./comms/asterisk16, The Asterisk Software PBX

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Branch: CURRENT, Version: 16.17.0nb3, Package name: asterisk-16.17.0nb3, 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

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

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:

Package options: asterisk-config, spandsp, speex

Master sites: (Expand)

Version history: (Expand)

CVS history: (Expand)

   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
      (Reported by Matthias Hensler)
 * ASTERISK-29071 - app_confbridge: Memory rises when
      jitterbuffer enabled and muting over AMI occurs
      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
 * 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
      (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
 * 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
 * ASTERISK-16799 - Callee declined when 'beep' audio file does
      not exist
      (Reported by IAMJames_)
 * ASTERISK-29313 - res_pjsip_refer:  Segfault in progress
      (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
      (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
 * ASTERISK-29196 - res_pjsip: Segmentation fault

      (Reported by Mauri de Souza Meneguzzo (3CPlus))
 * ASTERISK-29280 - chan_sip: Allow peers without audio
      (Reported by Alexander Traud)
 * ASTERISK-29265 - chan_sip: Allow text+video media streams,
      (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,
      (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
 * 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
      (Reported by Joshua C. Colp)
 * ASTERISK-29325 - res_pjsip_registrar: Include source IP
      address and port in log messages
      (Reported by Joshua C.
 * ASTERISK-29326 - asterisk: Update copyright/company

      (Reported by Joshua C. Colp)
 * ASTERISK-29244 - Add MixMonitorStart / Stop / Mute AMI
      (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)
   2021-02-11 12:54:13 by Ryo ONODERA | Files touched by this commit (3)
Log message:
asterisk16: Add forgotten patches
   2021-02-11 12:53:19 by Ryo ONODERA | Files touched by this commit (2) | Package updated
Log message:
asterisk16: Fix segfaut under NetBSD/aarch64 9.99.80. Bump PKGREVISION

The problem is reported by Markus Kilbinger on port-arm mailing list.
   2021-02-11 03:20:18 by Ryo ONODERA | Files touched by this commit (3) | Package updated
Log message:
asterisk16: Update to 16.16.0

The following issues are resolved in this release:

Security bugs fixed in this release:

  * [ASTERISK-29219]       res_pjsip_diversion: Crash if Tel URI contains
                             (Reported by Torrey Searle)

Bugs fixed in this release:

  * [ASTERISK-29229]       Stasis/messaging: text messages not dispatched to
                             all subscribers when using generic subscription
                             (Reported by Jean Aunis  Prescom)
  * [ASTERISK-29238]       chan_sip: SDP: Offers without any enabled stream
                             are accepted.
                             (Reported by Alexander Traud)
  * [ASTERISK-29237]       chan_sip: SDP: m=video is parsed even when
                             (Reported by Alexander Traud)
  * [ASTERISK-29222]       chan_sip: Hold/Resume an sRTP call on a video
                             enabled user-agent.
                             (Reported by Alexander Traud)
  * [ASTERISK-29240]       chan_pjsip: Incoming PJSIP calls set global
                             SIPDOMAIN instead of a channel variable
                             (Reported by Ivan Poddubny)
  * [ASTERISK-27902]       chan_pjsip isnt updating hangupcause on 4XX
                             (Reported by George Joseph)
  * [ASTERISK-28016]       PJSIP sends duplicate 183 Progress responses
                             (Reported by Alex Hermann)
  * [ASTERISK-28185]       chan_pjsip: Subsequent same responses are not
                             (Reported by Julien)
  * [ASTERISK-29230]       pjsip: Asterisk goes crazy and massively spams
                             logfile if registration cant be send
                             (Reported by Michael Maier)
  * [ASTERISK-29231]       pjsip: SIGSEGV in CLI if no trunk is registered
                             (Reported by Michael Maier)
  * [ASTERISK-29217]       LOCK() can grant the same lock to multiple
                             channels spuriously
                             (Reported by Jaco Kroon)
  * [ASTERISK-29201]       Crash occurs when Transfer and execute Hangup
                             before the Transfer result
                             (Reported by Dan Cropp)
  * [ASTERISK-28947]       Segmentation fault in mixmonitor_ds_destroy
                             (Reported by Robert Sutton)
  * [ASTERISK-29191]       tel: URI in Diversion header causes crash
                             (Reported by Mikhail Ivanov)
  * [ASTERISK-28883]       Spyee information ist missing in ChanSpyStop AMI
                             (Reported by Hendrik Wedhorn)
  * [ASTERISK-29188]       null media causing the Asterisk crash
                             (Reported by sungtae kim)
  * [ASTERISK-29209]       Debug messages printed by scope trace might be
                             missing newlines
                             (Reported by Alexander Traud)
  * [ASTERISK-29024]       pjsip: Route Header in Cancel request incorrectly
                             (Reported by Flole Systems)
  * [ASTERISK-29211]       res_musiconhold: Segfault on realtime music on
                             hold without entries
                             (Reported by Nathan Bruning)
  * [ASTERISK-29022]       Crash when manipulating PJSIP invite dlg ref
                             (Reported by Sean Bright)
  * [ASTERISK-29173]       Media cache URL requests allow infinite redirects
                             (Reported by Sean Bright)
  * [ASTERISK-29175]       res_pjsip_stir_shaken: Fix module description
                             (Reported by Stanislav Abramenkov)
  * [ASTERISK-29148]       AST_MODULE_INFO no, MODULEINFO depend
                             (Reported by Alexander Traud)
  * [ASTERISK-28798]       chan_sip: TCP/TLS client without server.
                             (Reported by Alexander Traud)
  * [ASTERISK-29165]       res_pjsip: malformed header Accept-Encoding in
                             OPTIONS response
                             (Reported by Alexander Greiner-Baer)
  * [ASTERISK-29161]       Incorrect setup of recall channels
                             (Reported by Boris P. Korzun)
  * [ASTERISK-29155]       app_queue: Deadlock between queues container and
                             individual queues
                             (Reported by George Joseph)

Improvements made in this release:

  * [ASTERISK-28549]       Two repeated 183
                             (Reported by Gant Liu)
  * [ASTERISK-29216]       contrib: systemd asterisk service for centos8 or
                             other newer linux versions
                             (Reported by Mark Petersen)
  * [ASTERISK-29143]       res_http_media_cache: HTTP media cache stored
                             hardcoded in /tmp
                             (Reported by laszlovl)
  * [ASTERISK-29118]       VoiceMail() should have an option to play
                             greetings as Early Media
                             (Reported by Juan Carlos Castro y Castro)
   2021-01-03 02:21:09 by Greg Troxel | Files touched by this commit (2) | Package updated
Log message:
asterisk16: Update to 16.15.1

upstream changes: security fixes and bug fixes