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Revision 1.82 / (download) - annotate - [select for diffs], Wed Sep 29 19:00:25 2021 UTC (2 weeks, 6 days ago) by adam
Branch: MAIN
CVS Tags: HEAD
Changes since 1.81: +2 -1 lines
Diff to previous 1.81 (colored)

revbump for boost-libs

Revision 1.81 / (download) - annotate - [select for diffs], Mon Aug 9 13:13:14 2021 UTC (2 months, 1 week ago) by ryoon
Branch: MAIN
CVS Tags: pkgsrc-2021Q3-base, pkgsrc-2021Q3
Changes since 1.80: +2 -3 lines
Diff to previous 1.80 (colored)

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)

Revision 1.80 / (download) - annotate - [select for diffs], Mon May 24 19:49:17 2021 UTC (4 months, 3 weeks ago) by wiz
Branch: MAIN
CVS Tags: pkgsrc-2021Q2-base, pkgsrc-2021Q2
Changes since 1.79: +2 -2 lines
Diff to previous 1.79 (colored)

*: recursive bump for perl 5.34

Revision 1.79 / (download) - annotate - [select for diffs], Wed Apr 21 13:24:28 2021 UTC (5 months, 4 weeks ago) by adam
Branch: MAIN
Changes since 1.78: +2 -2 lines
Diff to previous 1.78 (colored)

revbump for boost-libs

Revision 1.78 / (download) - annotate - [select for diffs], Wed Apr 21 11:41:09 2021 UTC (5 months, 4 weeks ago) by adam
Branch: MAIN
Changes since 1.77: +2 -1 lines
Diff to previous 1.77 (colored)

revbump for textproc/icu

Revision 1.77 / (download) - annotate - [select for diffs], Fri Mar 26 00:04:08 2021 UTC (6 months, 3 weeks ago) by gdt
Branch: MAIN
CVS Tags: pkgsrc-2021Q1-base, pkgsrc-2021Q1
Changes since 1.76: +2 -3 lines
Diff to previous 1.76 (colored)

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 ãàAsterisk 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)

Revision 1.76 / (download) - annotate - [select for diffs], Thu Feb 11 11:53:19 2021 UTC (8 months, 1 week ago) by ryoon
Branch: MAIN
Changes since 1.75: +2 -1 lines
Diff to previous 1.75 (colored)

asterisk16: Fix segfaut under NetBSD/aarch64 9.99.80. Bump PKGREVISION

The problem is reported by Markus Kilbinger on port-arm mailing list.

Revision 1.75 / (download) - annotate - [select for diffs], Thu Feb 11 02:20:18 2021 UTC (8 months, 1 week ago) by ryoon
Branch: MAIN
Changes since 1.74: +2 -2 lines
Diff to previous 1.74 (colored)

asterisk16: Update to 16.16.0

Changelog:
The following issues are resolved in this release:

Security bugs fixed in this release:

  * [ASTERISK-29219]       res_pjsip_diversion: Crash if Tel URI contains
                             History-Info
                             (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
                             disabled.
                             (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
                             responses
                             (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
                             stopped
                             (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
                             Event
                             (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
                             set
                             (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
                             counts
                             (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)

Revision 1.74 / (download) - annotate - [select for diffs], Sun Jan 3 01:21:09 2021 UTC (9 months, 2 weeks ago) by gdt
Branch: MAIN
Changes since 1.73: +2 -3 lines
Diff to previous 1.73 (colored)

asterisk16: Update to 16.15.1

upstream changes: security fixes and bug fixes

Revision 1.73 / (download) - annotate - [select for diffs], Thu Dec 31 11:07:01 2020 UTC (9 months, 2 weeks ago) by nia
Branch: MAIN
CVS Tags: pkgsrc-2020Q4-base, pkgsrc-2020Q4
Changes since 1.72: +4 -2 lines
Diff to previous 1.72 (colored)

asterisk16: Avoid using -march=native, it breaks binary packages.

Also avoid passing crazy optimization and debug flags in general, just
honor the user's CFLAGS.

Revision 1.72 / (download) - annotate - [select for diffs], Thu Dec 10 13:52:30 2020 UTC (10 months, 1 week ago) by gdt
Branch: MAIN
Changes since 1.71: +2 -3 lines
Diff to previous 1.71 (colored)

asterisk16: Update to 16.15.0

Upstream changes:

  bugfixes
  minor improvements
  STIR/SHAKEN support

Revision 1.71 / (download) - annotate - [select for diffs], Fri Dec 4 20:45:03 2020 UTC (10 months, 2 weeks ago) by nia
Branch: MAIN
Changes since 1.70: +2 -2 lines
Diff to previous 1.70 (colored)

Revbump packages with a runtime Python dep but no version prefix.

For the Python 3.8 default switch.

Revision 1.70 / (download) - annotate - [select for diffs], Thu Nov 5 09:07:38 2020 UTC (11 months, 2 weeks ago) by ryoon
Branch: MAIN
Changes since 1.69: +2 -2 lines
Diff to previous 1.69 (colored)

*: Recursive revbump from textproc/icu-68.1

Revision 1.69 / (download) - annotate - [select for diffs], Mon Aug 31 18:06:44 2020 UTC (13 months, 2 weeks ago) by wiz
Branch: MAIN
CVS Tags: pkgsrc-2020Q3-base, pkgsrc-2020Q3
Changes since 1.68: +2 -2 lines
Diff to previous 1.68 (colored)

*: bump PKGREVISION for perl-5.32.

Revision 1.68 / (download) - annotate - [select for diffs], Mon Aug 17 20:18:19 2020 UTC (14 months ago) by leot
Branch: MAIN
Changes since 1.67: +2 -1 lines
Diff to previous 1.67 (colored)

*: revbump after fontconfig bl3 changes (libuuid removal)

Revision 1.67 / (download) - annotate - [select for diffs], Thu Aug 13 09:24:25 2020 UTC (14 months ago) by ryoon
Branch: MAIN
Changes since 1.66: +3 -3 lines
Diff to previous 1.66 (colored)

asterisk16: Update to 16.12.0

Changelog:
 Bugs fixed in this release:

-----------------------------------
[ASTERISK-28878] -
		chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
(Reported by Joseph Ades)
[ASTERISK-28965] -
		res_pjsip: Apply outbound proxy to static contacts on AOR
(Reported by Joshua C. Colp)
[ASTERISK-28930] -
		./configure --without-ssl build failure
(Reported by Jaco Kroon)
[ASTERISK-28886] -
		chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
(Reported by Jared Smith)
[ASTERISK-28957] -
		chan_sip: chan_sip does not process 400 response to an INVITE.
(Reported by Frederic LE FOLL)
[ASTERISK-28888] -
		res_corosync: causes asterisk crash in huge distributed environment.
(Reported by Università di Bologna - CESIA VoIP)
[ASTERISK-28955] -
		"setvar" doesn't work properly in dahdi-channels.conf
(Reported by Marin Odrljin)
[ASTERISK-28954] -
		StreamEcho() only returns 1 active stream
(Reported by Bill Kervaski)
[ASTERISK-28942] -
		res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching
(Reported by Joshua C. Colp)
[ASTERISK-28953] -
		res_pjsip_session: Preserve stream label
(Reported by Joshua C. Colp)
[ASTERISK-28952] -
		Queue wrapuptime sometimes not respected (based on stale lastcall time)
(Reported by Walter Doekes)
[ASTERISK-28950] -
		Stale code in app_queue to check untouched channel
(Reported by Walter Doekes)
[ASTERISK-28644] -
		Stale comment in app_queue about ring_entry exception
(Reported by Walter Doekes)
[ASTERISK-28948] -
		ARI channel create doesn't referencing the channel_id parameter
(Reported by sungtae kim)
[ASTERISK-28938] -
		core_unreal / core_local: Add support for multistream and re-negotiation
(Reported by Joshua C. Colp)
[ASTERISK-28939] -
		res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC
(Reported by Joshua C. Colp)
[ASTERISK-28944] -
		bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation
(Reported by Joshua C. Colp)
[ASTERISK-28923] -
		T.38 Segfaults in chan_pjsip_queryoption
(Reported by Yury Kirsanov)
[ASTERISK-28940] -
		/channels/create doesn't get any parameters from the body
(Reported by sungtae kim)
[ASTERISK-28936] -
		res_pjsip: crash when dialing non-sip uri
(Reported by Walter Doekes)
[ASTERISK-28900] -
		res_fax: Double frame free when gateway in use with off-nominal format usage
(Reported by Gregory Massel)
[ASTERISK-28929] -
		pjproject_bundled: Honor --without-pjproject.
(Reported by Alexander Traud)
[ASTERISK-28932] -
		res_pjsip_logger writing too big packets
(Reported by nappsoft)
[ASTERISK-28921] -
		Wrong return value check for fwrite when writing to pcap file
(Reported by nappsoft)

Improvements made in this release:

-----------------------------------
[ASTERISK-28959] -
		res_pjsip: Added option for disable rport parameter set
(Reported by sungtae kim)
[ASTERISK-28958] -
		Continue reading string when ping received by websocket
(Reported by Nickolay V. Shmyrev)
[ASTERISK-28945] -
		AMI SendText - add Content-Type parameter
(Reported by Kevin Harwell)
[ASTERISK-28949] -
		res_http_websocket: Add masking to websocket client
(Reported by Moises Silva)
[ASTERISK-28899] -
		Upgrade Asterisk to bundled pjproject 2.10
(Reported by Kevin Harwell)

Revision 1.66 / (download) - annotate - [select for diffs], Fri Jun 12 16:23:53 2020 UTC (16 months, 1 week ago) by ryoon
Branch: MAIN
CVS Tags: pkgsrc-2020Q2-base, pkgsrc-2020Q2
Changes since 1.65: +2 -3 lines
Diff to previous 1.65 (colored)

asterisk16: Update to 16.11.0

Changelog:
Bugs fixed in this release:
-----------------------------------
[ASTERISK-28940] -
		/channels/create doesn't get any parameters from the body
(Reported by sungtae kim)
[ASTERISK-28932] -
		res_pjsip_logger writing too big packets
(Reported by nappsoft)
[ASTERISK-28921] -
		Wrong return value check for fwrite when writing to pcap file
(Reported by nappsoft)
[ASTERISK-28794] -
		res_pjsip: Crash when escaping during URI printing
(Reported by nappsoft)
[ASTERISK-28884] -
		x-ast-orig-host not filtered out from request URI and To header
(Reported by nappsoft)
[ASTERISK-28871] -
		res_pjsip_session: Unnecessary re-Invite on call answer
(Reported by Alexei Gradinari)
[ASTERISK-28903] -
		res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
(Reported by Alexander Traud)
[ASTERISK-28898] -
		bridge_softmix: Conference bridge not passing silent rtp packets
(Reported by Jonathan Hunter)
[ASTERISK-28892] -
		res_musiconhold: Module res_musiconhold throws false warning
(Reported by Nicholas John Koch)
[ASTERISK-28904] -
		RTP ICE leaks the memory
(Reported by sungtae kim)
[ASTERISK-26780] -
		res_pjsip: PJSIP Registration Fails when transport=transport-udp6
(Reported by Peter Sokolov)
[ASTERISK-28854] -
		SIGSEGV when pjsip show history encounters IPV6 address
(Reported by Roger James)
[ASTERISK-28804] -
		[patch] app_osplookup.c: Avoid a format truncation.
(Reported by Alexander Traud)
[ASTERISK-28797] -
		[patch] tcptls: Fix notice when TLS is enabled but not configured.
(Reported by Alexander Traud)
[ASTERISK-28776] -
		Non async-signal-safe syscalls used after fork before exec
(Reported by nappsoft)
[ASTERISK-28870] -
		streams: One memory leak and one issue cloning streams
(Reported by George Joseph)
[ASTERISK-28829] -
		app_queue: leaking stasis subscription when Redirecting call
(Reported by lvl)
[ASTERISK-25844] -
		app_queue: Ghost channels in "core show channels" output
(Reported by Etienne Lessard)
[ASTERISK-22920] -
		Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling
(Reported by Shlomi Gutman)
[ASTERISK-28859] -
		pjsip: Increase maximum candidate count
(Reported by Joshua C. Colp)
[ASTERISK-28852] -
		Unprotected access to nochecksums variable, causes build failures
(Reported by Guido Falsi)
[ASTERISK-28848] -
		app_fax: Compile.
(Reported by Alexander Traud)


Improvements made in this release:
-----------------------------------
[ASTERISK-28895] -
		res_pjsip_logger: Add tons'o'functionality
(Reported by Joshua C. Colp)
[ASTERISK-28896] -
		ari: Add support for specifying variables on channel create
(Reported by Joshua C. Colp)
[ASTERISK-28879] -
		pjproject has race conditions in it's build system
(Reported by Guido Falsi)
[ASTERISK-28866] -
		third-party/pjproject/configure.m4 contains bashisms
(Reported by Guido Falsi)
[ASTERISK-28853] -
		Missing include on FreeBSD
(Reported by Guido Falsi)
[ASTERISK-28832] -
		chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio
(Reported by Peter Turczak)

Revision 1.65 / (download) - annotate - [select for diffs], Tue Jun 2 08:23:22 2020 UTC (16 months, 2 weeks ago) by adam
Branch: MAIN
Changes since 1.64: +2 -2 lines
Diff to previous 1.64 (colored)

Revbump for icu

Revision 1.64 / (download) - annotate - [select for diffs], Fri May 22 10:55:59 2020 UTC (16 months, 4 weeks ago) by adam
Branch: MAIN
Changes since 1.63: +2 -2 lines
Diff to previous 1.63 (colored)

revbump after updating security/nettle

Revision 1.63 / (download) - annotate - [select for diffs], Wed May 6 14:04:22 2020 UTC (17 months, 2 weeks ago) by adam
Branch: MAIN
Changes since 1.62: +2 -1 lines
Diff to previous 1.62 (colored)

revbump after boost update

Revision 1.62 / (download) - annotate - [select for diffs], Fri May 1 07:57:36 2020 UTC (17 months, 2 weeks ago) by ryoon
Branch: MAIN
Changes since 1.61: +2 -3 lines
Diff to previous 1.61 (colored)

asterisk16: Update to 16.10.0

Changelog:
16.10.0:
New Features made in this release:

-----------------------------------
[ASTERISK-6863] -
		[patch] allow Asterisk to set high ToS bits as non-root on Linux
(Reported by Matt Addison)

Bugs fixed in this release:

-----------------------------------
[ASTERISK-28852] -
		Unprotected access to nochecksums variable, causes build failures
(Reported by Guido Falsi)
[ASTERISK-28846] -
		stream: Enforce formats immutability
(Reported by Joshua C. Colp)
[ASTERISK-28847] -
		ARI channels cuts the endpoint string over 80 characters
(Reported by sungtae kim)
[ASTERISK-28811] -
		Crash occurs when fax session switches from T.38 to audio
(Reported by Alexey Vasilyev)
[ASTERISK-28839] -
		Sporadic crashes with Segmentation fault
(Reported by Joeran Vinzens)
[ASTERISK-28835] -
		IPv6 addresses in SDP incorrectly formatted
(Reported by Daniel Heckl)
[ASTERISK-28372] -
		Asterisk REPLY Wrong Contact header port (TCP)
(Reported by Anton Satskiy)
[ASTERISK-24428] -
		Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used
(Reported by sstream)
[ASTERISK-28838] -
		AST_MODULE_INFO requires, MODULEINFO does not mention
(Reported by Alexander Traud)
[ASTERISK-28841] -
		app_confbridge: Add support for disabling text messaging for a user
(Reported by Joshua C. Colp)
[ASTERISK-28837] -
		pjproject_bundled: Honor --without-pjproject.
(Reported by Alexander Traud)
[ASTERISK-28827] -
		res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK
(Reported by nappsoft)
[ASTERISK-27195] -
		chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets
(Reported by Joshua Roys)
[ASTERISK-28826] -
		res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK
(Reported by nappsoft)
[ASTERISK-28812] -
		First DTMF is not get
(Reported by Bernard Merindol)
[ASTERISK-28758] -
		pjsip startup errors when using "with-ssl" configure option
(Reported by Patrick Wakano)
[ASTERISK-28824] -
		BuildSystem: Search for Python/C API when possibly needed only.
(Reported by Alexander Traud)
[ASTERISK-27717] -
		[patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7.
(Reported by Alexander Traud)
[ASTERISK-28798] -
		[patch] chan_sip: TCP/TLS client without server.
(Reported by Alexander Traud)
[ASTERISK-28817] -
		chan_pjsip: constant DTMF tone if RTP is not setup yet
(Reported by Kevin Harwell)
[ASTERISK-28819] -
		[patch] bridge_softmix_binaural: Show state in menuselect.
(Reported by Alexander Traud)
[ASTERISK-28816] -
		[patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
(Reported by Alexander Traud)
[ASTERISK-28818] -
		[patch] BuildSystem: Allow space in path.
(Reported by Alexander Traud)
[ASTERISK-28796] -
		func_channel: cannot read fields exten, context, userfield, channame from dialplan
(Reported by Sébastien Duthil)
[ASTERISK-28809] -
		[patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction.
(Reported by Alexander Traud)
[ASTERISK-28803] -
		[patch] chan_unistim: Avoid tautological warnings with clang.
(Reported by Alexander Traud)
[ASTERISK-28808] -
		[patch] test_stasis: Avoid always true warning with clang.
(Reported by Alexander Traud)
[ASTERISK-28056] -
		res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR
(Reported by Jason Hord)
[ASTERISK-28795] -
		channel: write to a stream on multi-frame writes
(Reported by Kevin Harwell)
[ASTERISK-28789] -
		test_utils: incorrectly printing error 'declined to load'
(Reported by Alexander Traud)
[ASTERISK-28788] -
		func_aes: incorrectly printing error 'declined to load'
(Reported by Alexander Traud)
[ASTERISK-28790] -
		Crash during conference call using confbridge and video
(Reported by Pascal Cadotte Michaud)
[ASTERISK-16676] -
		DAHDIRAS fails to properly initiate pppd unless asterisk is running as root
(Reported by Jaco Kroon)
[ASTERISK-21205] -
		[patch] dundi_read_result crash due to negative number
(Reported by Jaco Kroon)
[ASTERISK-28784] -
		res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
(Reported by Joshua C. Colp)
[ASTERISK-28743] -
		Asterisk is crashing if the 200 OK with SDP
(Reported by sungtae kim)
[ASTERISK-28783] -
		res_pjsip_session: Allow default non-audio streams to have reflected state
(Reported by Joshua C. Colp)
[ASTERISK-28774] -
		chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge
(Reported by Michael Neuhauser)
[ASTERISK-20325] -
		Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples.
(Reported by Olivier Krief)
[ASTERISK-28780] -
		app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup
(Reported by Joshua C. Colp)
[ASTERISK-28773] -
		Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
(Reported by Torrey Searle)
[ASTERISK-28769] -
		DTLS Handshake Fails to Occur if ice_support is enabled but not used
(Reported by Torrey Searle)
[ASTERISK-28759] -
		A non negotiated rtp frame causes call disconnection when there is a SSRC change
(Reported by Paulo Vicentini)
[ASTERISK-26711] -
		func_enum: ENUM code wrong case
(Reported by Vitold)
[ASTERISK-23407] -
		Fix the FSF address in the headers of lots of pjproject files
(Reported by Jared Smith)
[ASTERISK-19460] -
		[patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string
(Reported by George Joseph)

Improvements made in this release:

-----------------------------------
[ASTERISK-28853] -
		Missing include on FreeBSD
(Reported by Guido Falsi)
[ASTERISK-28813] -
		func_volume: Allow decimal numbers as parameter to improve granularity
(Reported by Jean Aunis - Prescom)
[ASTERISK-27946] -
		dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't
(Reported by Joshua Elson)
[ASTERISK-28782] -
		Add support for Content-Disposition header in multi-part INVITES
(Reported by Torrey Searle)
[ASTERISK-28787] -
		res_pjsip_session: Decide more intelligently when to add video
(Reported by Joshua C. Colp)


16.9.0:
Bugs fixed in this release:
-----------------------------------

    [ASTERISK-28766] -

	 	PJSIP blind transfer not completed after using Proceeding()
(Reported by lvl)

    [ASTERISK-28685] -

	 	check_expr2: linking (when hardening) and cross-compiling troubles
(Reported by Sebastian Kemper)

    [ASTERISK-28764] -

	 	res_rtp_asterisk: Improve NACK support and seqno handling
(Reported by Joshua C. Colp)

    [ASTERISK-28755] -

	 	SIP/Stasis: SIP headers not transmitted in the "variables" field
(Reported by Jean Aunis - Prescom)

    [ASTERISK-28754] -

	 	ASTERISK-28738 Causes Audio Issue After Hold
(Reported by Ross Beer)

    [ASTERISK-28697] -

	 	res_pjsip: Named ACL does not update on reload if changed
(Reported by Timothy Vanderaerden)

    [ASTERISK-28746] -

	 	res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set
(Reported by George Joseph)

    [ASTERISK-28716] -

	 	ICE: pjnath shouldn't wait for ICE to complete before allowing sending
(Reported by Benjamin Keith Ford)

    [ASTERISK-28738] -

	 	Incorrect state machine used when MOH_PASSTHRU is used
(Reported by Torrey Searle)

    [ASTERISK-28742] -

	 	res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup
(Reported by Kevin Harwell)

    [ASTERISK-28735] -

	 	Realtime MoH Unknown format '' -- defaulting to SLIN
(Reported by Ross Beer)

    [ASTERISK-28730] -

	 	res_pjsip_session: Fix out of order session refreshes
(Reported by Joshua C. Colp)

    [ASTERISK-28718] -

	 	chan_sip: Returns 403 if RTP ports are depleted, should return 503
(Reported by Walter Doekes)

    [ASTERISK-28719] -

	 	Cannot remove defaultrule from queue using realtime queues
(Reported by EDV O-TON)

    [ASTERISK-28713] -

	 	res_stasis_playback: Error building JSON
(Reported by Sébastien Duthil)

    [ASTERISK-28714] -

	 	REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults
(Reported by Ross Beer)

    [ASTERISK-26082] -

	 	res_pjsip_messaging: MessageSend Content-Type can't be changed
(Reported by Alex)

    [ASTERISK-28423] -

	 	ARI causes STASIS Deadlock
(Reported by Ross Beer)

    [ASTERISK-28679] -

	 	stasis application is destroyed after its creation
(Reported by Francois Blackburn)

    [ASTERISK-25421] -

	 	PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending
(Reported by Dmitriy Serov)

    [ASTERISK-28686] -

	 	chan_sip strictrtp=yes fails when media source is changed: no audio
(Reported by Walter Doekes)

    [ASTERISK-28139] -

	 	RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls
(Reported by Paul Brooks)

    [ASTERISK-26955] -

	 	pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected
(Reported by Peter Sokolov)



Improvements made in this release:
-----------------------------------



    [ASTERISK-28750] -

	 	TLS/SSL Key too small error
(Reported by Martin Zeh)

    [ASTERISK-28733] -

	 	stream: Add support for adding/removing streams during SFU/calls
(Reported by Joshua C. Colp)

    [ASTERISK-24798] -

	 	Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor
(Reported by xrobau)

    [ASTERISK-28726] -

	 	install_prereq script uses the interactive mode when installing aptitude
(Reported by Sylvain Afchain)


16.8.0:
 New Features made in this release:

-----------------------------------
[ASTERISK-17491] -
		CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything
(Reported by candrews)
[ASTERISK-28639] -
		res_pjsip_endpoint_identifier_ip: Add ability to match on source port
(Reported by Sean Bright)

Bugs fixed in this release:

-----------------------------------
[ASTERISK-28679] -
		stasis application is destroyed after its creation
(Reported by Francois Blackburn)
[ASTERISK-28423] -
		ARI causes STASIS Deadlock
(Reported by Ross Beer)
[ASTERISK-28714] -
		REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults
(Reported by Ross Beer)
[ASTERISK-28677] -
		CDR billsec is always 0 for transferred calls
(Reported by Maciej Michno)
[ASTERISK-28702] -
		chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40
(Reported by Andrew Siplas)
[ASTERISK-28706] -
		silk 24hHz doesn't show up in 'core show translation' output
(Reported by Sean Bright)
[ASTERISK-24484] -
		Update documentation for statsd module - usage requirements unclear
(Reported by Dan Jenkins)
[ASTERISK-28695] -
		core: minmemfree watermark uses free RAM, not available RAM
(Reported by Kevin Flyn)
[ASTERISK-28693] -
		chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan
(Reported by Frank Matano)
[ASTERISK-23739] -
		[patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used
(Reported by Stas Kobzar)
[ASTERISK-27622] -
		empty voicemail.conf required for ARA (realtime) voicemail to leave message
(Reported by Jim Van Meggelen)
[ASTERISK-28349] -
		Pause reason not reported in QueueMember AMI event
(Reported by Niksa Baldun)
[ASTERISK-21794] -
		CLI command 'realtime update2' syntax failure when using according to usage help
(Reported by Cedric BASSAGET)
[ASTERISK-25429] -
		res_pjsip_endpoint_identifier_ip: Document support for hostnames
(Reported by Joshua C. Colp)
[ASTERISK-27775] -
		res_pjsip_notify: Multiple Event headers can be present instead of just one
(Reported by AvayaXAsterisk)
[ASTERISK-28682] -
		app_record: Lack of `beep` audio file causes application to return error and hangup
(Reported by Corey Farrell)
[ASTERISK-28507] -
		Wiki docs missing for MessageWaiting
(Reported by David M. Lee)
[ASTERISK-27759] -
		res_pjsip_pubsub: Subscription persistence does not preserve XML version number
(Reported by Bryan Nelson)
[ASTERISK-28605] -
		chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X
(Reported by Dirk Wendland)
[ASTERISK-28633] -
		stasis bridge topic leak
(Reported by Joeran Vinzens)
[ASTERISK-28492] -
		pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group
(Reported by Jean-Denis Girard)
[ASTERISK-28562] -
		SIP WSS message not processed until next frame arrives
(Reported by Robert Sutton)
[ASTERISK-27243] -
		contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax
(Reported by Richard Kenner)
[ASTERISK-28497] -
		func_odbc: truncating Unicode string on readsql
(Reported by Boris P. Korzun)
[ASTERISK-28647] -
		chan_sip: RTP frames not transmitted after emitting a COLP
(Reported by Jean Aunis - Prescom)
[ASTERISK-28667] -
		Asterisk ignores parsing of config files if a Byte order mark is present
(Reported by Robin Leffmann)
[ASTERISK-28664] -
		"trustrpid" is misspelled in sip_to_pjsip.py
(Reported by Pascal Cadotte Michaud)
[ASTERISK-28604] -
		app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
(Reported by George Joseph)
[ASTERISK-28659] -
		res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them
(Reported by nappsoft)
[ASTERISK-28660] -
		res_fax: wrap Asterisk initiated negotiation with config option
(Reported by Kevin Harwell)
[ASTERISK-28636] -
		app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.
(Reported by Frederic LE FOLL)
[ASTERISK-28626] -
		Missing arguments in PJSIP_CONTACT function documentation
(Reported by Pascal Cadotte Michaud)
[ASTERISK-28609] -
		Memory Leak in res_rtp_asterisk.c
(Reported by Ted G)
[ASTERISK-28651] -
		chan_sip logs errors on tx to non-existent TCP connections
(Reported by Jaco Kroon)
[ASTERISK-28502] -
		chan_pjsip incorrectly re-writes REGISTER 200 Response Contact
(Reported by Ross Beer)
[ASTERISK-28625] -
		Playback of local files impacted by large media cache
(Reported by Kevin Reeves)

Improvements made in this release:

-----------------------------------
[ASTERISK-28710] -
		Should be able to disable the /httpstatus URI in the built-in HTTP server
(Reported by Sean Bright)
[ASTERISK-28638] -
		Simplify dialplan for Dial, Page, and ChanIsAvail
(Reported by cmaj)
[ASTERISK-28673] -
		GET FULL VARIABLE documentation clarification
(Reported by Jonathan Harris)
[ASTERISK-28658] -
		app_confbridge: Add support for setting maximum sample rate
(Reported by Joshua C. Colp)

Revision 1.61 / (download) - annotate - [select for diffs], Sun Apr 12 08:28:23 2020 UTC (18 months, 1 week ago) by adam
Branch: MAIN
Changes since 1.60: +2 -2 lines
Diff to previous 1.60 (colored)

Recursive revision bump after textproc/icu update

Revision 1.60 / (download) - annotate - [select for diffs], Sun Mar 22 23:09:24 2020 UTC (18 months, 4 weeks ago) by tnn
Branch: MAIN
CVS Tags: pkgsrc-2020Q1-base, pkgsrc-2020Q1
Changes since 1.59: +7 -2 lines
Diff to previous 1.59 (colored)

asterisk16: fix L§énux packaging issues

Revision 1.59 / (download) - annotate - [select for diffs], Sun Mar 22 22:36:51 2020 UTC (18 months, 4 weeks ago) by tnn
Branch: MAIN
Changes since 1.58: +2 -1 lines
Diff to previous 1.58 (colored)

asterisk16: configure asks for -ledit. Comply.

Revision 1.58 / (download) - annotate - [select for diffs], Tue Mar 10 22:09:30 2020 UTC (19 months, 1 week ago) by wiz
Branch: MAIN
Changes since 1.57: +2 -2 lines
Diff to previous 1.57 (colored)

librsvg: update bl3.mk to remove libcroco in rust case

recursive bump for the dependency change

Revision 1.57 / (download) - annotate - [select for diffs], Sun Mar 8 16:49:02 2020 UTC (19 months, 1 week ago) by wiz
Branch: MAIN
Changes since 1.56: +2 -2 lines
Diff to previous 1.56 (colored)

*: recursive bump for libffi

Revision 1.56 / (download) - annotate - [select for diffs], Mon Jan 27 20:43:07 2020 UTC (20 months, 3 weeks ago) by gdt
Branch: MAIN
Changes since 1.55: +2 -2 lines
Diff to previous 1.55 (colored)

comms/asterisk16: Check for clang correctly

(This is a simple pkglint autofix, testing for clang being in
PKGSRC_COMPILER, rather than equal to, avoiding failure with
ccache/distcc.)

Revision 1.55 / (download) - annotate - [select for diffs], Sun Jan 26 17:30:50 2020 UTC (20 months, 3 weeks ago) by rillig
Branch: MAIN
Changes since 1.54: +2 -2 lines
Diff to previous 1.54 (colored)

all: migrate homepages from http to https

pkglint -r --network --only "migrate"

As a side-effect of migrating the homepages, pkglint also fixed a few
indentations in unrelated lines. These and the new homepages have been
checked manually.

Revision 1.54 / (download) - annotate - [select for diffs], Sat Jan 18 21:48:54 2020 UTC (21 months ago) by jperkin
Branch: MAIN
Changes since 1.53: +2 -2 lines
Diff to previous 1.53 (colored)

*: Recursive revision bump for openssl 1.1.1.

Revision 1.53 / (download) - annotate - [select for diffs], Sun Jan 12 20:20:07 2020 UTC (21 months, 1 week ago) by ryoon
Branch: MAIN
Changes since 1.52: +2 -1 lines
Diff to previous 1.52 (colored)

*: Recursive revbump from devel/boost-libs

Revision 1.52 / (download) - annotate - [select for diffs], Sat Jan 11 08:36:13 2020 UTC (21 months, 1 week ago) by ryoon
Branch: MAIN
Changes since 1.51: +2 -3 lines
Diff to previous 1.51 (colored)

asterisk16: Update to 16.7.0

Changelog:
16.7.0
Security bugs fixed in this release:
-----------------------------------
    [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter Addr of a peer (Reported by Andrey V. T.)
    [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows system commands execution (Reported by Eliel Sardañons)

Improvements made in this release:
-----------------------------------
    [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached (Reported by Daniel)
    [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam Banks)
    [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column (Reported by cmaj)
    [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup.  (Reported by Michael)
    [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold re-invites (Reported by Torrey Searle)
    [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by Florian Floimair)

Bugs fixed in this release:
-----------------------------------
    [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
    [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0 (Reported by George Joseph)
    [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them (Reported by nappsoft)
    [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR (Reported by Ross Beer)
    [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception (Reported by Walter Doekes)
    [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled (Reported by Bernhard Schmidt)
    [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime.  (Reported by Frederic LE FOLL)
    [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the same (Reported by Ross Beer)
    [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received (Reported by Salah Ahmed)
    [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported by Kevin Harwell)
    [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported by Michael Cargile)
    [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a short alarm (Reported by Frederic LE FOLL)
    [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match (Reported by Joshua Elson)
    [ASTERISK-26481] - FILE function grabs garbage along with read data when target line has no newline (Reported by Jonathan Harris)
    [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix bridge (Reported by Kevin Harwell)
    [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by Joshua C. Colp)
    [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
    [ASTERISK-28572] - Memory leaks in res_calendar_exchange and res_calendar_icalendar (Reported by Yoooooo Ha)
    [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup (Reported by Kevin Harwell)
    [ASTERISK-28590] - utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid argument" (Reported by Speed Dial Dave)
    [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by Salah Ahmed)
    [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column (Reported by Christoph Moench-Tegeder)
    [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
    [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported by Niklas Larsson)
    [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
    [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi container (Reported by Kevin Harwell)
    [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported by Marian Piater)
    [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
    [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd)
    [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by Jørgen H)
    [ASTERISK-28534] - Segmentation fault when there is no priority for an extension (Reported by Timothy Vanderaerden)
    [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured (Reported by Juan Martin)
    [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
    [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
    [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp)
    [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi)
    [ASTERISK-23756] - setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov)

New Features made in this release:
-----------------------------------
    [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" (Reported by lvl)
    [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec)
    [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
    [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI (Reported by Jeremiah Gadd)
    [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by Jørgen H)
    [ASTERISK-28534] - Segmentation fault when there is no priority for an extension (Reported by Timothy Vanderaerden)
    [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured (Reported by Juan Martin)
    [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
    [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
    [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp)
    [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi)
    [ASTERISK-23756] - setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov)

New Features made in this release:
-----------------------------------
    [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending" (Reported by lvl)
    [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported by Martin Tomec)
    [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization (Reported by Joshua C. Colp)

16.6.0
Security bugs fixed in this release:
-----------------------------------
[ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash (Reported by Alexei Gradinari)

Bugs fixed in this release:
-----------------------------------
[ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
[ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
[ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C. Colp)
[ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi)
[ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G)
[ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL)
[ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL)
[ASTERISK-28499] - translate: Crash when frame does not have a "src" field set (Reported by Gregory Massel)
[ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud)
[ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by Chris Savinovich)
[ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters (Reported by Dan Cropp)
[ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream (Reported by Alexei Gradinari)
[ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
[ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV (Reported by Jonas Swiatek)
[ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua C. Colp)
[ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by Salah Ahmed)
[ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries (Reported by Ian Jones)
[ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
[ASTERISK-28477] - Crash when not specifying "dbfile" in res_config_sqlite3.conf (Reported by Dennis)
[ASTERISK-28478] - Crash performing "core reload" with modified res_config_sqlite3.conf (Reported by Dennis)
[ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer (Reported by Dan Cropp)
[ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip) (Reported by Walter Doekes)

New Features made in this release:
-----------------------------------
[ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron Clark)
[ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain (Reported by Stas Kobzar)

Revision 1.51 / (download) - annotate - [select for diffs], Sat Dec 21 23:29:05 2019 UTC (21 months, 4 weeks ago) by joerg
Branch: MAIN
CVS Tags: pkgsrc-2019Q4-base, pkgsrc-2019Q4
Changes since 1.50: +3 -2 lines
Diff to previous 1.50 (colored)

Look into ${PREFIX}/lib when checking for libBlocksRuntime.

Revision 1.50 / (download) - annotate - [select for diffs], Sun Nov 24 01:14:10 2019 UTC (22 months, 3 weeks ago) by gdt
Branch: MAIN
Changes since 1.49: +2 -2 lines
Diff to previous 1.49 (colored)

comms/asterisk16: Fix compiler check via pkglint

AUTOFIX: Makefile:290: Replacing "${PKGSRC_COMPILER} == \"clang\"" with "${PKGSRC_COMPILER:Mclang}".

The PKGSRC_COMPILER can be a list of chained compilers, e.g. "ccache
distcc clang". Therefore, comparing it using == or != leads to wrong
results in these cases.

Revision 1.49 / (download) - annotate - [select for diffs], Sun Nov 3 12:04:12 2019 UTC (23 months, 2 weeks ago) by rillig
Branch: MAIN
Changes since 1.48: +7 -5 lines
Diff to previous 1.48 (colored)

comms: align variable assignments

pkglint -Wall -F --only aligned --only indent -r

Manually adjusted the indentation in asterisk15 and asterisk16 to avoid
too deep indentation.

Revision 1.48 / (download) - annotate - [select for diffs], Thu Aug 22 12:22:54 2019 UTC (2 years, 1 month ago) by ryoon
Branch: MAIN
CVS Tags: pkgsrc-2019Q3-base, pkgsrc-2019Q3
Changes since 1.47: +2 -1 lines
Diff to previous 1.47 (colored)

Recursive revbump from boost-1.71.0

Revision 1.47 / (download) - annotate - [select for diffs], Tue Aug 20 21:16:20 2019 UTC (2 years, 2 months ago) by ryoon
Branch: MAIN
Changes since 1.46: +5 -2 lines
Diff to previous 1.46 (colored)

Fix build on NetBSD 8

Revision 1.46 / (download) - annotate - [select for diffs], Tue Aug 20 13:47:42 2019 UTC (2 years, 2 months ago) by ryoon
Branch: MAIN
Changes since 1.45: +156 -109 lines
Diff to previous 1.45 (colored)

comms/asterisk16: import asterisk-16.5.0

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).

Revision 1.45, Fri Sep 14 02:41:04 2012 UTC (9 years, 1 month ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2013Q2-base, pkgsrc-2013Q2, pkgsrc-2012Q4-base, pkgsrc-2012Q4
Changes since 1.44: +1 -1 lines
FILE REMOVED

Remove Asterisk 1.6.  This version series went end-of-line on April
21st, 2012.  It most likely has multiple security issues.  By this
point, all users of this package should have migrated to comms/asterisk18
or comms/asterisk10 as this version has been marked as being
deprecated for some time now.

Note that this directory is likely to re-appear in late 2017 when
Asterisk 16 comes out, assuming the current schedule is followed.
However that will be a vastly different version as Asterisk 11 is
only in the RC stage now (i.e. it will be five major versions after
the one that is expected to be released later this year).

Revision 1.44 / (download) - annotate - [select for diffs], Sun Jul 15 16:26:11 2012 UTC (9 years, 3 months ago) by marino
Branch: MAIN
Changes since 1.43: +2 -0 lines
Diff to previous 1.43 (colored)

comms/asterisk16: Mark NOT-FOR-DRAGONFLY

This package has not been patched for DragonFly.
There are two newer packages, asterisk10 and asterisk18
According to commit messages, this package will be removed in
"not too distant future" due to being EOL.

Revision 1.43 / (download) - annotate - [select for diffs], Thu Jun 14 07:45:29 2012 UTC (9 years, 4 months ago) by sbd
Branch: MAIN
CVS Tags: pkgsrc-2012Q2-base, pkgsrc-2012Q2
Changes since 1.42: +2 -1 lines
Diff to previous 1.42 (colored)

Recursive PKGREVISION bump for libxml2 buildlink addition.

Revision 1.42 / (download) - annotate - [select for diffs], Fri May 4 16:06:13 2012 UTC (9 years, 5 months ago) by joerg
Branch: MAIN
Changes since 1.41: +2 -1 lines
Diff to previous 1.41 (colored)

Don't override optimizer settings with absurd levels.
Fix inline definitions to work with C99 compiler.

Revision 1.41 / (download) - annotate - [select for diffs], Mon Apr 30 03:19:40 2012 UTC (9 years, 5 months ago) by jnemeth
Branch: MAIN
Changes since 1.40: +2 -3 lines
Diff to previous 1.40 (colored)

Update to Asterisk 1.6.2.24.  This fixes AST-2012-004 and AST-2012-005.
The 1.6.2 series went End of Life on April 21st 2012, so this was
the last update.  This package will be deleted in the not too
distnat future.

The Asterisk Development Team has announced security releases for
Asterisk 1.6.2 , 1.8, and 10. The available security releases are
released as versions 1.6.2.24, 1.8.11.1, and 10.3.1.

The release of Asterisk 1.6.2.24, 1.8.11.1, and 10.3.1 resolve the
following two issues:

 * A permission escalation vulnerability in Asterisk Manager
   Interface.  This would potentially allow remote authenticated
   users the ability to execute commands on the system shell with
   the privileges of the user running the Asterisk application.

 * A heap overflow vulnerability in the Skinny Channel driver.
   The keypad button message event failed to check the length of
   a fixed length buffer before appending a received digit to the
   end of that buffer.  A remote authenticated user could send
   sufficient keypad button message events that th e buffer would
   be overrun.

These issues and their resolution are described in the security
advisories.

For more information about the details of these vulnerabilities,
please read security advisories AST-2012-004, AST-2012-005, and
AST-2012-006, 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/asterisk/releases/ChangeLog-1.6.2.24

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2012-004.pdf
 * http://downloads.asterisk.org/pub/security/AST-2012-005.pdf

Thank you for your continued support of Asterisk!

Revision 1.40 / (download) - annotate - [select for diffs], Fri Apr 27 12:31:37 2012 UTC (9 years, 5 months ago) by obache
Branch: MAIN
Changes since 1.39: +2 -1 lines
Diff to previous 1.39 (colored)

Recursive bump from icu shlib major bumped to 49.

Revision 1.39 / (download) - annotate - [select for diffs], Sun Mar 25 02:59:53 2012 UTC (9 years, 6 months ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2012Q1-base, pkgsrc-2012Q1
Changes since 1.38: +2 -3 lines
Diff to previous 1.38 (colored)

Update to 1.6.2.23:

This is a security fix update.  It fixes AST-2012-002.

NOTE NOTE NOTE

This is likely to be the last update to this package.  This version
of Asterisk will be EOLed on April 21st, 2012.  It will probably
be removed from pkgsrc not long after that.  If you are still using
this package, you should consider switching to comms/asterisk18,
the Long Term Support version, or comms/asterisk10 in the near
future.

NOTE NOTE NOTE

The Asterisk Development Team has announced security releases for
Asterisk 1.4, 1.6.2, 1.8, and 10. The available security releases
are released as versions 1.4.44, 1.6.2.23, 1.8.10.1, and 10.2.1.

The release of Asterisk 1.4.44 and 1.6.2.23 resolve an issue wherein
app_milliwatt can potentially overrun a buffer on the stack, causing
Asterisk to crash.  This does not have the potential for remote
code execution.

These issues and their resolution are described in the security
advisory.

For more information about the details of these vulnerabilities,
please read the security advisories AST-2012-002 and AST-2012-003,
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/asterisk/releases/ChangeLog-1.6.2.23

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2012-002.pdf

Thank you for your continued support of Asterisk!

Revision 1.38 / (download) - annotate - [select for diffs], Thu Feb 16 16:30:03 2012 UTC (9 years, 8 months ago) by hans
Branch: MAIN
Changes since 1.37: +5 -1 lines
Diff to previous 1.37 (colored)

Fix build on SunOS.

Revision 1.37 / (download) - annotate - [select for diffs], Tue Jan 17 02:12:52 2012 UTC (9 years, 9 months ago) by jnemeth
Branch: MAIN
Changes since 1.36: +2 -1 lines
Diff to previous 1.36 (colored)

PR/35369 -- David Wetzel -- add support for speex codec (enabled by default)

Revision 1.36 / (download) - annotate - [select for diffs], Sat Jan 14 08:30:15 2012 UTC (9 years, 9 months ago) by jnemeth
Branch: MAIN
Changes since 1.35: +2 -2 lines
Diff to previous 1.35 (colored)

Update to Asterisk 1.6.2.22:

The release of Asterisk 1.6.2.22 corrects two flaws in sip.conf.sample
related to AST-2011-013:

* The sample file listed *two* values for the 'nat' option as being the default.
   Only 'yes' is the default.

* The warning about having differing 'nat' settings confusingly referred to both
   peers and users.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.22

Thank you for your continued support of Asterisk!

Revision 1.35 / (download) - annotate - [select for diffs], Mon Dec 12 05:05:33 2011 UTC (9 years, 10 months ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2011Q4-base, pkgsrc-2011Q4
Changes since 1.34: +2 -3 lines
Diff to previous 1.34 (colored)

This update fixes AST-2011-013 and AST-2011-014.  It also adapts to changes
in the iLBC codec files.

     __________________________________________________________________

               Asterisk Project Security Advisory - AST-2011-013

         Product        Asterisk
         Summary        Possible remote enumeration of SIP endpoints with
                        differing NAT settings
    Nature of Advisory  Unauthorized data disclosure
      Susceptibility    Remote unauthenticated sessions
         Severity       Minor
      Exploits Known    Yes
       Reported On      2011-07-18
       Reported By      Ben Williams
        Posted On
     Last Updated On    December 7, 2011
     Advisory Contact   Terry Wilson <twilson at digium.com>

         CVE Name

    Description  It is possible to enumerate SIP usernames when the general
                 and user/peer NAT settings differ in whether to respond to
                 the port a request is sent from or the port listed for
                 responses in the Via header. In 1.4 and 1.6.2, this would
                 mean if one setting was nat=yes or nat=route and the other
                 was either nat=no or nat=never. In 1.8 and 10, this would
                 mean when one was nat=force_rport or nat=yes and the other
                 was nat=no or nat=comedia.

    Resolution  Handling NAT for SIP over UDP requires the differing
                behavior introduced by these options.

                To lessen the frequency of unintended username disclosure,
                the default NAT setting was changed to always respond to the
                port from which we received the request-the most commonly
                used option.

                Warnings were added on startup to inform administrators of
                the risks of having a SIP peer configured with a different
                setting than that of the general setting. The documentation
                now strongly suggests that peers are no longer configured
                for NAT individually, but through the global setting in the
                "general" context.

                               Affected Versions
                Product              Release Series
         Asterisk Open Source             All        All versions

                                  Corrected In
     As this is more of an issue with SIP over UDP in general, there is no
     fix supplied other than documentation on how to avoid the problem. The
        default NAT setting has been changed to what we believe the most
      commonly used setting for the respective version in Asterisk 1.4.43,
                             1.6.2.21, and 1.8.7.2.

            Links

    Asterisk Project Security Advisories are posted at
    http://www.asterisk.org/security

    This document may be superseded by later versions; if so, the latest
    version will be posted at
    http://downloads.digium.com/pub/security/AST-2011-013.pdf and
    http://downloads.digium.com/pub/security/AST-2011-013.html

                                Revision History
           Date                 Editor                 Revisions Made

               Asterisk Project Security Advisory - AST-2011-013
              Copyright (c) 2011 Digium, Inc. All Rights Reserved.
  Permission is hereby granted to distribute and publish this advisory in its
                           original, unaltered form.

     __________________________________________________________________

               Asterisk Project Security Advisory - AST-2011-014

         Product        Asterisk
         Summary        Remote crash possibility with SIP and the "automon"
                        feature enabled
    Nature of Advisory  Remote crash vulnerability in a feature that is
                        disabled by default
      Susceptibility    Remote unauthenticated sessions
         Severity       Moderate
      Exploits Known    Yes
       Reported On      November 2, 2011
       Reported By      Kristijan Vrban
        Posted On       2011-11-03
     Last Updated On    December 7, 2011
     Advisory Contact   Terry Wilson <twilson at digium.com>

         CVE Name

    Description  When the "automon" feature is enabled in features.conf, it
                 is possible to send a sequence of SIP requests that cause
                 Asterisk to dereference a NULL pointer and crash.

    Resolution  Applying the referenced patches that check that the pointer
                is not NULL before accessing it will resolve the issue. The
                "automon" feature can be disabled in features.conf as a
                workaround.

                               Affected Versions
                Product              Release Series
         Asterisk Open Source           1.6.2.x      All versions
         Asterisk Open Source            1.8.x       All versions

                                  Corrected In
                   Product                              Release
            Asterisk Open Source                   1.6.2.21, 1.8.7.2

                                     Patches
                              Download URL                            Revision
   http://downloads.asterisk.org/pub/security/AST-2011-014-1.6.2.diff 1.6.2.20
   http://downloads.asterisk.org/pub/security/AST-2011-014-1.8.diff   1.8.7.1

            Links

    Asterisk Project Security Advisories are posted at
    http://www.asterisk.org/security

    This document may be superseded by later versions; if so, the latest
    version will be posted at
    http://downloads.digium.com/pub/security/AST-2011-014.pdf and
    http://downloads.digium.com/pub/security/AST-2011-014.html

                                Revision History
           Date                 Editor                 Revisions Made

               Asterisk Project Security Advisory - AST-2011-014
              Copyright (c) 2011 Digium, Inc. All Rights Reserved.
  Permission is hereby granted to distribute and publish this advisory in its
                           original, unaltered form.

Revision 1.34 / (download) - annotate - [select for diffs], Mon Dec 5 04:18:32 2011 UTC (9 years, 10 months ago) by jnemeth
Branch: MAIN
Changes since 1.33: +3 -2 lines
Diff to previous 1.33 (colored)

Now that -current has sqlite3 included in base, enable it here.

Revision 1.33 / (download) - annotate - [select for diffs], Sun Aug 7 02:40:32 2011 UTC (10 years, 2 months ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2011Q3-base, pkgsrc-2011Q3
Changes since 1.32: +2 -1 lines
Diff to previous 1.32 (colored)

Bump PKGREVISION for perl update.

Revision 1.32 / (download) - annotate - [select for diffs], Tue Jul 5 08:34:47 2011 UTC (10 years, 3 months ago) by jnemeth
Branch: MAIN
CVS Tags: pkgsrc-2011Q2-base, pkgsrc-2011Q2
Changes since 1.31: +2 -2 lines
Diff to previous 1.31 (colored)

Update to 1.6.2.19 (fixes several security issues):

Please note that Asterisk 1.6.2.19 is the final maintenance release
from the 1.6.2 branch. Support for security related issues will
continue until April 21, 2012. For more information about support
of the various Asterisk branches, see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

The release of Asterisk 1.6.2.19 resolves several issues reported
by the community and would have not been possible without your
participation.  Thank you!

The following is a sample of the issues resolved in this release:

* Don't broadcast FullyBooted to every AMI connection
   The FullyBooted event should not be sent to every AMI connection
   every time someone connects via AMI. It should only be sent to
   the user who just connected.
   (Closes issue #18168. Reported, patched by FeyFre)
* Fix thread blocking issue in the sip TCP/TLS implementation.
   (Closes issue #18497. Reported by vois. Tested by vois, rossbeer, kowalma,
   Freddi_Fonet. Patched by dvossel)
* Don't delay DTMF in core bridge while listening for DTMF features.
   (Closes issue #15642, #16625. Reported by jasonshugart, sharvanek. Tested by
   globalnetinc, jde. Patched by oej, twilson)
* Fix chan_local crashs in local_fixup()
   Thanks OEJ for tracking down the issue and submitting the patch.
   (Closes issue #19053. Reported, patched by oej)
* Don't offer video to directmedia callee unless caller offered it as well
   (Closes issue #19195. Reported, patched by one47)

Additionally security announcements AST-2011-008, AST-2011-010, and
AST-2011-011 have been resolved in this release.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.19

Revision 1.31 / (download) - annotate - [select for diffs], Mon Jun 6 06:25:05 2011 UTC (10 years, 4 months ago) by jnemeth
Branch: MAIN

Upgrade to 1.6.2.18.  This fixes several security issues including:
AST-2011-002, AST-2011-003, AST-2011-004, AST-2011-005, and AST-2011-006.

===========================================================================
1.6.2.18:

The Asterisk Development Team has announced the release of Asterisk 1.6.2.18.

The release of Asterisk 1.6.2.18 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following is a sample of the issues resolved in this release:

 * Only offer codecs both sides support for directmedia.

 * Resolution of several DTMF based attended transfer issues.
   NOTE: Be sure to read the ChangeLog for more information about these changes.

 * Resolve deadlocks related to device states in chan_sip

 * Fix channel redirect out of MeetMe() and other issues with channel softhangup

 * Fix voicemail sequencing for file based storage.

 * Guard against retransmitting BYEs indefinitely during attended transfers with
   chan_sip.

In addition to the changes listed above, commits to resolve security issues
AST-2011-005 and AST-2011-006 have been merged into this release. More
information about AST-2011-005 and AST-2011-006 can be found at:

http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.18

===========================================================================
1.6.2.17.3

The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3.

The releases of Asterisk 1.4.40.1, 1.6.1.25, 1.6.2.17.3, and 1.8.3.3 resolve two
issues:

* File Descriptor Resource Exhaustion (AST-2011-005)
* Asterisk Manager User Shell Access (AST-2011-006)

The issues and resolutions are described in the AST-2011-005 and AST-2011-006
security advisories.

For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-005 and AST-2011-006, which were released at the
same time as this announcement.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.3

Security advisory AST-2011-005 and AST-2011-006 are available at:

http://downloads.asterisk.org/pub/security/AST-2011-005.pdf
http://downloads.asterisk.org/pub/security/AST-2011-006.pdf

===========================================================================
1.6.2.17.2:

The Asterisk Development Team has announced security releases for Asterisk
branches 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.6.1.24, 1.6.2.17.2, and 1.8.3.2.

** This is a re-release of Asterisk 1.6.1.23, 1.6.2.17.1 and 1.8.3.1 which
    contained a bug which caused duplicate manager entries (issue #18987).

The releases of Asterisk 1.6.1.24, 1.6.2.17.2, and 1.8.3.2 resolve two issues:

  * Resource exhaustion in Asterisk Manager Interface (AST-2011-003)
  * Remote crash vulnerability in TCP/TLS server (AST-2011-004)

The issues and resolutions are described in the AST-2011-003 and AST-2011-004
security advisories.

For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-003 and AST-2011-004, which were released at the
same time as this announcement.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.2

Security advisory AST-2011-003 and AST-2011-004 are available at:

http://downloads.asterisk.org/pub/security/AST-2011-003.pdf
http://downloads.asterisk.org/pub/security/AST-2011-004.pdf

===========================================================================
1.6.2.17.1:

The Asterisk Development Team has announced security releases for Asterisk
branches 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.6.1.23, 1.6.2.17.1, and 1.8.3.1.

These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

The releases of Asterisk 1.6.1.23, 1.6.2.17.1, and 1.8.3.1 resolve two issues:

  * Resource exhaustion in Asterisk Manager Interface (AST-2011-003)
  * Remote crash vulnerability in TCP/TLS server (AST-2011-004)

The issues and resolutions are described in the AST-2011-003 and AST-2011-004
security advisories.

For more information about the details of these vulnerabilities, please read the
security advisories AST-2011-003 and AST-2011-004, which were released at the
same time as this announcement.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.17.1

Security advisory AST-2011-003 and AST-2011-004 are available at:

http://downloads.asterisk.org/pub/security/AST-2011-003.pdf
http://downloads.asterisk.org/pub/security/AST-2011-004.pdf

===========================================================================
1.6.2.17:

The Asterisk Development Team has announced the release of Asterisk 1.6.2.17.

The release of Asterisk 1.6.2.17 resolves several issues reported by the
community and would have not been possible without your participation.

The following is a sample of the issues resolved in this release:

* Resolve duplicated data in the AstDB when using DIALGROUP()

* Correct issue where res_config_odbc could populate fields with invalid data.

* When using cdr_pgsql the billsec field was not populated correctly on
   unanswered calls.

* Resolve issue where re-transmissions of SUBSCRIBE could break presence.

* Fix regression causing forwarding voicemails to not work with file storage.

* This version of Asterisk includes the new Compiler Flags option
   BETTER_BACKTRACES which uses libbfd to search for better symbol information
   within both the Asterisk binary, as well as loaded modules, to assist when
   using inline backtraces to track down problems.

* Resolve several issues with DTMF based attended transfers.
   NOTE: Be sure to read the ChangeLog for more information about these changes.

* Resolve issue where no Music On Hold may be triggered when using
   res_timing_dahdi.

* Fix regression that changed behavior of queues when ringing a queue member.

Additionally, this release has the changes related to security bulletin
AST-2011-002 which can be found at
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.17

===========================================================================
1.6.2.16.2:

The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.

The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an
issue that when decoding UDPTL packets, multiple stack and heap based arrays can
be made to overflow by specially crafted packets. Systems configured for
T.38 pass through or termination are vulnerable. The issue and resolution are
described in the AST-2011-002 security advisory.

For more information about the details of this vulnerability, please read the
security advisory AST-2011-002, which was released at the same time as this
announcement.

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.16.2

Security advisory AST-2011-002 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

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