./comms/asterisk16, The Asterisk Software PBX

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Branch: CURRENT, Version: 16.19.0nb1, Package name: asterisk-16.19.0nb1, Maintainer: ryoon

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 2022-10-09, and EOL on 2023-10-09. For more
information about Asterisk versions, see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions


Required to run:
[textproc/libxml2] [www/curl] [audio/speex] [lang/perl5] [shells/bash] [security/openssl] [devel/libuuid] [comms/spandsp] [textproc/jansson] [audio/speexdsp] [comms/srtp] [lang/python37]

Required to build:
[pkgtools/cwrappers]

Package options: asterisk-config, spandsp, speex

Master sites: (Expand)


Version history: (Expand)


CVS history: (Expand)


   2021-10-26 12:06:07 by Nia Alarie | Files touched by this commit (85)
Log message:
comms: Replace RMD160 checksums with BLAKE2s checksums

All checksums have been double-checked against existing RMD160 and
SHA512 hashes
   2021-10-07 15:27:10 by Nia Alarie | Files touched by this commit (85)
Log message:
comms: Remove SHA1 hashes for distfiles
   2021-09-29 21:01:31 by Adam Ciarcinski | Files touched by this commit (872)
Log message:
revbump for boost-libs
   2021-08-09 15:13:14 by Ryo ONODERA | Files touched by this commit (3) | Package updated
Log message:
asterisk16: Update to 16.19.0

16.19.0
New Features made in this release:

  * [ASTERISK-29446]         app_confbridge: New ConfKick application
                             (Reported by N A)
  * [ASTERISK-29440]         app_confbridge: Allow ConfBridge answer to be
                             suppressed
                             (Reported by N A)
  * [ASTERISK-29431]         Minimum and maximum dialplan functions
                             (Reported by N A)
  * [ASTERISK-29439]         func_volume: Volume function can  t be read
                             (Reported by N A)

Bugs fixed in this release:

  * [ASTERISK-29475]         SayNumber triggers WARNING if caller hangs up
                             during application execution
                             (Reported by N A)
  * [ASTERISK-29404]         Consolidate res_pjsip_messaging fixes for domain
                             name
                             (Reported by George Joseph)
  * [ASTERISK-29441]         Core reload making TCP endpoints go offline
                             (Reported by Luke Escude)
  * [ASTERISK-29433]         res_rtp_asterisk: Server reflexive candidates use
                             incorrect raddr for RTCP
                             (Reported by Chris)
  * [ASTERISK-28237]           FRACK!, Failed assertion bad magic number
                             happens when unsubscribe an application from an
                             event source
                             (Reported by Lucas Tardioli Silveira)
  * [ASTERISK-28393]         Multidomain support issue
                             (Reported by Andrea Sannucci)
  * [ASTERISK-29397]         pjsip: Asterisk isn  t tolerant of RFC8760 UASs
                             (Reported by George Joseph)
  * [ASTERISK-24601]         Missing RFC4235 tags and attributes in PJSIP
                             NOTIFY event: dialog XML body
                             (Reported by Marco Paland)
  * [ASTERISK-29372]         file.c switch does not account for flash events
                             (Reported by N A)
  * [ASTERISK-29377]         cpool_release_pool   double free or corruption
                             (out)
                             (Reported by Robert Sutton)
  * [ASTERISK-29370]         chan_sip does not recognize application/hook-flash
                             (Reported by N A)
  * [ASTERISK-29358]         chan_pjsip: Trace message for progress is output
                             even if frame is not queued
                             (Reported by Michael Maier)
  * [ASTERISK-29030]         res_rtp_asterisk: Additional RTP-frame (with wrong
                             SSRC) gets inserted when switching from progress
                             to established
                             (Reported by Matthias Hensler)
  * [ASTERISK-29407]         chan_local: Filtering audio formats should not
                             occur on removed streams
                             (Reported by Joshua C. Colp)

Improvements made in this release:

  * [ASTERISK-29450]         Allow setting channel variables using Originate
                             application
                             (Reported by N A)
  * [ASTERISK-29460]         Recognize application/hook-flash in PJSIP
                             (Reported by N A)
  * [ASTERISK-29459]         Missing configuration from PJSIP to SIP conversion
                             script
                             (Reported by N A)
  * [ASTERISK-29434]         Asterisk reveals pjproject version in STUN packets
                             (Reported by Jeremy Lain  )
  * [ASTERISK-29349]         Silent voicemail option is not completely silent
                             (Reported by N A)
  * [ASTERISK-29380]         Add Flash AMI event to handle flash events
                             (Reported by N A)

16.18.0
Bugs fixed in this release:

  * [ASTERISK-29328]         translate.c: possible buffer overflow when
                             upsampling
                             (Reported by Jean Aunis    Prescom)
  * [ASTERISK-29379]         Segfault    ast_channel_is_multistream (chan=0x0)
                             at channel_internal_api.c:1590
                             (Reported by Ross Beer)
  * [ASTERISK-29364]         res_rtp_asterisk: standard deviation
                             miscalculation
                             (Reported by Kevin Harwell)
  * [ASTERISK-29373]         res_rtp_asterisk: Flash events are duplicated
                             (Reported by N A)
  * [ASTERISK-28356]         app_queue: CLI set ringinuse for realtime member
                             not working
                             (Reported by Michael)
  * [ASTERISK-24631]         Incorrect description of option   context   in
                             queues.conf.sample
                             (Reported by Etienne Lessard)
  * [ASTERISK-26614]         app_queue: updatecdr option in queues.conf does
                             effectively nothing
                             (Reported by Alexander Gonchiy)
  * [ASTERISK-25358]         dateformat not read from logger.conf by remote
                             console
                             (Reported by Igor Liferenko)
  * [ASTERISK-27542]         app_queue: When   queue show   CLI command is
                             executed a crash occurs
                             (Reported by Miguel Sanz)
  * [ASTERISK-29215]         res_pjsip_session: NULL active_media_state
                             topology caused asterisk crash
                             (Reported by sungtae kim)
  * [ASTERISK-29355]         app_queue: Queue member status message sent even
                             if status doesn  t change
                             (Reported by Roman Pertsev)
  * [ASTERISK-29035]         chan_local: Multistream support breaks T.38 faxing
                             (Reported by Matthias Hensler)
  * [ASTERISK-29354]         res_pjsip: Allow partial reloading of transports
                             (Reported by Joshua C. Colp)
  * [ASTERISK-29348]         menuselect doesn  t return errors in many cases
                             (Reported by George Joseph)
  * [ASTERISK-29352]         res_rtp_asterisk: Fix frame delivery time when
                             SSRC changes
                             (Reported by Joshua C. Colp)

Improvements made in this release:

  * [ASTERISK-29339]         loader: Let  s output warnings for deprecated
                             modules!
                             (Reported by Joshua C. Colp)
  * [ASTERISK-29337]         menuselect: Add ability to set deprecated in and
                             removed in versions for modules
                             (Reported by Joshua C. Colp)
  * [ASTERISK-29335]         xml: Embed module information into core XML
                             documentation.
                             (Reported by Joshua C. Colp)
  * [ASTERISK-29336]         documentation: Fix inconsistent support levels
                             (Reported by Joshua C. Colp)
   2021-05-24 21:56:06 by Thomas Klausner | Files touched by this commit (3575)
Log message:
*: recursive bump for perl 5.34
   2021-04-21 15:25:34 by Adam Ciarcinski | Files touched by this commit (864)
Log message:
revbump for boost-libs
   2021-04-21 13:43:04 by Adam Ciarcinski | Files touched by this commit (1822)
Log message:
revbump for textproc/icu
   2021-03-26 01:04:08 by Greg Troxel | Files touched by this commit (2) | Package updated
Log message:
comms/asterisk16: Update to 16.17.0

This is a micro update that is mostly security fixes and bug fixes
with very small improvements.  In addition to this being a security
fix, asterisk16 is a leaf package.

Upstream changes:

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-29305 - ASTERISK-29203 / AST-2021-002 -- Another
      scenario is causing a crash
      (Reported by Gregory Massel)
 * ASTERISK-29260 - sRTP Replay Protection ignored; even tears
      down long calls
      (Reported by Alexander Traud)
 * ASTERISK-29227 - res_pjsip_diversion: sending multiple 181
      responses causes memory corruption and crash
      (Reported by
      Ivan Poddubny)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29215 - res_pjsip_session: NULL active_media_state
      topology caused asterisk crash
      (Reported by sungtae kim)
 * ASTERISK-29035 - chan_local: Multistream support breaks T.38
      faxing
      (Reported by Matthias Hensler)
 * ASTERISK-29071 - app_confbridge: Memory rises when
      jitterbuffer enabled and muting over AMI occurs
      (Reported
      by Stefan Ruf)
 * ASTERISK-29329 - app_dial: DTMF to 'D' option gets duplicated
      if there are multiple progress events
      (Reported by N A)
 * ASTERISK-24434 - Fix differing usage of assignment operators
      in modules.conf
      (Reported by Rusty Newton)
 * ASTERISK-29306 - strings: Incorrect use of
      __attribute__((pure)) in ast_str_to_lower definition

      (Reported by Vitezslav Novy)
 * ASTERISK-29300 - res_rtp_asterisk: When native local bridging
      the remote SSRC becomes permanent
      (Reported by Sebastian
      Damm)
 * ASTERISK-29235 - res_pjsip_nat: Contact is rewritten on
      REGISTER responses with external_signaling_address

      (Reported by Brian Paboojian)
 * ASTERISK-29266 - ICE Role conflict with an unauthorized
      session
      (Reported by Salah Ahmed)
 * ASTERISK-29105 - chan_pjsip: 180 Ringing with SDP not changed
      into progress
      (Reported by Sebastian Damm)
 * ASTERISK-29297 - say: Y2021 problem – Asterisk cannot say
      year 2021 in Dutch
      (Reported by Jacek Konieczny)
 * ASTERISK-29315 - res_pjsip: re-registration gets stuck if
      setting initial auth credentials fails
      (Reported by Nick
      French)
 * ASTERISK-29312 - res_fax: asterisk fails to publish the
      Stasis and ReceiveFax status messages if the remote Station ID
      contains invalid UTF-8 characters
      (Reported by Alexei
      Gradinari)
 * ASTERISK-16799 - Callee declined when 'beep' audio file does
      not exist
      (Reported by IAMJames_)
 * ASTERISK-29313 - res_pjsip_refer:  Segfault in progress
      notify
      (Reported by George Joseph)
 * ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to
      return one (no more) record
      (Reported by Boris P. Korzun)
 * ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't

      (Reported by Benjamin Keith Ford)
 * ASTERISK-29311 - res_odbc_transaction sets forcecommit
      default value based on isolation level instead of forcecommit

      (Reported by Jaco Kroon)
 * ASTERISK-28452 - pjsip: <sess-version> of SDP is not
      incremented though SDP may be changed on reinvite without SDP
      offer
      (Reported by Michael Maier)
 * ASTERISK-29287 - app.h: C++ compatibility broken

      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28369 - app_queue: Member device state "invalid"
      when second call is ringing and hint is used
      (Reported by
      Boolah )
 * ASTERISK-29203 - res_pjsip_t38: Crash when changing state

      (Reported by Gregory Massel)
 * ASTERISK-29205 - res_rtp_asterisk: Asterisk crashes when
      making hold/unhold from webrtc client
      (Reported by Edvin
      Vidmar)
 * ASTERISK-29196 - res_pjsip: Segmentation fault

      (Reported by Mauri de Souza Meneguzzo (3CPlus))
 * ASTERISK-29280 - chan_sip: Allow peers without audio
      (text+video).
      (Reported by Alexander Traud)
 * ASTERISK-29265 - chan_sip: Allow text+video media streams,
      again.
      (Reported by Alexander Traud)
 * ASTERISK-29261 - res_pjsip: user=phone validation fail for
      isup numbers containing *#
      (Reported by Mark Petersen)
 * ASTERISK-29259 - channel: Allow text+video media streams,
      again.
      (Reported by Alexander Traud)
 * ASTERISK-29258 - chan_sip: Audio stream rejected, Other
      stream present: Invalid SDP.
      (Reported by Alexander Traud)
 * ASTERISK-29220 - After T38 reinvite response of 488 a
      subsequent G711 reinvite is not processed correctly. Instead the
      previous T38 session media is used
      (Reported by Robert
      Cripps)
 * ASTERISK-29248 - res_pjsip_session: res sometimes
      uninitialized reported by compiler Clang.
      (Reported by
      Alexander Traud)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29321 - sorcery: Add support for more intelligent
      reloading.
      (Reported by Joshua C. Colp)
 * ASTERISK-29325 - res_pjsip_registrar: Include source IP
      address and port in log messages
      (Reported by Joshua C.
      Colp)
 * ASTERISK-29326 - asterisk: Update copyright/company

      (Reported by Joshua C. Colp)
 * ASTERISK-29244 - Add MixMonitorStart / Stop / Mute AMI
      events
      (Reported by Sébastien Duthil)
 * ASTERISK-29275 - Support of MIME-type for wav16

      (Reported by Boris P. Korzun)
 * ASTERISK-29252 - TRANSFERSTATUSPROTOCOL variable to report
      Transfer (REFER) failure SIP code
      (Reported by Dan Cropp)
 * ASTERISK-29262 - Support of various URL-schemes by MoH

      (Reported by Boris P. Korzun)