * ASTERISK-24719 - ConfBridge recording channels get stuck when
  -       recording started/stopped more than once (Reported by Richard
  -       Mudgett)
  -  * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
  -       by Kevin Harwell)
  -  * ASTERISK-24728 - tcptls: Bad file descriptor error when
  -       reloading chan_sip (Reported by Kevin Harwell)
  -  * ASTERISK-24729 - Outbound registration not occuring on new
  -       registrations after reload. (Reported by Richard Mudgett)
  -  * ASTERISK-24676 - Security Vulnerability: URL request injection
  -       in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
  -  * ASTERISK-24666 - Security Vulnerability: RTP not closed after
  -       sip call using unsupported codec (Reported by Y Ateya)
  -  * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
  -       versions (Reported by Jared Biel)
  -  * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
  -       Stephan Eisvogel)
  -  * ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson)
  -  * ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response
  -       is ever received (Reported by Marco Paland)
  -  * ASTERISK-24737 - When agent not logged in, agent status shows
  -       unavailable, queue status shows agent invalid (Reported by
  -       Richard Mudgett)
  -
  - Improvements made in this release:
  - -----------------------------------
  -  * ASTERISK-24552 - ARI: Allow associating a channel as an
  -       initiator of an Origination for record keeping purposes
  -       (Reported by Matt Jordan)
  -  * ASTERISK-24553 - ARI/AMI: Include language in standard channel
  -       snapshot output (Reported by Matt Jordan)
  -  * ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by
  -       Matt Jordan)
  -  * ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for
  -       connection-oriented transports. (Reported by Matt Jordan)
  -  * ASTERISK-24412 - [patch]Incomplete channel originate/continue
  -       handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
  -       Israel))
  -  * ASTERISK-24678 - [PATCH] Added atxfer* settings to
  -       features.conf.sample (Reported by Niklas Larsson)
  -  * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
  -       by cloos)
  -  * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by
  -       Dan Jenkins)
  -  * ASTERISK-24316 - For httpd server, need option to define server
  -       name for security purposes (Reported by Andrew Nagy)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 13.3.0-1:
changelog:
  - The Asterisk Development Team has announced the release of Asterisk 13.3.0.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 13.3.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following are the issues resolved in this release:
  -
  - New Features made in this release:
  - -----------------------------------
  -  * ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a
  -       channel (Reported by Matt Jordan)
  -  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
  -       (Reported by Dwayne Hubbard)
  -
  - Bugs fixed in this release:
  - -----------------------------------
  -  * ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid
  -       string copy (Reported by Yura Kocyuba)
  -  * ASTERISK-24748 - res_pjsip: If wizards explicitly configured in
  -       sorcery.conf false ERROR messages may occur (Reported by Joshua
  -       Colp)
  -  * ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked
  -       (Reported by Matt Jordan)
  -  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
  -       res_odbc (Reported by ibercom)
  -  * ASTERISK-24479 - Enable REF_DEBUG for module references
  -       (Reported by Corey Farrell)
  -  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
  -       fully disconnect underlying socket, leading to events being
  -       dropped with no additional information (Reported by Matt Jordan)
  -  * ASTERISK-24772 - ODBC error in realtime sippeers when device
  -       unregisters under MariaDB (Reported by Richard Miller)
  -  * ASTERISK-24752 - Crash in bridge_manager_service_req when bridge
  -       is destroyed by ARI during shutdown (Reported by Richard
  -       Mudgett)
  -  * ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported
  -       by Zane Conkle)
  -  * ASTERISK-24015 - app_transfer fails with PJSIP channels
  -       (Reported by Private Name)
  -  * ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk
  -       transfer scenario. (Reported by Mark Michelson)
  -  * ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by
  -       Niklas Larsson)
  -  * ASTERISK-24716 - Improve pjsip log messages for presence
  -       subscription failure (Reported by Rusty Newton)
  -  * ASTERISK-24612 - res_pjsip: No information if a required sorcery
  -       wizard is not loaded (Reported by Joshua Colp)
  -  * ASTERISK-24768 - res_timing_pthread: file descriptor leak
  -       (Reported by Matthias Urlichs)
  -  * ASTERISK-24685 - "pjsip show version" CLI command (Reported by
  -       Joshua Colp)
  -  * ASTERISK-24632 - install_prereq script installs pjproject
  -       without IPv6 support (Reported by Rusty Newton)
  -  * ASTERISK-24085 - Documentation - We should remove or further
  -       document the 'contact' section in pjsip.conf (Reported by Rusty
  -       Newton)
  -  * ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by
  -       JoshE)
  -  * ASTERISK-24700 - CRASH: NULL channel is being passed to
  -       ast_bridge_transfer_attended() (Reported by Zane Conkle)
  -  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
  -       (Reported by Corey Farrell)
  -  * ASTERISK-24799 - [patch] make fails with undefined reference to
  -       SSLv3_client_method (Reported by Alexander Traud)
  -  * ASTERISK-22670 - Asterisk crashes when processing ISDN AoC
  -       Events (Reported by klaus3000)
  -  * ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn
  -       call (Reported by Marcel Manz)
  -  * ASTERISK-24740 - [patch]Segmentation fault on aoc-e event
  -       (Reported by Panos Gkikakis)
  -  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
  -       for playing back messages stored in IMAP - play_message: No
  -       origtime (Reported by Graham Barnett)
  -  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
  -       OSX with 64 bit integers (Reported by Corey Farrell)
  -  * ASTERISK-24796 - Codecs and bucket schema's prevent module
  -       unload (Reported by Corey Farrell)
  -  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
  -       (Reported by Ashley Sanders)
  -  * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
  -       is invalid (Reported by Rusty Newton)
  -  * ASTERISK-24785 - 'Expires' header missing from 200 OK on
  -       REGISTER (Reported by Ross Beer)
  -  * ASTERISK-24677 - ARI GET variable on channel provides unhelpful
  -       response on non-existent variable (Reported by Joshua Colp)
  -  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
  -       (Reported by Kevin Harwell)
  -  * ASTERISK-24812 - ARI: Creating channels through /channels
  -       resource always uses SLIN, which results in unneeded transcoding
  -       (Reported by Matt Jordan)
  -  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
  -       thread ID being passed to pthread_kill (Reported by JoshE)
  -  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
  -       fail (Reported by Terry Wilson)
  -  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
  -       SRTP for audio, but they responded without it' is ambiguous and
  -       wrong in some cases (Reported by Rusty Newton)
  -  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
  -       error response and BYE are sent to the caller (Reported by
  -       Makoto Dei)
  -  * ASTERISK-18105 - most of asterisk modules are unbuildable in
  -       cygwin environment (Reported by feyfre)
  -  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
  -  * ASTERISK-24751 - Integer values in json payload to ARI cause
  -       asterisk to crash (Reported by jeffrey putnam)
  -  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
  -       building a peer causes a peer poke during request handling
  -       (Reported by Richard Mudgett)
  -  * ASTERISK-24825 - Caller ID not recognized using
  -       Centrex/Distinctive dialing (Reported by Richard Mudgett)
  -  * ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not
  -       HAVE_PJPROJECT (Reported by Stefan Engström)
  -  * ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers
  -       (Reported by Kevin Harwell)
  -  * ASTERISK-24755 - Asterisk sends unexpected early BYE to
  -       transferrer during attended transfer when using a Stasis bridge
  -       (Reported by John Bigelow)
  -  * ASTERISK-24739 - [patch] - Out of files -- call fails --
  -       numerous files with inodes from under /usr/share/zoneinfo,
  -       mostly posixrules (Reported by Ed Hynan)
  -  * ASTERISK-23390 - NewExten Event with application AGI shows up
  -       before and after AGI runs (Reported by Benjamin Keith Ford)
  -  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
  -       voicemail stored in LDAP (Reported by Graham Barnett)
  -  * ASTERISK-24808 - res_config_odbc: Improper escaping of
  -       backslashes occurs with MySQL (Reported by Javier Acosta)
  -  * ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported
  -       by Anatoli)
  -  * ASTERISK-20850 - [patch]Nested functions aren't portable.
  -       Adapting RAII_VAR to use clang/llvm blocks to get the
  -       same/similar functionality. (Reported by Diederik de Groot)
  -  * ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI
  -       connection on error (Reported by Dmitriy Serov)
  -  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
  -       by Frank DiGennaro)
  -  * ASTERISK-21038 - Bad command completion of "core set debug
  -       channel" (Reported by Richard Kenner)
  -  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
  -       Dave Cabot)
  -  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
  -       Atis Lezdins)
  -  * ASTERISK-24876 - Investigate reference leaks from
  -       tests/channels/local/local_optimize_away (Reported by Corey
  -       Farrell)
  -  * ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported
  -       by Corey Farrell)
  -  * ASTERISK-24817 - init_logger_chain: unreachable code block
  -       (Reported by Corey Farrell)
  -  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
  -       snuffy)
  -  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
  -       under OpenBSD (Reported by snuffy)
  -
  - Improvements made in this release:
  - -----------------------------------
  -  * ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes
  -       (Reported by Ben Merrills)
  -  * ASTERISK-24811 - asterisk-publication sorcery object does not
  -       use realtime (Reported by Matt Hoskins)
  -  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
  -       Couldn't find mailbox %s in context (Reported by Graham Barnett)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0


author: Jeffrey C. Ollie <jeff@ocjtech.us> - 13.3.1-1:
changelog:
  - The Asterisk Development Team has announced the release of Asterisk 13.3.1.
  - This release is available for immediate download at
  - http://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 13.3.1 resolves an issue reported by the
  - community and would have not been possible without your participation.
  - Thank you!
  -
  - The following is the issue resolved in this release:
  -
  - * --- pjsip: resolve compatibility problem with ast_sip_session
  -   (Closes issue ASTERISK-24941. Reported by Matt Jordan)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  -
  - http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.1


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


author: Jared Smith <jsmith@fedoraproject.org> - 1.5.0-1
changelog:
  - Update to version 1.5
  - Add Swedish language files
  - Remove calls to %defattr macro, as they are no longer needed


author: Bruno Wolff III - 2.10.0-4
changelog:
  - Fix FTBFS from gcc error due to bad use of logical negation operator
  - This should also take care of the rebuild needed for perl


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


