./comms/asterisk13, The Asterisk Software PBX

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Branch: CURRENT, Version: 13.19.0nb4, Package name: asterisk-13.19.0nb4, Maintainer: jnemeth

Asterisk is a complete PBX in software. It provides all of the
features you would expect from a PBX and more. Asterisk does voice
over IP in three protocols, and can interoperate with almost all
standards-based telephony equipment using relatively inexpensive
hardware.

Asterisk provides Voicemail services with Directory, Call Conferencing,
Interactive Voice Response, Call Queuing. It has support for
three-way calling, caller ID services, ADSI, SIP and H.323 (as both
client and gateway).

This is a long term support version. It is scheduled to go to
security fixes only on October 24th, 2018, and EOL on October 24th,
2019. See here for more information about Asterisk versions:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

NOTE: This version does not work with the zaptel drivers. It
requires the newer DAHDI drivers which are still being ported.
So, there is no hardware support available at this moment.


Required to run:
[textproc/libxml2] [www/curl] [audio/speex] [lang/perl5] [shells/bash] [devel/libuuid] [textproc/iksemel] [textproc/jansson] [audio/speexdsp] [comms/srtp]

Required to build:
[pkgtools/cwrappers]

Package options: asterisk-config, jabber, ldap, speex

Master sites: (Expand)


Version history: (Expand)


CVS history: (Expand)


   2018-07-04 15:40:45 by Jonathan Perkin | Files touched by this commit (423)
Log message:
*: Move SUBST_STAGE from post-patch to pre-configure

Performing substitutions during post-patch breaks tools such as mkpatches,
making it very difficult to regenerate correct patches after making changes,
and often leading to substituted string replacements being committed.
   2018-07-02 13:28:50 by D'Arcy J.M. Cain | Files touched by this commit (1)
Log message:
Remove redundant, commented PKGREVISION.
   2018-04-29 23:32:09 by Adam Ciarcinski | Files touched by this commit (629) | Package updated
Log message:
revbump for boost-libs update
   2018-04-16 16:35:28 by Thomas Klausner | Files touched by this commit (1284)
Log message:
Recursive bump for new fribidi dependency in pango.
   2018-04-14 09:34:46 by Adam Ciarcinski | Files touched by this commit (681) | Package updated
Log message:
revbump after icu update
   2018-03-12 12:18:01 by Thomas Klausner | Files touched by this commit (2155)
Log message:
Recursive bumps for fontconfig and libzip dependency changes.
   2018-01-23 09:26:08 by John Nemeth | Files touched by this commit (5) | Package updated
Log message:
update to Asterisk 13.19.0 -- this contains both security fixes
and general bug fixes, fixes AST-2017-005, AST-2017-006, AST-2017-007,
AST-2017-008, AST-2017-009, AST-2017-10, AST-2017-11, AST-2017-12,
AST-2017-13, and AST-2017-14 (note that a number of these only
pertain to PJSIP which isn't used in pkgsrc)

----- 13.19.0 -----

The Asterisk Development Team would like to announce the release
of Asterisk 13.19.0.

The release of Asterisk 13.19.0 resolves several issues reported
by the community and would have not been possible without your
participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
      incoming INVITE Request-URI.
      (Reported by Richard Mudgett)
 * ASTERISK-27413 - Add cache_media_frames debugging option.
      (Reported by Richard Mudgett)
 * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
      endpoint identification to IP only
      (Reported by Ben Merrills)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27531 - Compiler optimizations can break module load
      sequence.
      (Reported by abelbeck)
 * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
      Contact crashes asterisk
      (Reported by Ross Beer)
 * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
      read()
      (Reported by Abhay Gupta)
 * ASTERISK-25079 - AMI bridge of channels results in MOH not
      destroyed and robotic audio on one channel
      (Reported by Zane Conkle)
 * ASTERISK-27490 - chan_console: 'set active' fails to work
      (Reported by Tzafrir Cohen)
 * ASTERISK-24756 - ConfBridge sound_muted does not work from
      CLI or AMI
      (Reported by Thomas Frederiksen)
 * ASTERISK-25649 - Transfer application does not work with
      Local channels - documentation misleading
      (Reported by Ivan Ullmann)
 * ASTERISK-25869 - chan_sip: "rejected because extension not
      found" should be logged as a security event
      (Reported by Brian J. Murrell)
 * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
      streams
      (Reported by Wim De Vlaminck)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
      music the first time it is played
      (Reported by Thomas Frederiksen)
 * ASTERISK-19657 - Coverity Report: Fix issues for error type
      CHAR_IO
      (Reported by Matt Jordan)
 * ASTERISK-27175 - iax.conf demo peer is invalid
      (Reported by Tzafrir Cohen)
 * ASTERISK-27430 - README refers to security documents that do
      not exist.
      (Reported by Corey Farrell)
 * ASTERISK-20281 - "core set verbose" behaves strangely, can't
      alias it, cli.conf example broken
      (Reported by Tim Ringenbach at Asteria Solutions Group)
 * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
      FXS gateway
      (Reported by Tzafrir Cohen)
 * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
      RTCP packet will write past where it should
      (Reported by Vitezslav Novy)
 * ASTERISK-27408 - Identify causes and fix
      pjsip/resolver/srv/failover/in_dialog/transport_tcp
      (Reported by Corey Farrell)
 * ASTERISK-18411 - Queue members with hints for state_interface
      get stuck in "In Use" state.
      (Reported by Steven T.  Wheeler)
 * ASTERISK-26131 - chan_sip: Crash Asterisk (in
      sip_request_call at chan_sip.c) by making a call to a single
      character in a dot pattern match
      (Reported by Dwayne Hubbard)
 * ASTERISK-27475 - codec_opus requires libcurl
      (Reported by Samuel For)
 * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
      not applied on reload
      (Reported by John Bigelow)
 * ASTERISK-27465 - CLI Completion Not Working
      (Reported by Ross Beer)
 * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
      Variable CDR(amaflags)=...
      (Reported by Richard Mudgett)
 * ASTERISK-27453 - RTP: Blind transfer direct media scenario
      results in one way audio.
      (Reported by Richard Mudgett)
 * ASTERISK-20643 - SIP ICE support - remove hardcoded
      limitation on SDP size, make ICE support disabled by default in
      SIP, maybe provide a better warning message
      (Reported by Roy)
 * ASTERISK-26980 - pjsip: Clean up WebRTC disables
      (Reported by abelbeck)
 * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
      flooded with unauthenticated requests
      (Reported by George Joseph)
 * ASTERISK-27454 - res_http_post: Don't require
      GMIME_MAJOR_VERSION
      (Reported by Joshua Colp)
 * ASTERISK-23735 - Transcoding makes bad choice in high-rate
      translations
      (Reported by Richard Kenner)
 * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
      message.
      (Reported by Frank Durden)
 * ASTERISK-24662 - [patch] column and row headers for Signed
      Linear format variants in output of 'core show translation' are
      ambiguous
      (Reported by Rusty Newton)
 * ASTERISK-27353 - H323 audio starts with a delay of 2
      seconds.
      (Reported by Marco Giordani)
 * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
      media
      (Reported by Kevin Harwell)
 * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
      (srflx) with Dual-Stack.
      (Reported by Alexander Traud)
 * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
      IPv6 addresses.
      (Reported by Alexander Traud)
 * ASTERISK-27435 - [patch] configure:
      pjsip_evsub_set_uas_timeout not found.
      (Reported by Alexander Traud)
 * ASTERISK-27431 - Asterisk fails to build when openssl headers
      are not installed.
      (Reported by Corey Farrell)
 * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
      (Reported by Ivan Larionov)
 * ASTERISK-27421 - RTP source learning not working with devices
      that have some clock issues
      (Reported by nappsoft)
 * ASTERISK-27361 - Attended transfer crashes in Asterisk
      13.17.2
      (Reported by Alessandro Pimenta)
 * ASTERISK-27238 - Bridging: Crash freeing a frame that's
      already been freed
      (Reported by Richard Kenner)
 * ASTERISK-27412 - core: Audiohook freeing interpolated frame
      when it shouldn't.
      (Reported by Mikhail)
 * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
      channel variable before closing the file
      (Reported by George Joseph)
 * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
      insert same ip address in "source ip address" and "destination
      ip address" fields in HEP packets
      (Reported by Max Norba)
 * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
      is equal to RemoteAddress)
      (Reported by Vasilii Rogin)
 * ASTERISK-27415 - asterisk.conf: Setting astctl without
      setting astrundir is ineffective.
      (Reported by Corey Farrell)
 * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
      (Reported by Joshua Colp)
 * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
      responses.
      (Reported by Corey Farrell)
 * ASTERISK-27337 - chan_sip: Security vulnerability with client
      code header (revisited)
      (Reported by Richard Mudgett)
 * ASTERISK-27319 - (Security) Function in PJSIP 2.7
      miscalculates the length of an unsigned long variable in 64bit
      machines
      (Reported by Kim youngsung)
 * ASTERISK-27391 - Regression: Deadlock between AOR named lock
      and pjproject grp lock
      (Reported by shaurya jain)
 * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
      contact read from astdb or database
      (Reported by Aaron An)
 * ASTERISK-27290 - res_pjsip: PIDF contact field has
      malformed/invalid XML
      (Reported by basildane)
 * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
      values
      (Reported by seanchann.zhou)
 * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
      enabled but not configured.
      (Reported by Alexander Traud)
 * ASTERISK-26426 - format_ogg_opus: remove from source
      (Reported by Kevin Harwell)
 * ASTERISK-27378 - Modules: Fix issues with CLI completion.
      (Reported by Corey Farrell)
 * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
      character isn't allowed any more
      (Reported by Michael Maier)
 * ASTERISK-27390 - Audit menuselect module dependencies
      (Reported by Corey Farrell)
 * ASTERISK-27389 - Optional API modules should not allow
      unload.
      (Reported by Corey Farrell)
 * ASTERISK-27369 - Bridge() dialplan application fails without
      setting BRIDGERESULT channel variable
      (Reported by James Terhune)
 * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
      documentation
      (Reported by Igor Goncharovsky)
 * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
      'imap_delete_old_greeting'
      (Reported by Anthony Messina)
 * ASTERISK-27194 - jitterbuffer: Does not handle case where
      translator returns null frame.
      (Reported by Joshua Elson)
 * ASTERISK-26639 - core: Disabling xmldoc support does not
      work. Also results in abort during Asterisk startup.
      (Reported by Mr Dini)
 * ASTERISK-27372 - ARI: Node ARI client broken in latest
      versions of 13 and 14
      (Reported by Benjamin Keith Ford)
 * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
      absence of the Expires header field with an unsubscribe action.
      (Reported by Jonathan Cloots)
 * ASTERISK-25960 - The config_hook unit test causes Asterisk to
      crash if run a second time
      (Reported by George Joseph)
 * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
      when rtp_ipv6 set to yes
      (Reported by Martin Cisárik)
 * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
      curl is loaded
      (Reported by Ronald Raikes)
 * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
      but first on SDP media level.
      (Reported by Alexander Traud)
 * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
      Assertion on un/re-load: mod.id == -1
      (Reported by Tzafrir Cohen)
 * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
      enabling with conf file option - also 'sip set debug off'
      reports debugging disabled, when it really isn't
      (Reported by Rusty Newton)
 * ASTERISK-27328 - Missing openssl dependencies in
      res_rtp_asterisk and tcptls
      (Reported by Tzafrir Cohen)
 * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
      (o=) contains local address.
      (Reported by Alexander Traud)
 * ASTERISK-27343 - Fails to build in FreeBSD due to
      sys/sysmacros.h not existing there
      (Reported by Guido Falsi)
 * ASTERISK-27340 - backtrace.c: Crash due to double-free.
      (Reported by Corey Farrell)
 * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
      stopping.
      (Reported by Alexander Traud)
 * ASTERISK-27333 - sip_to_pjsip not correctly handling
      disallow=all directive
      (Reported by Torrey Searle)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24297 - cdr.c: Minor code optimizations.
      (Reported by Richard Mudgett)
 * ASTERISK-27449 - [PATCH] When failing to acquire target
      during attended transfer, display wanted extension
      (Reported by Niklas Larsson)
 * ASTERISK-27456 - app_voicemail: Add new object for
      VoicemailUserEntry
      (Reported by sungtae kim)
 * ASTERISK-27380 - ast_coredumper: allow pointing out the
      asterisk binary explicitly
      (Reported by Tzafrir Cohen)
 * ASTERISK-23556 - Compilation warning for invert.c (array
      subscript is above array bounds)
      (Reported by Marcello Ceschia)
 * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
      (Reported by Richard Mudgett)
 * ASTERISK-27335 - CDR performance needs improvement.
      (Reported by Richard Mudgett)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telep … og-13.19.0

Thank you for your continued support of Asterisk!

----- 13.18.5 -----

The Asterisk Development Team would like to announce security
releases for Asterisk 13, 14 and 15, and Certified Asterisk 13.18.
The available releases are released as versions 13.18.5, 14.7.5,
15.1.5 and 13.18-cert2.

The following security vulnerabilities were resolved in these versions:

* AST-2017-014: Crash in PJSIP resource when missing a contact header
  A select set of SIP messages create a dialog in Asterisk. Those
  SIP messages must contain a contact header. For those messages,
  if the header was not present and using the PJSIP channel driver,
  it would cause Asterisk to crash.  The severity of this vulnerability
  is somewhat mitigated if authentication is enabled. If authentication
  is enabled a user would have to first be authorized before reaching
  the crash point.

For a full list of changes in the current releases, please see the ChangeLogs:

https://downloads.asterisk.org/pub/tele … og-13.18.5

The security advisory is available at:

https://downloads.asterisk.org/pub/secu … 17-014.pdf

Thank you for your continued support of Asterisk!

----- 13.18.4 -----

The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
security releases are released as versions 13.13-cert9, 13.18.4,
14.7.4 and 15.1.4.

The release of these versions resolves the following security
vulnerabilities:

* AST-2017-012: Remote Crash Vulnerability in RTCP Stack
  If a compound RTCP packet is received containing more than
  one report (for example a Receiver Report and a Sender
  Report) the RTCP stack will incorrectly store report
  information outside of allocated memory potentially causing
  a crash.

For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telep … og-13.18.4

The security advisories are available at:
http://downloads.asterisk.org/pub/secur … 7-012.html
http://downloads.asterisk.org/pub/secur … 17-012.pdf

Thank you for your continued support of Asterisk!

----- 13.18.3 -----

The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
security releases are released as versions 13.13-cert8, 13.18.3,
14.7.3 and 15.1.3.

The release of these versions resolves the following security
vulnerabilities:

* AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
  If the chan_skinny (AKA SCCP protocol) channel driver is
  flooded with certain requests it can cause the asterisk
  process to use excessive amounts of virtual memory
  eventually causing asterisk to stop processing requests of
  any kind.

For a full list of changes in the current releases, please see the
ChangeLogs:
http://downloads.asterisk.org/pub/telep … og-13.18.3

The security advisories are available at:
http://downloads.asterisk.org/pub/secur … 17-013.pdf

Thank you for your continued support of Asterisk!

----- 13.18.2 -----

The Asterisk Development Team would like to announce the release
of Asterisk 13.18.2.

The release of Asterisk 13.18.2 resolves several issues reported
by the community and would have not been possible without your
participation.

Thank you!

The following issues are resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
      character isn't allowed any more
      (Reported by Michael Maier)
 * ASTERISK-27391 - Regression: Deadlock between AOR named lock
      and pjproject grp lock
      (Reported by shaurya jain)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telep … og-13.18.2

Thank you for your continued support of Asterisk!

----- 13.18.0 -----

The Asterisk Development Team would like to announce the release
of Asterisk 13.18.0.

The release of Asterisk 13.18.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Improvements made in this release:
-----------------------------------
 * ASTERISK-27278 - [patch] chan_sip: Provide access to read the
      full SIP Request-URI from INVITE
      (Reported by David J.  Pryke)
 * ASTERISK-27255 - alembic: Add support for Microsoft SQL
      server
      (Reported by Florian Floimair)
 * ASTERISK-27253 - [patch] libsrtp-2.1.x support
      (Reported by Alexander Traud)
 * ASTERISK-27220 - Enable CHANNEL function to get from and to
      tag from SIP Headers
      (Reported by Andre Nazario)
 * ASTERISK-27169 - Google OAuth 2.0 support for XMPP / Motif
      (Reported by Andrey)
 * ASTERISK-27173 - Support for GMIME 3.0
      (Reported by Tzafrir Cohen)
 * ASTERISK-27092 - [patch] app_queue: Add Priority to AMI
      QueueStatus
      (Reported by Niklas Larsson)
 * ASTERISK-27085 - [patch] chan_pjsip: Port SIPDtmfMode to
      chan_pjsip
      (Reported by Torrey Searle)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
      curl is loaded
      (Reported by Ronald Raikes)
 * ASTERISK-27372 - ARI: Node ARI client broken in latest
      versions of 13 and 14
      (Reported by Benjamin Keith Ford)
 * ASTERISK-27047 - res_pjsip: user=phone added to Anonymous
      caller-id when it shouldn't be.
      (Reported by dtryba)
 * ASTERISK-26988 - res_pjsip_session: user_eq_phone adds double
      user=phone parameters to URIs
      (Reported by dtryba)
 * ASTERISK-27270 - cdr_mysql: various crashes at second module
      reload if cdr_mysql.conf is configured
      (Reported by Tzafrir Cohen)
 * ASTERISK-27301 - [patch] app_queue: Music On Hold for
      real-time queues is not reset to default
      (Reported by Nathan Bruning)
 * ASTERISK-25266 - Application Originate returns SUCCESS to
      ORIGINATE_STATUS upon failure to originate
      (Reported by Allen Ford)
 * ASTERISK-27192 - res_pjsip: Loss of SIP registrations causing
      unavailable endpoints
      (Reported by Richard Mudgett)
 * ASTERISK-27305 - res_ari: Memory leaks in ARI when using
      Content-Type: application/json
      (Reported by David Hajek)
 * ASTERISK-26922 - chan_sip: tcpbind uses wrong source address
      (Reported by Ksenia)
 * ASTERISK-27324 - [patch] Dual-Stack server cannot be used as
      IPv4 client via TCP/TLS
      (Reported by Alexander Traud)
 * ASTERISK-27317 - vector: multiple evaluation of elem in
      AST_VECTOR_ADD_SORTED.
      (Reported by Corey Farrell)
 * ASTERISK-27318 - res_pjsip_mwi: uninitialized value from
      ast_strings_match
      (Reported by Corey Farrell)
 * ASTERISK-27296 - [patch] False positive busy checks when
      icalendar's recurrence-id mechanism is involved
      (Reported by Benoît Dereck-Tricot)
 * ASTERISK-27284 - Status of RFC 3323 and PJSIP
      (Reported by dtryba)
 * ASTERISK-27216 - app_queue: does its
      check-makeannouncement-logic twice each head-caller-loop
      (Reported by Stefan Engström)
 * ASTERISK-27295 - Contact is improperly translated after
      d178f497
      (Reported by Sean Bright)
 * ASTERISK-27292 - Multiple RTP Stream Created Breaking RFC2833
      (SSRC Changes)
      (Reported by Ross Beer)
 * ASTERISK-27289 - A codeblock that maintains a bug,but maybe
      the codeblock will never run
      (Reported by Huangyx)
 * ASTERISK-27283 - Realtime config fail with PostgreSQL version
      before 9.1
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-27257 - bridge_native_rtp: half-way direct media
      when using early bridging
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-27279 - Crash in pubsub_on_rx_request NULL pointer -
      Possible PJSIP Vulnerability
      (Reported by Ross Beer)
 * ASTERISK-26606 - tcptls: Incorrect OpenSSL function call
      leads to misleading error report
      (Reported by Bob Ham)
 * ASTERISK-16898 - SRTP unprotect: authentication failure when
      RTP sequence number switches from 65535 -> 0
      (Reported by Marcello Ceschia)
 * ASTERISK-27274 - RTCP needs better packet validation to
      resist port scans.
      (Reported by Richard Mudgett)
 * ASTERISK-27252 - RTP: One way audio with direct media and
      strictrtp=yes.
      (Reported by Richard Mudgett)
 * ASTERISK-25524 - module reload res_calendar.so does not
      reload everything in calendar.conf
      (Reported by Jesper)
 * ASTERISK-24588 - res_calendar does not process CalDAV from
      Owncloud [fix included]
      (Reported by Stefan Gofferje)
 * ASTERISK-25523 - res_calendar: Warning about invalid channel
      value (for notification) occurs even when event has no
      notification configured.
      (Reported by Jesper)
 * ASTERISK-21399 - RTP Multicast of L16 (type 10): Asterisk and
      wireshark disagree
      (Reported by Tzafrir Cohen)
 * ASTERISK-27248 - [patch]external_media_address and
      external_signaling_address don't always honor localnet
      (Reported by Walter Doekes)
 * ASTERISK-27165 - CDR: CDR(start,u) function won't work in
      cdr_custom config
      (Reported by Jacek Konieczny)
 * ASTERISK-27217 - chan_sip: Asterisk crashing when
      subscription doesn't get set
      (Reported by Bryan Walters)
 * ASTERISK-24066 - res_smdi: convert to astobj2
      (Reported by Corey Farrell)
 * ASTERISK-17540 - SDP origin attribute modified when issuing
      re-INVITE because of directmedia=yes
      (Reported by saghul)
 * ASTERISK-27254 - alembic: prune_on_boot fix erroneous
      (Reported by Florian Floimair)
 * ASTERISK-27232 - When in queue on g722 with interruptions,
      music on hold can get stuck and no longer play
      (Reported by Jens T.)
 * ASTERISK-27024 - nat/external_media settings ignored in
      14.4.1
      (Reported by Christopher van de Sande)
 * ASTERISK-26879 - PJSIP external_media_address ignored if no
      local_net options are provided
      (Reported by Matt Jordan)
 * ASTERISK-27236 - Segfault ast_channel_name (chan=0x0) at
      channel_internal_api.c:478 during T.38 Fax Receive
      (Reported by Ross Beer)
 * ASTERISK-27225 - Crash when freeing dtls_cfg->cafile
      (Reported by Richard Kenner)
 * ASTERISK-27177 - ooh323c: misleading indentation in
      addons/ooh323c/src/ooSocket.c
      (Reported by Tzafrir Cohen)
 * ASTERISK-27241 - libc segfault upon entry into app_directory
      (Reported by David Moore)
 * ASTERISK-27152 - Sending a "tel" uri in a From or To header
      in an unauthenticated message causes asterisk to crash
      (Reported by Ross Beer)
 * ASTERISK-27103 - core: ast_safe_system command injection
      possible.
      (Reported by Corey Farrell)
 * ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even
      with strict RTP enabled
      (Reported by Joshua Colp)
 * ASTERISK-26994 - Confbridge: CBAnn channels intermittently
      become stuck when caller hangs up before recording name
      (Reported by James Terhune)
 * ASTERISK-20858 - app_minivm fails to clean up mkstemp files
      (Reported by Walter Doekes)
 * ASTERISK-16777 - several filename bugs in Record()
      application
      (Reported by klaus3000)
 * ASTERISK-27168 - alembic: PJSIP scripts are missing column
      dtls_fingerprint in ps_endpoints table
      (Reported by Florian Floimair)
 * ASTERISK-23608 - ControlPlayback fails to play files with
      names containing certain non-alpha characters
      (Reported by Jonathan White)
 * ASTERISK-19103 - When using realtime queues, function
      QUEUE_MEMBER_LIST() will return an error if no other
      app/function has loaded the queues first. This problem does not
      exist if queues.conf is used.
      (Reported by Jim Van Meggelen)
 * ASTERISK-21241 - When using voicemail as announce only
      (maxmsg=0), the star dtmf to enter the voicemail is not honored
      (Reported by Eelco Brolman)
 * ASTERISK-27204 - [patch] app_queue: Wrong queue stat
      calculation
      (Reported by sungtae kim)
 * ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE
      is used
      (Reported by Torrey Searle)
 * ASTERISK-27207 - XMPP OAuth not working due to inverted
      logic
      (Reported by Michael Kuron)
 * ASTERISK-27174 - res_calendar_icalendar: Recurring events not
      being loaded from Google calendar using ical
      (Reported by Mark Thompson)
 * ASTERISK-27202 - If wget is not installed and "or" is not
      available, external components (excluding pjsip) are not
      installed
      (Reported by Seán C. McCord)
 * ASTERISK-27147 - Either asterisk or pjproject isn't re-using
      tcp connections (again)
      (Reported by George Joseph)
 * ASTERISK-27193 - IPv6 receive address in message doesn't
      include brackets
      (Reported by Scott Griepentrog)
 * ASTERISK-27158 - [patch] res_rtp_asterisk: RTCP statistics
      are not available when native bridge is used
      (Reported by Torrey Searle)
 * ASTERISK-27110 - RTP session is not fully destroyed on
      channel hangup
      (Reported by Matt Jordan)
 * ASTERISK-26745 - Asymmetric codecs when
      asymmetric_rtp_codec=no
      (Reported by Jesse Ross)
 * ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
      (Reported by Ira Emus)
 * ASTERISK-26659 - res_pjsip: PJSIP presence - missing braces
      around the status element in XML
      (Reported by Abraham Liebsch)
 * ASTERISK-27156 - Asterisk won't compile on Fedora 26 with
      devmode enabled.
      (Reported by Corey Farrell)
 * ASTERISK-27130 - Applications ARI: Unsubscribe action for
      deviceStates does not remove old subscriptions properly
      (Reported by Sergej Kasumovic)
 * ASTERISK-25810 - say.c calls for sounds in the subdir
      "digits" that don't exist (in Core). SayUnixTime or other Say...
      apps will fail out when they call these sounds.
      (Reported by Nicolas Riendeau)
 * ASTERISK-27142 - sounds: Conflict between files in
      asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
      (Reported by Corey Farrell)
 * ASTERISK-27124 - app_playback.c:say_date_generic use
      timezonename parameter
      (Reported by Holger Hans Peter Freyther)
 * ASTERISK-27133 - res_rtp_asterisk: RTCP does not use ICE when
      RTCP-MUX in use
      (Reported by Joshua Colp)
 * ASTERISK-27128 - [patch]res_stasis_snoop: When recording a
      snoop channel (using ARI) where no media is being received, no
      recording happens when theres no media
      (Reported by Dan Jenkins)
 * ASTERISK-27123 - confbridge: Name recordings are left on
      filesystem
      (Reported by Sergej Kasumovic)
 * ASTERISK-27122 - chan_iax2: On reload MWI taskprocessors keep
      adding up
      (Reported by Sergej Kasumovic)
 * ASTERISK-26807 - sounds: New 3-D Binaural audio features
      require new sound prompts
      (Reported by Rusty Newton)
 * ASTERISK-25816 - French conf-adminmenu, conf-usermenu prompts
      differ in content from the English files
      (Reported by Benoit Duverger)
 * ASTERISK-26274 - Resolve open sounds issues and then create a
      new sounds release (1.5.1? or 1.6?)
      (Reported by Rusty Newton)
 * ASTERISK-27127 - configs: Erroneous load directive in sample
      configuration results in "Error loading module
      'res_pjsip_multihomed.so'"
      (Reported by HZMI8gkCvPpom0tM)
 * ASTERISK-27105 - [patch]core: when setting 'maxfiles' in
      asterisk.conf, a message is printed, even in rasterisk -x
      (Reported by Tzafrir Cohen)
 * ASTERISK-27036 - res_pjsip: Asterisk crashes when an
      extension tries to use PJSIP trunk with from_user containing
      '@'
      (Reported by Maxim Vasilev)
 * ASTERISK-27023 - res_rtp_asterisk: Deadlock when TURN session
      in use
      (Reported by Jatin Jain)
 * ASTERISK-27093 - ODBC deadlocks when app_directory tries to
      play back non-existent voicemail greeting
      (Reported by James Terhune)

New Features made in this release:
-----------------------------------
 * ASTERISK-27215 - [patch]AMI : Add CancelAtxfer Action
      (Reported by Thomas Sevestre)
 * ASTERISK-27117 - core: Add support for timelen parsing to
      ast_parse_arg and ACO.
      (Reported by Corey Farrell)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telep … og-13.18.0

Thank you for your continued support of Asterisk!

----- 13.17.0 ----

The Asterisk Development Team would like to announce the release
of Asterisk 13.17.0.

The release of Asterisk 13.17.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27108 - Crash using 'data get' CLI command
      (Reported by Sean Bright)
 * ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add
      only really different domain with TLS.
      (Reported by Alexander Traud)
 * ASTERISK-27100 - channel: ast_waitfordigit_full fails to
      clear flag in an error branch.
      (Reported by Corey Farrell)
 * ASTERISK-27090 - PJSIP: Deadlock using TCP transport
      (Reported by Richard Mudgett)
 * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
      events
      (Reported by Ove Aursand)
 * ASTERISK-27065 - call hangup after leaving app_queue
      (Reported by Marek Cervenka)
 * ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code()
      (Reported by Ross Beer)
 * ASTERISK-27074 - core_local: local channel data not being
      properly unref'ed and unlocked
      (Reported by Kevin Harwell)
 * ASTERISK-27075 - bridge: stuck channel(s) after failed
      attended transfer
      (Reported by Kevin Harwell)
 * ASTERISK-24052 - app_voicemail reloads result in leaked IMAP
      sockets.
      (Reported by Louis Jocelyn Paquet)
 * ASTERISK-27060 - Comment typo format_g729.c
      (Reported by Matthew Fredrickson)
 * ASTERISK-27026 - res_ari: Crash when no ari.conf
      configuration file exists
      (Reported by Ronald Raikes)
 * ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan
      execution and application unregistration
      (Reported by Frederic LE FOLL)
 * ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at
      sorcery.c
      (Reported by Ryan Smith)
 * ASTERISK-27024 - nat/external_media settings ignored in
      14.4.1
      (Reported by Christopher van de Sande)
 * ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
      get_write_timeout
      (Reported by Jørgen H)
 * ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for
      RTCP component
      (Reported by Michael Walton)
 * ASTERISK-26923 - bridging: T.38 request is lost when channels
      are added to bridge
      (Reported by Torrey Searle)
 * ASTERISK-27053 - res_pjsip_refer/session: Calls dropped
      during transfer
      (Reported by Kevin Harwell)
 * ASTERISK-27052 - Asterisk build process fails with flag
      --with-pjproject-bundled with curl download command and slow
      network
      (Reported by alex)
 * ASTERISK-27039 - chan_pjsip: Device state is idle when
      channel from endpoint is in early media
      (Reported by Joshua Colp)
 * ASTERISK-26996 - chan_pjsip: Flipping between codecs
      (Reported by Michael Maier)
 * ASTERISK-26281 - chan_pjsip would send INVITE to
      'Unreachable' endpoints
      (Reported by Jacek Konieczny)
 * ASTERISK-26973 - bridge: Crash when freeing frame and
      snooping
      (Reported by Michel R. Vaillancourt)
 * ASTERISK-19291 - Background in realtime
      (Reported by Andrew Nowrot)
 * ASTERISK-27025 - channel / meetme: Fix missing parentheses
      (Reported by Joshua Colp)
 * ASTERISK-27021 - GET /recordings/stored returns 500 Internal
      Server Error
      (Reported by Tim Morgan)
 * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets
      in wrong byte order on Intel platform when using slin codec
      (Reported by Frankie Chin)
 * ASTERISK-23951 -  Asterisk attempts and fails to build
      format_mp3 even if mp3lib was not downloaded
      (Reported by Tzafrir Cohen)
 * ASTERISK-25294 - srtp's crypto_get_random deprecated
      (Reported by Tzafrir Cohen)
 * ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't
      describe BEEP argument
      (Reported by Rusty Newton)
 * ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set
      variable" command without args
      (Reported by Antoine Pitrou)
 * ASTERISK-25662 - Malformed AGI 520 Usage response
      (Reported by Tony Mountifield)
 * ASTERISK-25101 - DTLS configuration can not be specified in
      the general section - documentation
      (Reported by Ben Langfeld)
 * ASTERISK-27008 - res_format_attr_h264: SDP parse fails if
      fmtp optional parameters have a space
      (Reported by John Harris)
 * ASTERISK-26399 - app_queue: Agent not called when caller is
      parked
      (Reported by wushumasters)
 * ASTERISK-26400 - app_queue: Queue member stops being called
      after AMI "Redirect" action for queues with wrapuptime
      (Reported by Etienne Lessard)
 * ASTERISK-26715 - app_queue: Member will not receive any new
      calls after doing a transfer if wrapuptime = greater than 0 and
      using Local channel
      (Reported by David Brillert)
 * ASTERISK-26975 - app_queue: Non-zero wrapup time can cause
      agents not to receive queue calls after transfer queue call
      (Reported by Lorne Gaetz)
 * ASTERISK-27012 - app_confbridge: ConfBridge sometimes does
      not play user name recording while leaving
      (Reported by Robert Mordec)
 * ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with
      authentication failure 10 or 110
      (Reported by Javier Riveros)
 * ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice
      completion failure/delay if client offers rtcp-mux as
      negotiable
      (Reported by Stefan Engström)
 * ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
      when request and To URI differ
      (Reported by Yasin CANER)
 * ASTERISK-26789 - Audit manipulation of channel flags without
      locks
      (Reported by Joshua Colp)
 * ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
      6869i)
      (Reported by Matthias Binder)

Improvements made in this release:
-----------------------------------
 * ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could
      block PJSIP taskprocessor on startup
      (Reported by Alexei Gradinari)
 * ASTERISK-27043 - Core/BuildSystem: Add defines to fix build
      with LibreSSL
      (Reported by Guido Falsi)
 * ASTERISK-27042 - Unpatched asterisk sources fail to build on
      FreeBSD due to missing crypt.h file
      (Reported by Guido Falsi)
 * ASTERISK-26419 - audiohooks: Remove redundant codec
      translations when using audiohooks
      (Reported by Michael Walton)
 * ASTERISK-26976 - libsrtp-2.x.x support
      (Reported by Alex)
 * ASTERISK-26124 - res_agi: Set audio format for EAGI audio
      stream
      (Reported by John Fawcett)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telep … og-13.17.0

Thank you for your continued support of Asterisk!
   2018-01-01 22:18:57 by Adam Ciarcinski | Files touched by this commit (629) | Package updated
Log message:
Revbump after boost update