Commit graph

2111 commits

Author SHA1 Message Date
he
c65ebb132e Don't enable the inet6 option on the various BSDs, since their stack
require separate inet6 and inet sockets, and conserver as of 8.2.1
doesn't do that.
Bump PKGREVISION.
2017-02-10 10:35:06 +00:00
wiz
7ac05101c6 Recursive bump for harfbuzz's new graphite2 dependency. 2017-02-06 13:54:36 +00:00
agc
30b55df38e Convert all occurrences (353 by my count) of
MASTER_SITES= 	site1 \
			site2

style continuation lines to be simple repeated

	MASTER_SITES+= site1
	MASTER_SITES+= site2

lines. As previewed on tech-pkg. With thanks to rillig for fixing pkglint
accordingly.
2017-01-19 18:52:01 +00:00
he
3f5ef9eb0e Add two patches so that this at least semi-works when the inet6
option is used:

 * Use correct sockaddr length when doing getnameinfo() for inet6,
   so we avoid an early return with "permanent failure" from getnameinfo()
 * Use temp variables for walking the address lists so that we avoid trying
   freeaddrinfo(NULL) and getting SEGV

This still isn't fully baked and backward compatible: with the
inet6 option turned on, on NetBSD the conserver process only opens
an inet6 server socket and no longer serves an inet socket (a
Linuxism, I suspect), making it troublesome to interoperate with
older versions of conserver or installations on hosts without IPv6
connectivity.

PKGREVISION bumped.
2017-01-18 09:54:51 +00:00
adam
76632718ac Revbump after boost update 2017-01-01 16:05:55 +00:00
wiz
7f84153239 Add python-3.6 to incompatible versions. 2017-01-01 14:43:22 +00:00
wiz
7135fcadcc Revert "Specify readline requirement on 30 packages"
Many of these definitely do not depend on readline.
So there must be a different underlying problem, and that
should be tracked down instead of papering over it.
2016-12-12 14:22:01 +00:00
jnemeth
4abf01490b Update to Asterisk 11.25.1: this fixes AST-2016-009.
Asterisk Project Security Advisory - ASTERISK-2016-009

         Product        Asterisk
         Summary
    Nature of Advisory  Authentication Bypass
      Susceptibility    Remote unauthenticated sessions
         Severity       Minor
      Exploits Known    No
       Reported On      October 3, 2016
       Reported By      Walter Doekes
        Posted On
     Last Updated On    December 8, 2016
     Advisory Contact   Mmichelson AT digium DOT com
         CVE Name

    Description  The chan_sip channel driver has a liberal definition for
                 whitespace when attempting to strip the content between a
                 SIP header name and a colon character. Rather than
                 following RFC 3261 and stripping only spaces and horizontal
                 tabs, Asterisk treats any non-printable ASCII character as
                 if it were whitespace. This means that headers such as

                 Contact\x01:

                 will be seen as a valid Contact header.

                 This mostly does not pose a problem until Asterisk is
                 placed in tandem with an authenticating SIP proxy. In such
                 a case, a crafty combination of valid and invalid To
                 headers can cause a proxy to allow an INVITE request into
                 Asterisk without authentication since it believes the
                 request is an in-dialog request. However, because of the
                 bug described above, the request will look like an
                 out-of-dialog request to Asterisk. Asterisk will then
                 process the request as a new call. The result is that
                 Asterisk can process calls from unvetted sources without
                 any authentication.

                 If you do not use a proxy for authentication, then this
                 issue does not affect you.

                 If your proxy is dialog-aware (meaning that the proxy keeps
                 track of what dialogs are currently valid), then this issue
                 does not affect you.

                 If you use chan_pjsip instead of chan_sip, then this issue
l
                 does not affect you.

    Resolution  chan_sip has been patched to only treat spaces and
                horizontal tabs as whitespace following a header name. This
                allows for Asterisk and authenticating proxies to view
                requests the same way

                               Affected Versions
                         Product                       Release
                                                       Series
                  Asterisk Open Source                  11.x    All Releases
                  Asterisk Open Source                  13.x    All Releases
                  Asterisk Open Source                  14.x    All Releases
                   Certified Asterisk                   13.8    All Releases


                                  Corrected In
          Product                              Release
    Asterisk Open Source               11.25.1, 13.13.1, 14.2.1
     Certified Asterisk                11.6-cert16, 13.8-cert4

                                    Patches
                 SVN URL                              Revision

           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/ASTERISK-2016-009.pdf and
    http://downloads.digium.com/pub/security/ASTERISK-2016-009.html

                                Revision History
                     Date                        Editor      Revisions Made
    November 28, 2016                        Mark Michelson  Initial writeup

             Asterisk Project Security Advisory - ASTERISK-2016-009
              Copyright (c) 2016 Digium, Inc. All Rights Reserved.
  Permission is hereby granted to distribute and publish this advisory in its
                           original, unaltered form.
2016-12-11 00:50:15 +00:00
leot
1e4e66e7c0 Update comms/py-gammu to py-gammu-2.7
Changes:
2.7
===
* Needs Gammu >= 1.37.90 due to API changes.

2.6
===
* Fixed error when creating new contact.
* Fixed possible testsuite errors.
2016-12-09 14:57:06 +00:00
leot
6c8b1cfa28 Update comms/gammu to gammu-1.37.91
Changes:
20161023 - 1.37.91

[!] * Changed version of the shared library.
[-] * Improved support for ZTE MF100.
[-] * Ignore unsolicited +CLCC: reply.
[-] * Correctly report when some SMSD SQL backend is not compiled in.
[-] * Fix build of MySQL backend on Linux.

20161018 - 1.37.90

[-] * Improved support Huawei K3770.
[!] * API changes in some parameter types.
[-] * Fixed various Windows compilation issues.
[-] * Fixed several resource leaks.
[-] * Create outbox SMS atomically in FILES backend.
[!] * Removed getlocation command as we no longer fit into their usage policy.
[-] * Fixed call diverts on TP-LINK MA260.
[+] * Initial support for Oracle database.
[!] * Removed unused daemons, pbk and pbk_groups tables from the SMSD schema.
[+] * SMSD outbox entries now can have priority set in the database.
[+] * Added SIM IMSI to the SMSD status table.
[+] * Added CheckNetwork directive.
[+] * SMSD attempts to power on radio if disabled.
[-] * Fixed processing of AT unsolicited responses in some cases.
[-] * Fixed parsing USSD responses from some devices.

20160816 - 1.37.4

[-] * Improved support for Huawei E3131.
[-] * Fixed SMS support for MULTIBAND 900E.
[-] * Fixed SMS created in text mode.

20160524 - 1.37.3

[-] * Improved support for Huawei E398.
[-] * Improved support for Huawei/Vodafone K4505.
[-] * Fixed possible crash if SMSD used in library.
[-] * Improved support for Huawei E180.

20160413 - 1.37.2

[-] * Fixed compilation of SMSD.

20160413 - 1.37.1

[-] * Properly report errors in HEX encoded strings from SMSD SQL backends.
[-] * Configurable SMSD table names.
[-] * Improved support for Huawei E303.
[-] * Improved support for Vodafone K4511.
[-] * Improved support for Telit M2M modules.
2016-12-09 14:56:34 +00:00
ryoon
36ed025474 Recursive revbump from textproc/icu 58.1 2016-12-04 05:17:03 +00:00
marino
938dfe006b Specify readline requirement on 30 packages
Solves:
/usr/libexec/binutils225/elf/ld.gold: error: cannot find -lreadline

The missing specification is obvious on DragonFly because there's
no publically accessible version of readline in base.
2016-12-04 03:51:14 +00:00
sevan
8222a619bb Correct the if statement to AND, not OR.
Unbreak builds on FreeBSD & DragonFly BSD
2016-12-03 13:02:22 +00:00
sevan
3dc96d292c Add dfu-util. 2016-12-03 03:32:35 +00:00
sevan
7043fd9af6 Import dfu-util 0.9
ok wiedi
2016-12-03 03:26:07 +00:00
jnemeth
133aa2c812 Update to Asterisk 14.2.0: this is mostly a bugfix release with some minor
improvements.

pkgsrc change: adapt to new res_resolver_unbound module.

The Asterisk Development Team has announced the release of Asterisk 14.2.0.

The release of Asterisk 14.2.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Improvements made in this release:
-----------------------------------
 * ASTERISK-26558 - app_queue: add variable to know if the call is
      not answered after a queue (Reported by scgm11)
 * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
      (Reported by scgm11)
 * ASTERISK-26538 - codec_opus: Add sample to
      configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
 * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and
      'ari set debug' CLI commands (Reported by Matt Jordan)
 * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by
      blacklisting host subnets that are not involved in RTP (Reported
      by Michael Walton)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported
      by snuffy)
 * ASTERISK-26520 - codec_opus: Generated fmtp line has no content
      (Reported by scgm11)
 * ASTERISK-26605 - codec_opus: Spammed warning when Opus
      negotiated but codec_opus not loaded. (Reported by Richard
      Mudgett)
 * ASTERISK-26516 - pjsip: Memory corruption with possible memory
      leak. (Reported by Richard Mudgett)
 * ASTERISK-26556 - manager: AMI version report same in Ast 13 &
      14, despite Ast 14 syntax changes (Reported by Michelle Dupuis)
 * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
      manipulation through agi (Reported by Morten Tryfoss)
 * ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and
      makes asterisk CLI read garbage (Reported by George Joseph)
 * ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold
      temporarily locks up set (Reported by Jason)
 * ASTERISK-26575 - testsuite: Need to check PJSIP functionality
      when res_srtp is not loaded. (Reported by Joshua Colp)
 * ASTERISK-26571 - res_pjsip: Resolution incorrect when explicit
      IPv6 transport configured (Reported by Joshua Colp)
 * ASTERISK-26468 - ari: Bridge events stop working after this
      sequence of ARI calls (Reported by Daniele Pallastrelli)
 * ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by
      Dmitry Melekhov)
 * ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11
      regressions (Reported by Matt Jordan)
 * ASTERISK-26412 - build:  Prepare for gcc 6.2 (Reported by George
      Joseph)
 * ASTERISK-26509 - A few non-critical deprecation warnings when
      building on Ubuntu 16.10 (Reported by Jonathan Harris)
 * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
      calls after 2 minutes - rtptimeout behaving badly - regression
      (Reported by Michael Keuter)
 * ASTERISK-26549 - app_dial: When PickupChan() is used some
      channels may have incorrect device state (Reported by Joshua
      Colp)
 * ASTERISK-24274 - [patch]Codec Format Is Not Included in the SDP
      Media Attributes When SLIN48 Codec Is Used (Reported by Frankie
      Chin)
 * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic
      payload types. (Reported by Alexander Traud)
 * ASTERISK-26506 - [patch]res_pjsip_outbound_publish: Crash when
      publishing, in publisher_client_send at
      res_pjsip_outbound_publish.c (Reported by Matt Krokosz)
 * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele
      Giacone)
 * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound
      reg. retry 403" in "sip show settings" (Reported by Sergey
      Grachev)
 * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats
      to maximum (Reported by Joshua Colp)
 * ASTERISK-26537 - AMI: NewConnectedLine event is not documented
      (Reported by Etienne Lessard)
 * ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as
      argument 2 to memcpy (Reported by Badalian Vyacheslav)
 * ASTERISK-26524 - astobj2: data_size variable is wasted space
      when AO2_DEBUG is not enabled. (Reported by Corey Farrell)
 * ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian
      Gilmour)
 * ASTERISK-26387 - Asterisk segfaults shortly after starting even
      with no active calls.  (Reported by Harley Peters)
 * ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing
      enough to be a nuisance (Reported by Joshua Colp)
 * ASTERISK-26514 - Super Awesome Company: Don't specify transport
      in pjsip.conf (Reported by Rusty Newton)
 * ASTERISK-26510 - pjproject_bundled uses the --strip-components
      option of tar which isn't supported in older versions (Reported
      by George Joseph)
 * ASTERISK-22480 - Embedded pjproject: build.mak contains
      hardcoded full path to version.mak (Reported by Matt Jordan)
 * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change
      (Reported by Bill Brigden)
 * ASTERISK-26503 - app_voicemail: Asterisk crashes when
      MailboxExists is used (Reported by Doug Lytle)
 * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can
      cause audio loss and wonkiness (Reported by Andreas Wetzel)
 * ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack)
      installations. (Reported by Alexander Traud)
 * ASTERISK-26482 - [patch] chan_pjsip: segfault on already
      disconnected session (Reported by Alexei Gradinari)
 * ASTERISK-26421 - Segmentation Fault with ARI originate into
      mixing bridge with 43 clients (Reported by Andrew Nagy)
 * ASTERISK-26444 - 'features show' command in CLI does not return
      prompt. (Reported by John Kiniston)
 * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes
      File not Module (Reported by Alexander Traud)
 * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by
      Tzafrir Cohen)
 * ASTERISK-26462 - [patch] app_queue: While using queues with
      realtime, setting back to an empty context doesn't stop the exit
      key usage (Reported by Leandro Dardini)
 * ASTERISK-26439 - chan_rtp: Crash when originating (Reported by
      Kayode)
 * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT
      detection triggered. (Reported by Alexander Traud)
 * ASTERISK-26618 - build: Backport addition of librt check to
      configure.ac (Reported by Kevin Harwell)

New Features made in this release:
-----------------------------------
 * ASTERISK-26595 - ARI: Add the ability to control the source of
      video in a multi-party mixing bridge (Reported by Matt Jordan)
 * ASTERISK-26492 - ARI: Add ability to specify channel variables
      on websocket events (Reported by Mark Michelson)
 * ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing
      events (Reported by Matt Jordan)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.2.0

Thank you for your continued support of Asterisk!
2016-11-27 22:55:51 +00:00
jnemeth
046d73f90a Update to Asterisk 13.13.0: this is mainly a bug fix release with some
minor improvements.

The Asterisk Development Team has announced the release of Asterisk 13.13.0.

The release of Asterisk 13.13.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-26595 - ARI: Add the ability to control the source of
      video in a multi-party mixing bridge (Reported by Matt Jordan)
 * ASTERISK-26470 - ARI: Add an 'asterisk_id' field to outgoing
      events (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26608 - Compile and link failures on OpenBSD (Reported
      by snuffy)
 * ASTERISK-26343 - ASTERISK-25951 causes issues for callerid
      manipulation through agi (Reported by Morten Tryfoss)
 * ASTERISK-26520 - codec_opus: Generated fmtp line has no content
      (Reported by scgm11)
 * ASTERISK-26605 - codec_opus: Spammed warning when Opus
      negotiated but codec_opus not loaded. (Reported by Richard
      Mudgett)
 * ASTERISK-26516 - pjsip: Memory corruption with possible memory
      leak. (Reported by Richard Mudgett)
 * ASTERISK-26592 - Latest libedit (3.1) defaults to unicode and
      makes asterisk CLI read garbage (Reported by George Joseph)
 * ASTERISK-26565 - chan_unistim on 11, 13, 14 placing call on hold
      temporarily locks up set (Reported by Jason)
 * ASTERISK-26575 - testsuite: Need to check PJSIP functionality
      when res_srtp is not loaded. (Reported by Joshua Colp)
 * ASTERISK-24400 - ooh323 sends wrong hangup code (Reported by
      Dmitry Melekhov)
 * ASTERISK-26555 - Multi-party Video: Fix some post Asterisk-11
      regressions (Reported by Matt Jordan)
 * ASTERISK-26412 - build:  Prepare for gcc 6.2 (Reported by George
      Joseph)
 * ASTERISK-26509 - A few non-critical deprecation warnings when
      building on Ubuntu 16.10 (Reported by Jonathan Harris)
 * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
      calls after 2 minutes - rtptimeout behaving badly - regression
      (Reported by Michael Keuter)
 * ASTERISK-26468 - ari: Bridge events stop working after this
      sequence of ARI calls (Reported by Daniele Pallastrelli)
 * ASTERISK-26311 - [patch] rtp_engine: Allow more than 32 dynamic
      payload types. (Reported by Alexander Traud)
 * ASTERISK-26549 - app_dial: When PickupChan() is used some
      channels may have incorrect device state (Reported by Joshua
      Colp)
 * ASTERISK-26541 - res_pjsip_sdp_rtp: Restrict number of formats
      to maximum (Reported by Joshua Colp)
 * ASTERISK-25070 - Fix FTBFS on Hurd (Reported by Gabriele
      Giacone)
 * ASTERISK-26476 - chan_sip: Incorrect display option "Outbound
      reg. retry 403" in "sip show settings" (Reported by Sergey
      Grachev)
 * ASTERISK-26537 - AMI: NewConnectedLine event is not documented
      (Reported by Etienne Lessard)
 * ASTERISK-26526 - [UBSAN] vector.h: null pointer can be passed as
      argument 2 to memcpy (Reported by Badalian Vyacheslav)
 * ASTERISK-26524 - astobj2: data_size variable is wasted space
      when AO2_DEBUG is not enabled. (Reported by Corey Farrell)
 * ASTERISK-26344 - Asterisk 13.11.0 + PJSIP crash (Reported by Ian
      Gilmour)
 * ASTERISK-26387 - Asterisk segfaults shortly after starting even
      with no active calls.  (Reported by Harley Peters)
 * ASTERISK-26514 - Super Awesome Company: Don't specify transport
      in pjsip.conf (Reported by Rusty Newton)
 * ASTERISK-26513 - tests/channels/pjsip/qualify/auth: Crashing
      enough to be a nuisance (Reported by Joshua Colp)
 * ASTERISK-26510 - pjproject_bundled uses the --strip-components
      option of tar which isn't supported in older versions (Reported
      by George Joseph)
 * ASTERISK-22480 - Embedded pjproject: build.mak contains
      hardcoded full path to version.mak (Reported by Matt Jordan)
 * ASTERISK-26307 - res_pjsip_caller_id: Crash on outgoing change
      (Reported by Bill Brigden)
 * ASTERISK-26503 - app_voicemail: Asterisk crashes when
      MailboxExists is used (Reported by Doug Lytle)
 * ASTERISK-26423 - res_pjsip_sdp_rtp: Asymmetric RTP codec can
      cause audio loss and wonkiness (Reported by Andreas Wetzel)
 * ASTERISK-26309 - [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack)
      installations. (Reported by Alexander Traud)
 * ASTERISK-26421 - Segmentation Fault with ARI originate into
      mixing bridge with 43 clients (Reported by Andrew Nagy)
 * ASTERISK-26444 - 'features show' command in CLI does not return
      prompt. (Reported by John Kiniston)
 * ASTERISK-26482 - [patch] chan_pjsip: segfault on already
      disconnected session (Reported by Alexei Gradinari)
 * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes
      File not Module (Reported by Alexander Traud)
 * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by
      Tzafrir Cohen)
 * ASTERISK-26439 - chan_rtp: Crash when originating (Reported by
      Kayode)
 * ASTERISK-26462 - [patch] app_queue: While using queues with
      realtime, setting back to an empty context doesn't stop the exit
      key usage (Reported by Leandro Dardini)
 * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT
      detection triggered. (Reported by Alexander Traud)
 * ASTERISK-26618 - build: Backport addition of librt check to
      configure.ac (Reported by Kevin Harwell)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25063 - [patch]add X.509 subject alternative name
      support to Asterisk TLS support (Reported by Maciej Szmigiero)
 * ASTERISK-26558 - app_queue: add variable to know if the call is
      not answered after a queue (Reported by scgm11)
 * ASTERISK-26176 - chan_sip: Add AccountCode to AMI PeerEntry
      (Reported by scgm11)
 * ASTERISK-26538 - codec_opus: Add sample to
      configs/samples/codecs.conf.sample (Reported by Kevin Harwell)
 * ASTERISK-26488 - ARI: Add 'ari show app', 'ari show apps', and
      'ari set debug' CLI commands (Reported by Matt Jordan)
 * ASTERISK-26418 - res_rtp_asterisk: Speed up ICE resolution by
      blacklisting host subnets that are not involved in RTP (Reported
      by Michael Walton)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.13.0

Thank you for your continued support of Asterisk!
2016-11-27 08:48:18 +00:00
jnemeth
f2c309ff70 Update to Asterisk 11.25.0: this is a bug fix release.
The Asterisk Development Team has announced the release of Asterisk 11.25.0.

The release of Asterisk 11.25.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26503 - app_voicemail: Asterisk crashes when
      MailboxExists is used (Reported by Doug Lytle)
 * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes
      File not Module (Reported by Alexander Traud)
 * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by
      Tzafrir Cohen)
 * ASTERISK-26462 - [patch] app_queue: While using queues with
      realtime, setting back to an empty context doesn't stop the exit
      key usage (Reported by Leandro Dardini)
 * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT
      detection triggered. (Reported by Alexander Traud)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0

Thank you for your continued support of Asterisk!
2016-11-27 04:42:26 +00:00
mef
bc244d8876 Update doxygen-depend version to 1.8.12 (or add new BUILD_DEPENDS+) 2016-11-24 14:11:31 +00:00
mef
734f59fb0d Adjust PLIST for doxygen update 1.8.11 to 1.8.12, PKGREVISION++. 2016-11-24 13:43:35 +00:00
jnemeth
c298c6c5aa Update to Asterisk 14.1.2: this is a critical bug fix release.
The Asterisk Development Team has announced the release of Asterisk 14.1.2.

The release of Asterisk 14.1.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
      calls after 2 minutes - rtptimeout behaving badly - regression
      (Reported by Michael Keuter)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.2

Thank you for your continued support of Asterisk!
2016-11-11 16:19:14 +00:00
jnemeth
d550cf80f2 Update the Asterisk 13.12.2: this is a critical bug fix release.
The Asterisk Development Team has announced the release of Asterisk 13.12.2.

The release of Asterisk 13.12.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26523 - chan_sip: Asterisk 13.12.1 disconnects incoming
      calls after 2 minutes - rtptimeout behaving badly - regression
      (Reported by Michael Keuter)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.2

Thank you for your continued support of Asterisk!
2016-11-11 15:44:16 +00:00
jnemeth
952e00ae39 Update to Asterisk 13.12.1: this is a critical bug fix release.
The Asterisk Development Team has announced the release of Asterisk 13.12.1.

The release of Asterisk 13.12.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26503 - app_voicemail: Asterisk crashes when
      MailboxExists is used (Reported by Doug Lytle)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.1

Thank you for your continued support of Asterisk!
2016-10-29 02:10:06 +00:00
jnemeth
8c07b60a63 Update to Asterisk 14.1.1: this is a critical bug fix release.
The Asterisk Development Team has announced the release of Asterisk 14.1.1.

The release of Asterisk 14.1.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26503 - app_voicemail: Asterisk crashes when
      MailboxExists is used (Reported by Doug Lytle)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.1

Thank you for your continued support of Asterisk!
2016-10-28 08:25:20 +00:00
jnemeth
620b1aca37 Update to Asterisk 11.24.1: this is a critical bug fix release.
The Asterisk Development Team has announced the release of Asterisk 11.24.1.

The release of Asterisk 11.24.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26503 - app_voicemail: Asterisk crashes when
      MailboxExists is used (Reported by Doug Lytle)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1

Thank you for your continued support of Asterisk!
2016-10-28 07:26:26 +00:00
jnemeth
97fb43d7db Update to Asterisk 14.1.0: this is mostly a bug fix release.
The Asterisk Development Team has announced the release of Asterisk 14.1.0.

The release of Asterisk 14.1.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-26277 - Add dialplan function
      PJSIP_SEND_SESSION_REFRESH that sends a session refresh to
      update formats on a channel after session establishment
      (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26477 - pjproject:  SEGV during SSL operations
      (Reported by George Joseph)
 * ASTERISK-17470 - [patch] - When videosupport=yes, asterisk
      allows one end peer to send video, even though the other end
      supports only audio. (Reported by effie mouzeli)
 * ASTERISK-26416 - pjproject-bundled: configure fails to check for
      all required utilities (Reported by Corey Farrell)
 * ASTERISK-26466 - core: Be forgiving on external callerid that
      may be flawed so we don't drop events (Reported by Richard
      Mudgett)
 * ASTERISK-26362 - res_config_mysql:  Broken after 13.10 (Reported
      by Carlos Chavez)
 * ASTERISK-26446 - app_dial:  There's no way to override the
      hangupcause on unanswered channels (Reported by George Joseph)
 * ASTERISK-26410 - core: Asterisk 14 doesn't show the header in
      the console or verbose when starting (Reported by Dan Jenkins)
 * ASTERISK-24311 - Populating database via Alembic fails when
      using same database for multiple schema sets (Reported by Dafi
      Ni)
 * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No
      Symmetric Response. (Reported by Alexander Traud)
 * ASTERISK-26426 - format_ogg_opus: remove from source (Reported
      by Kevin Harwell)
 * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
      bindaddr=[::] (Reported by Jacek)
 * ASTERISK-25468 - Deadlock in chan_sip - core show locks shows
      do_monitor lock (Reported by Barry Flanagan)
 * ASTERISK-26397 - manager: PresenceState action crashes Asterisk
      14 (Reported by Andrew Nagy)
 * ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by
      Joshua Colp)
 * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so
      instructed (Reported by Tzafrir Cohen)
 * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled
      (Reported by Anthony Messina)
 * ASTERISK-26391 - Consoles do not display verbose logger messages
      even when requested. (Reported by Marcelo Terres)
 * ASTERISK-26263 - SQL error when using realtime and registering
      extension / inserting into ps_contacts (Reported by Jeppe Ryskov
      Larsen)
 * ASTERISK-26365 - rtp: Offer with multiple payloads for same
      codec is incorrectly handled (Reported by Joshua Colp)
 * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
      by Aaron Hamstra)
 * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten
      for connectionful protocols (Reported by Joshua Colp)
 * ASTERISK-26367 - rtp: Timestamps broken when video frame is
      across multiple RTP packets (Reported by Joshua Colp)
 * ASTERISK-26375 - res_pjsip_transport_management: Log message
      states seconds, but time value is milliseconds (Reported by
      Joshua Colp)
 * ASTERISK-26364 - res_pjsip: Don't assume a request will have
      target addresses (Reported by Joshua Colp)
 * ASTERISK-26360 - app_queue: "queue show" output gets "failed to
      extend from 240 to 327" msgs. (Reported by Richard Mudgett)
 * ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes
      unexpected callerid (Reported by Kevin Harwell)
 * ASTERISK-26349 -  13.11.1 res_pjsip/pjsip_distributor.c: Request
      'REGISTER' failed (Reported by Dmitry Melekhov)
 * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets)
      (Reported by Etienne Lessard)
 * ASTERISK-26264 - res_pjsip: Crash when applying ACL from
      non-existent endpoint (Reported by nappsoft)
 * ASTERISK-26341 - ARI: Stopping a media playlist only stops the
      current media URI being played back, and not the whole list
      (Reported by Matt Jordan)
 * ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p'
      argument) is enabled and callee rejects a call or hangs up.
      (Reported by Etienne Lessard)
 * ASTERISK-26331 - Crash on "core show channeltype Surrogate" in
      ast_format_cap_get_names (Reported by CGI.NET)
 * ASTERISK-26269 - res_pjsip: Wrong state for aors without
      registered contacts after startup (Reported by nappsoft)
 * ASTERISK-26282 - AEL: macro-call in Dial application, macro
      "lacks 's' extension" (Reported by chris de rock)
 * ASTERISK-26226 - pbx: Asterisk crash on AMI action
      "ShowDialplan" when there's a circular dependency between
      contexts (Reported by Etienne Lessard)
 * ASTERISK-26299 - app_queue: Queue application sometimes stops
      calling members with Local interface (Reported by Etienne
      Lessard)
 * ASTERISK-26279 - pjproject-bundled:  Fails to compile on Debian
      6 (Reported by George Joseph)
 * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not
      cleaning up properly (Reported by Alexander Traud)
 * ASTERISK-26203 - res_fax: Deadlock when using
      FAXOPT(gateway)=yes with Local channels (Reported by Etienne
      Lessard)
 * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking
      inversion in T.38 query option with features bridging code
      (Reported by David Brillert)
 * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with
      local channels. (Reported by Richard Mudgett)
 * ASTERISK-26288 - followme: fails to reset config items to
      default values on reload (Reported by Tzafrir Cohen)
 * ASTERISK-22374 - Finish mapping the sip.conf parameters to
      res_sip.conf parameters (Reported by Matt Jordan)
 * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
      SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
      abelbeck)
 * ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include
      annexb=no attribute. (Reported by Ali Ghavidel)
 * ASTERISK-25472 - Swagger scripts are not replacing format
      variable in file brief (Reported by Corey Farrell)
 * ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but
      it's not mandatory to compile it (Reported by József Dudás)
 * ASTERISK-26305 - Asterisk 14: Two resolver unbound testsuite
      tests fail (Reported by Richard Mudgett)
 * ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities
      not detected in PJProject. (Reported by Alexander Traud)
 * ASTERISK-25492 - ARI: Path parameters are case sensitive
      (Reported by Joshua Colp)
 * ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via
      chan_pjsip (Reported by Ross Beer)
 * ASTERISK-26233 - pbx: Failure to remove inconsistent extension
      names (Reported by Corey Farrell)
 * ASTERISK-26246 - Security: Privilege escalation by AMI adding
      dialplan extensions. (Reported by Richard Mudgett)
 * ASTERISK-26267 - ast_register_atexit callbacks should be run on
      failed startup. (Reported by Corey Farrell)
 * ASTERISK-26241 - res_pjsip:  When using compact headers, rpid
      and pai are incorrectly generated (Reported by George Joseph)
 * ASTERISK-25797 - app_queue: Crash when calling a queue with a
      member with a forward to an nonexistent extension (Reported by
      Etienne Lessard)
 * ASTERISK-26239 - res_pjsip_logger:  An empty global/debug option
      is treated as a "match all" hostname (Reported by George Joseph)
 * ASTERISK-26238 - res_pjsip: Empty global default_from_user
      causes crash (Reported by Joshua Colp)
 * ASTERISK-26268 - alembic: 'auth_username' not in PJSIP
      'identify_by' enum (Reported by Joshua Colp)
 * ASTERISK-26253 - sdp_srtp: libsrtp now a required dependency,
      shouldn't be (Reported by Ben Merrills)
 * ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade +
      indicate (Reported by Ross Beer)
 * ASTERISK-26183 - alembic: error when using sqlalchemy version
      1.1.0b2 (Reported by Kevin Harwell)
 * ASTERISK-26283 - res_resolver_unbound:  fails configure on older
      Ubuntu and CentOS (Reported by George Joseph)
 * ASTERISK-26280 - DNS lookups can block channel media paths
      (Reported by Mark Michelson)
 * ASTERISK-26278 - asterisk.h should produce a reasonable error
      for external modules that fail to define AST_MODULE_SELF_SYM.
      (Reported by Corey Farrell)
 * ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a
      similar treatment for module unloading as
      res_pjsip_outbound_registration.c (Reported by Richard Mudgett)
 * ASTERISK-26265 - Errors ignored from some parts of system
      initialization. (Reported by Corey Farrell)
 * ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex
      for get all (Reported by Dmitry)
 * ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets
      with IP6 (Reported by Alexander Traud)
 * ASTERISK-25996 - Remove "live_dangerously" requirement on
      DB(read) (Reported by Andrew Nagy)
 * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1:
      "libasteriskpj.so: undefined reference to..." (Reported by Hans
      van Eijsden)
 * ASTERISK-26237 - Fax is detected on regular calls. (Reported by
      Richard Mudgett)

Improvements made in this release:
-----------------------------------
 * ASTERISK-26409 - codec_opus: Update Asterisk to support the
l
      translation codec. (Reported by Kevin Harwell)
 * ASTERISK-26289 - Announcer channels in ConfBridges cause
      inefficiencies (Reported by Mark Michelson)
 * ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let
      dialplan know what fax transport was used (Reported by Alexei
      Gradinari)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.1.0

Thank you for your continued support of Asterisk!
2016-10-27 06:43:39 +00:00
jnemeth
8f3acf29c1 Update to Asterisk 13.12.0: this is mostly a bug fix release.
The Asterisk Development Team has announced the release of Asterisk 13.12.0.

The release of Asterisk 13.12.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-26277 - Add dialplan function
      PJSIP_SEND_SESSION_REFRESH that sends a session refresh to
      update formats on a channel after session establishment
      (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26477 - pjproject:  SEGV during SSL operations
      (Reported by George Joseph)
 * ASTERISK-17470 - [patch] - When videosupport=yes, asterisk
      allows one end peer to send video, even though the other end
      supports only audio. (Reported by effie mouzeli)
 * ASTERISK-26416 - pjproject-bundled: configure fails to check for
      all required utilities (Reported by Corey Farrell)
 * ASTERISK-26466 - core: Be forgiving on external callerid that
      may be flawed so we don't drop events (Reported by Richard
      Mudgett)
 * ASTERISK-26362 - res_config_mysql:  Broken after 13.10 (Reported
      by Carlos Chavez)
 * ASTERISK-26446 - app_dial:  There's no way to override the
      hangupcause on unanswered channels (Reported by George Joseph)
 * ASTERISK-24311 - Populating database via Alembic fails when
      using same database for multiple schema sets (Reported by Dafi
      Ni)
 * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No
      Symmetric Response. (Reported by Alexander Traud)
 * ASTERISK-26426 - format_ogg_opus: remove from source (Reported
      by Kevin Harwell)
 * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
      bindaddr=[::] (Reported by Jacek)
 * ASTERISK-25468 - Deadlock in chan_sip - core show locks shows
      do_monitor lock (Reported by Barry Flanagan)
 * ASTERISK-26397 - manager: PresenceState action crashes Asterisk
      14 (Reported by Andrew Nagy)
 * ASTERISK-26389 - res_odbc: Clean up pooling options (Reported by
      Joshua Colp)
 * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so
      instructed (Reported by Tzafrir Cohen)
 * ASTERISK-26273 - core: Won't compile when LOW_MEMORY is enabled
      (Reported by Anthony Messina)
 * ASTERISK-26263 - SQL error when using realtime and registering
      extension / inserting into ps_contacts (Reported by Jeppe Ryskov
      Larsen)
 * ASTERISK-26374 - res_pjsip_multihomed: Contact port is rewritten
      for connectionful protocols (Reported by Joshua Colp)
 * ASTERISK-26367 - rtp: Timestamps broken when video frame is
      across multiple RTP packets (Reported by Joshua Colp)
 * ASTERISK-26375 - res_pjsip_transport_management: Log message
      states seconds, but time value is milliseconds (Reported by
      Joshua Colp)
 * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
      by Aaron Hamstra)
 * ASTERISK-26360 - app_queue: "queue show" output gets "failed to
      extend from 240 to 327" msgs. (Reported by Richard Mudgett)
 * ASTERISK-26316 - res_pjsip_callerid: Irregular URI causes
      unexpected callerid (Reported by Kevin Harwell)
 * ASTERISK-26349 -  13.11.1 res_pjsip/pjsip_distributor.c: Request
      'REGISTER' failed (Reported by Dmitry Melekhov)
 * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets)
      (Reported by Etienne Lessard)
 * ASTERISK-26264 - res_pjsip: Crash when applying ACL from
      non-existent endpoint (Reported by nappsoft)
 * ASTERISK-26288 - followme: fails to reset config items to
      default values on reload (Reported by Tzafrir Cohen)
 * ASTERISK-25691 - Crash occurs when screening mode (Dial's 'p'
      argument) is enabled and callee rejects a call or hangs up.
      (Reported by Etienne Lessard)
 * ASTERISK-26331 - Crash on "core show channeltype Surrogate" in
      ast_format_cap_get_names (Reported by CGI.NET)
 * ASTERISK-26282 - AEL: macro-call in Dial application, macro
      "lacks 's' extension" (Reported by chris de rock)
 * ASTERISK-26226 - pbx: Asterisk crash on AMI action
      "ShowDialplan" when there's a circular dependency between
      contexts (Reported by Etienne Lessard)
 * ASTERISK-26279 - pjproject-bundled:  Fails to compile on Debian
      6 (Reported by George Joseph)
 * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not
      cleaning up properly (Reported by Alexander Traud)
 * ASTERISK-26299 - app_queue: Queue application sometimes stops
      calling members with Local interface (Reported by Etienne
      Lessard)
 * ASTERISK-26203 - res_fax: Deadlock when using
      FAXOPT(gateway)=yes with Local channels (Reported by Etienne
      Lessard)
 * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking
      inversion in T.38 query option with features bridging code
      (Reported by David Brillert)
 * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with
      local channels. (Reported by Richard Mudgett)
 * ASTERISK-26269 - res_pjsip: Wrong state for aors without
      registered contacts after startup (Reported by nappsoft)
 * ASTERISK-22374 - Finish mapping the sip.conf parameters to
      res_sip.conf parameters (Reported by Matt Jordan)
 * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
      SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
      abelbeck)
 * ASTERISK-25472 - Swagger scripts are not replacing format
      variable in file brief (Reported by Corey Farrell)
 * ASTERISK-26228 - res_pjsip_sdp_rtp: G729A does not include
      annexb=no attribute. (Reported by Ali Ghavidel)
 * ASTERISK-25984 - res_odbc relies on res_odbc_transaction, but
      it's not mandatory to compile it (Reported by József Dudás)
 * ASTERISK-26303 - [patch] BuildSystem: ca_list_path capabilities
      not detected in PJProject. (Reported by Alexander Traud)
 * ASTERISK-25492 - ARI: Path parameters are case sensitive
      (Reported by Joshua Colp)
 * ASTERISK-26233 - pbx: Failure to remove inconsistent extension
      names (Reported by Corey Farrell)
 * ASTERISK-26164 - XMPP no longer triggers NOTIFY to device via
      chan_pjsip (Reported by Ross Beer)
 * ASTERISK-26246 - Security: Privilege escalation by AMI adding
      dialplan extensions. (Reported by Richard Mudgett)
 * ASTERISK-26267 - ast_register_atexit callbacks should be run on
      failed startup. (Reported by Corey Farrell)
 * ASTERISK-26241 - res_pjsip:  When using compact headers, rpid
      and pai are incorrectly generated (Reported by George Joseph)
 * ASTERISK-26239 - res_pjsip_logger:  An empty global/debug option
      is treated as a "match all" hostname (Reported by George Joseph)
 * ASTERISK-26238 - res_pjsip: Empty global default_from_user
      causes crash (Reported by Joshua Colp)
 * ASTERISK-25797 - app_queue: Crash when calling a queue with a
      member with a forward to an nonexistent extension (Reported by
      Etienne Lessard)
 * ASTERISK-26268 - alembic: 'auth_username' not in PJSIP
      'identify_by' enum (Reported by Joshua Colp)
 * ASTERISK-26145 - pjsip: Deadlock with suspend + masquerade +
      indicate (Reported by Ross Beer)
 * ASTERISK-26183 - alembic: error when using sqlalchemy version
      1.1.0b2 (Reported by Kevin Harwell)
 * ASTERISK-26280 - DNS lookups can block channel media paths
      (Reported by Mark Michelson)
 * ASTERISK-25217 - [patch]res_pjsip_outbound_publish.c needs a
      similar treatment for module unloading as
      res_pjsip_outbound_registration.c (Reported by Richard Mudgett)
 * ASTERISK-26265 - Errors ignored from some parts of system
      initialization. (Reported by Corey Farrell)
 * ASTERISK-26206 - [patch] res_pjsip: Use more compatible regex
      for get all (Reported by Dmitry)
 * ASTERISK-26256 - [patch] SIP/SDP origin (o=) contains brackets
      with IP6 (Reported by Alexander Traud)
 * ASTERISK-25996 - Remove "live_dangerously" requirement on
      DB(read) (Reported by Andrew Nagy)
 * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1:
      "libasteriskpj.so: undefined reference to..." (Reported by Hans
      van Eijsden)

Improvements made in this release:
-----------------------------------
 * ASTERISK-26409 - codec_opus: Update Asterisk to support the
      translation codec. (Reported by Kevin Harwell)
 * ASTERISK-26289 - Announcer channels in ConfBridges cause
      inefficiencies (Reported by Mark Michelson)
 * ASTERISK-25980 - [patch]res_fax: set FAXMODE variable to let
      dialplan know what fax transport was used (Reported by Alexei
      Gradinari)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.12.0

Thank you for your continued support of Asterisk!
2016-10-27 01:08:17 +00:00
jnemeth
e2e10f71c4 Update to Asterisk 11.24.0: this is a bug fix release.
The Asterisk Development Team has announced the release of Asterisk 11.24.0.

The release of Asterisk 11.24.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No
      Symmetric Response. (Reported by Alexander Traud)
 * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when
      bindaddr=[::] (Reported by Jacek)
 * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so
      instructed (Reported by Tzafrir Cohen)
 * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
      by Aaron Hamstra)
 * ASTERISK-26360 - app_queue: "queue show" output gets "failed to
      extend from 240 to 327" msgs. (Reported by Richard Mudgett)
 * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets)
      (Reported by Etienne Lessard)
 * ASTERISK-26288 - followme: fails to reset config items to
      default values on reload (Reported by Tzafrir Cohen)
 * ASTERISK-26282 - AEL: macro-call in Dial application, macro
      "lacks 's' extension" (Reported by chris de rock)
 * ASTERISK-26226 - pbx: Asterisk crash on AMI action
      "ShowDialplan" when there's a circular dependency between
      contexts (Reported by Etienne Lessard)
 * ASTERISK-26299 - app_queue: Queue application sometimes stops
      calling members with Local interface (Reported by Etienne
      Lessard)
 * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not
      cleaning up properly (Reported by Alexander Traud)
 * ASTERISK-26203 - res_fax: Deadlock when using
      FAXOPT(gateway)=yes with Local channels (Reported by Etienne
      Lessard)
 * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking
      inversion in T.38 query option with features bridging code
      (Reported by David Brillert)
 * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with
      local channels. (Reported by Richard Mudgett)
 * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when
      channels have multiple native formats (Reported by Matt Jordan)
 * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of
      SSLv3, security fix POODLE (CVE-2014-3566) (Reported by
      abelbeck)
 * ASTERISK-25706 - pbx: Abort asterisk on features reload
      (handle_hint_change) (Reported by Krzysztof Trempala)
 * ASTERISK-26233 - pbx: Failure to remove inconsistent extension
      names (Reported by Corey Farrell)
 * ASTERISK-26267 - ast_register_atexit callbacks should be run on
      failed startup. (Reported by Corey Farrell)
 * ASTERISK-26265 - Errors ignored from some parts of system
      initialization. (Reported by Corey Farrell)
 * ASTERISK-25996 - Remove "live_dangerously" requirement on
      DB(read) (Reported by Andrew Nagy)
 * ASTERISK-26237 - Fax is detected on regular calls. (Reported by
      Richard Mudgett)
 * ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
      command and attended transfer handling (Reported by Ben
      Smithurst)
 * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in
      conditional code. (Reported by Corey Farrell)
 * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence
      number even on lost packets. (Reported by Alexander Traud)
 * ASTERISK-26038 - 'make install' doesn't seem to install OS/X
      init files (Reported by Tzafrir Cohen)
 * ASTERISK-26133 - app_queue: Queue members receive multiple calls
      (Reported by Richard Miller)
 * ASTERISK-26196 - pbx: Time based includes can leak timezone
      string (Reported by Corey Farrell)
 * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing
      DTLS failure occurred on RTP instance (Reported by Edwin
      Vandamme)
 * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
      (Reported by Alexander Traud)
 * ASTERISK-25289 - Build System does not respect CFLAGS and
      CXXFLAGS when building menuselect (Reported by Jeffrey Walton)
 * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out
      of bounds and bugs (Reported by Alexei Gradinari)
 * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported
      by Joshua Colp)
 * ASTERISK-26157 - Build:   Fix errors highlighted by GCC 6.x
      (Reported by George Joseph)

Improvements made in this release:
-----------------------------------
 * ASTERISK-26220 - Add support for noreturn function attributes.
      (Reported by Corey Farrell)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0

Thank you for your continued support of Asterisk!
2016-10-26 05:53:37 +00:00
jnemeth
dde8383cf8 add and enable asterisk14 2016-10-25 08:29:16 +00:00
jnemeth
2f45135701 Initial import of Asterisk 14. It has been tested to compile
and run, but not a lot of functional testing.  This does not have
the new PJSIP, which will be coming in a followup commit.  This
also does not have the patches for compiling with Clang.  For
upgrading instructions, please see:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+14

----- 14.0.0 -----

The Asterisk Development Team is pleased to announce the release of
Asterisk 14.0.0.

Asterisk 14 is the next major release series of Asterisk. It is a
Standard Support release, similar to Asterisk 12. For more information
about support time lines for Asterisk releases, see the Asterisk
versions page:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 14, please
see the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+14

A short list of new features includes:

* A complete overhaul of the core DNS support in Asterisk, including
  implementing full NAPTR and SRV support in the PJSIP stack via the
  libunbound library.

* The ability to publish extension state to a SIP Subscription server,
  such as Kamailio. This includes the ability to automatically generate
  a hint in the dialplan based on device state changes using the new
  autohint setting.

* Playback of media from a remote HTTP server via a URI is now supported
  by all dialplan applications and AGI. Media retrieved using a URI is
  cached in a media cache and re-used when possible.

* When using ARI to manipulate media on a resource, a list of media
  resources can now be supplied. The media resources will be played back
  sequentially in the order that they are provided.

* Channels created via ARI can now be created and handed off to Stasis
  for external control prior to performing the outbound dial. This
  enables applications to set additional state on the channel prior to
  dialing, as well as enabling certain early media scenarios.

And much more!


More information about the new features can be found on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Documentation

A full list of all new features can also be found in the CHANGES file:
https://github.com/asterisk/asterisk/blob/14/CHANGES

For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.0.0

Thank you for your continued support of Asterisk!

----- 14.0.1 -----

The Asterisk Development Team has announced the release of Asterisk 14.0.1.

The release of Asterisk 14.0.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Improvements made in this release:
-----------------------------------
 * ASTERISK-26409 - codec_opus: Update Asterisk to support the
      translation codec. (Reported by Kevin Harwell)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.1

Thank you for your continued support of Asterisk!

----- 14.0.2 -----

The Asterisk Development Team has announced the release of Asterisk 14.0.2.

The release of Asterisk 14.0.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26410 - core: Asterisk 14 doesn't show the header in
      the console or verbose when starting (Reported by Dan Jenkins)
 * ASTERISK-26426 - format_ogg_opus: remove from source (Reported
      by Kevin Harwell)
 * ASTERISK-26425 - download_externals: ignore xmlstarlet return
      code for optional element (Reported by Kevin Harwell)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.2

Thank you for your continued support of Asterisk!
2016-10-25 08:16:31 +00:00
wiz
982c8f22e9 Recursive bump for all users of pgsql now that the default is 95. 2016-10-09 21:41:55 +00:00
adam
3b88bd43a5 Revbump post boost update 2016-10-07 18:25:29 +00:00
maya
ea106fdd68 srtp: do not conflict with builtin hmac in netbsd-7.99.x, use another name
(local_hmac). Fixes build on NetBSD.

Patch by Sérgio de Almeida Lenzi
2016-09-26 13:20:41 +00:00
jnemeth
66556849fd Update to Asterisk 11.23.1: this is a security fix release to fix
AST-2016-007.  Note that on Oct. 25th, this branch of Asterisk will
switch to security fixes, and one year later it will read end-of-life.

pkgsrc changes:
- don't use gethostbyname_r on NetBSD
- eliminate conflict with new hmac(1) function on NetBSd

----- AST-2016-007

The overlap dialing feature in chan_sip allows chan_sip to report
to a device that the number that has been dialed is incomplete and
more digits are required. If this functionality is used with a
device that has performed username/password authentication RTP
resources are leaked.  This occurs because the code fails to release
the old RTP resources before allocating new ones in this scenario.
If all resources are used then RTP port exhaustion will occur and
no RTP sessions are able to be set up.
2016-09-23 19:16:29 +00:00
jnemeth
e35d4086f7 Update to Asterisk 13.11.2: this is mainly a bug fix release
including two security issues:  AST-2016-006 and AST-2016-007.
Note that AST-2016-006 only affected setups using PJSIP, which
pkgsrc Asterisk does not.

pkgsrc changes:
- don't use gethostbyname_r on NetBSD
- eliminte conflict with new hmac(1) function on NetBSD

----- AST-2016-006

Asterisk can be crashed remotely by sending an ACK to it from an
endpoint username that Asterisk does not recognize.  Most SIP
request types result in an "artificial" endpoint being looked up,
but ACKs bypass this lookup. The resulting NULL pointer results in
a crash when attempting to determine if ACLs should be applied.

This issue was introduced in the Asterisk 13.10 release and only
affects that release.

This issue only affects users using the PJSIP stack with Asterisk.
Those users that use chan_sip are unaffected.

----- AST-2016-007

The overlap dialing feature in chan_sip allows chan_sip to report
to a device that the number that has been dialed is incomplete and
more digits are required. If this functionality is used with a
device that has performed username/password authentication RTP
resources are leaked.  This occurs because the code fails to release
the old RTP resources before allocating new ones in this scenario.
If all resources are used then RTP port exhaustion will occur and
no RTP sessions are able to be set up.

----- 13.11.2

The Asterisk Development Team has announced the release of Asterisk 13.11.2.

The release of Asterisk 13.11.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26349 -  13.11.1 res_pjsip/pjsip_distributor.c: Request
      'REGISTER' failed (Reported by Dmitry Melekhov)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.2

Thank you for your continued support of Asterisk!

----- 13.11.0

The Asterisk Development Team has announced the release of Asterisk 13.11.0.

The release of Asterisk 13.11.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-25904 - PJSIP: add contact.updated event (Reported by
      Alexei Gradinari)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26269 - res_pjsip: Wrong state for aors without
      registered contacts after startup (Reported by nappsoft)
 * ASTERISK-26299 - app_queue: Queue application sometimes stops
      calling members with Local interface (Reported by Etienne
      Lessard)
 * ASTERISK-26148 - pjsip: Cannot compile 13.10.0-rc1:
      "libasteriskpj.so: undefined reference to..." (Reported by Hans
      van Eijsden)
 * ASTERISK-26237 - Fax is detected on regular calls. (Reported by
      Richard Mudgett)
 * ASTERISK-26227 - sqlalchemy error due to long identifier name
      (Reported by Mark Michelson)
 * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported
      by Aaron Hamstra)
 * ASTERISK-26214 - Allow arbitrary time for fax detection to end
      on a channel (Reported by Richard Mudgett)
 * ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
      command and attended transfer handling (Reported by Ben
      Smithurst)
 * ASTERISK-26216 - res_fax: Deadlock when detect fax while channel
      executing Playback (Reported by Richard Mudgett)
 * ASTERISK-26212 - [patch] Makefile: Retain XML Declaration and
      DTD in docs. (Reported by Alexander Traud)
 * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in
      conditional code. (Reported by Corey Farrell)
 * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence
      number even on lost packets. (Reported by Alexander Traud)
 * ASTERISK-26038 - 'make install' doesn't seem to install OS/X
      init files (Reported by Tzafrir Cohen)
 * ASTERISK-26200 - [patch] res_pjsip_mwi: improve realtime
      performance - remove unneeded check on endpoint's contacts.
      (Reported by Alexei Gradinari)
 * ASTERISK-26133 - app_queue: Queue members receive multiple calls
      (Reported by Richard Miller)
 * ASTERISK-26196 - pbx: Time based includes can leak timezone
      string (Reported by Corey Farrell)
 * ASTERISK-26193 - chan_sip: reference leak in mwi_event_cb
      (Reported by Corey Farrell)
 * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing
      DTLS failure occurred on RTP instance (Reported by Edwin
      Vandamme)
 * ASTERISK-26191 - threadpool: Leak on duplicate taskprocessor for
      ast_threadpool_serializer_group (Reported by Corey Farrell)
 * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf.
      (Reported by Alexander Traud)
 * ASTERISK-26160 - pjsip: Updated->Reachable during qualify
      (Reported by Matt Jordan)
 * ASTERISK-25289 - Build System does not respect CFLAGS and
      CXXFLAGS when building menuselect (Reported by Jeffrey Walton)
 * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out
      of bounds and bugs (Reported by Alexei Gradinari)
 * ASTERISK-26177 - func_odbc: Database handle is kept when it
      should be released (Reported by Leandro Dardini)
 * ASTERISK-26184 - chan_sip: Reference leaks in error paths.
      (Reported by Corey Farrell)
 * ASTERISK-26181 - REF_DEBUG: Node object incorrectly logged
      during duplicate replacement (Reported by Corey Farrell)
 * ASTERISK-26180 - PJSIP: provide valid tcp nodelay option for
      reuse (Reported by Scott Griepentrog)
 * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported
      by Joshua Colp)
 * ASTERISK-26172 - res_sorcery_realtime: fix bug when successful
      sql UPDATE is treated as failed if there is no affected rows.
      (Reported by Alexei Gradinari)
 * ASTERISK-25772 - res_pjsip: Unexpected two BYE when answered
      (Reported by Dmitriy Serov)
 * ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
      due to server timeout (Reported by Ross Beer)
 * ASTERISK-26144 - Crash on loading codecs g729/g723 (Reported by
      Alexei Gradinari)
 * ASTERISK-26157 - Build:   Fix errors highlighted by GCC 6.x
      (Reported by George Joseph)
 * ASTERISK-26021 - Build codecs siren7 and siren14 for Asterisk 13
      (Reported by Daniel Denson)
 * ASTERISK-26326 - Crash when dialing MulticastRTP channel
      (Reported by George Joseph)

Improvements made in this release:
-----------------------------------
 * ASTERISK-26220 - Add support for noreturn function attributes.
      (Reported by Corey Farrell)
 * ASTERISK-22131 - Update the make dependencies script to pull,
      build, and install the correct pjproject (Reported by Matt
      Jordan)
 * ASTERISK-25471 - [patch]Add subscribe_context to res_pjsip
      (Reported by JoshE)
 * ASTERISK-26159 - res_hep: enabled by default and information
      sent to default address (Reported by Ross Beer)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.11.0

Thank you for your continued support of Asterisk!
2016-09-23 17:50:19 +00:00
jperkin
9c9760e3c6 Use PKGMANDIR. 2016-09-08 14:46:49 +00:00
adam
77b8ed74db Revbump after graphics/gd update 2016-08-03 10:22:08 +00:00
wen
e4813216a3 Update to 0.8
No upstream changelog found.
2016-07-24 23:40:31 +00:00
wen
140bc2944f Update to 1.61
Upstream changes:
1.61  Tue Jun 21 21:05:12 CEST 2016
    - Fixed RT#115491, remove the use of the encodings pragma, now deprecated.
    - Plenty of style, test and functionality fixes contributed by Joel Maslak
      and Paul Cochrane, as part of the CPAN PR Challenge. Awesome job, thanks!
    - Amended the main module documentation to make it clear this module is
      in maintenance mode and hasn't seen any major development work in years.
2016-07-24 23:30:13 +00:00
jnemeth
99d3471b70 Update to Asterisk 13.10.0: this is mainly a bug fix release.
The Asterisk Development Team has announced the release of Asterisk 13.10.0.

The release of Asterisk 13.10.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Improvements made in this release:
-----------------------------------
 * ASTERISK-26088 - Investigate heavy memory utilization by
      res_pjsip_pubsub (Reported by Richard Mudgett)
 * ASTERISK-26011 - [patch]PJSIP: add "via_addr", "via_port",
      "call_id" to contacts (Reported by Alexei Gradinari)
 * ASTERISK-25994 - [patch]res_pjsip: module load priority
      (Reported by Alexei Gradinari)
 * ASTERISK-25931 - PJSIP: add "reg_server" to contacts. (Reported
      by Alexei Gradinari)
 * ASTERISK-25835 - Authentication using 'Username' field from
      Digest (Reported by Ross Beer)
 * ASTERISK-25930 - PJSIP: disable multi domain to improve realtime
      performace (Reported by Alexei Gradinari)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26160 - pjsip: Updated->Reachable during qualify
      (Reported by Matt Jordan)
 * ASTERISK-26177 - func_odbc: Database handle is kept when it
      should be released (Reported by Leandro Dardini)
 * ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request
      due to server timeout (Reported by Ross Beer)
 * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to
      v21_details (Reported by Corey Farrell)
 * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a
      self-comparison (Reported by George Joseph)
 * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim
      generates a compile error (Reported by George Joseph)
 * ASTERISK-26128 - Alembic scripts are failing (Reported by Mark
      Michelson)
 * ASTERISK-26139 - test_res_pjsip_scheduler:  Compile failure if
      pjproject isn't installed in a system location (Reported by
      George Joseph)
 * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
      (Reported by Alexander Traud)
 * ASTERISK-26127 - res_pjsip_session: Crash due to race condition
      between res_pjsip_session unload and timer (Reported by Joshua
      Colp)
 * ASTERISK-26083 - ARI: Announcer channels staying around after
      playback to a bridge is finished (Reported by Per Jensen)
 * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in
      http.conf (Reported by Alexander Traud)
 * ASTERISK-26069 - Asterisk truncates To: header, dropping the
      closing '>' (Reported by Vasil Kolev)
 * ASTERISK-26097 - [patch] CLI: show maximum file descriptors
      (Reported by Alexander Traud)
 * ASTERISK-25262 - Memory leak when a caller channel does multiple
      dials and CEL is enabled (Reported by Etienne Lessard)
 * ASTERISK-26092 - [Segfault] in res_rtp_asterisk.c:4268 after
      Remotely bridged channels (Reported by Niklas Larsson)
 * ASTERISK-26096 - res_hep: Crash when configuration file is
      missing (Reported by Niklas Larsson)
 * ASTERISK-26089 - Invalid security events during boot using PJSIP
      Realtime (Reported by Scott Griepentrog)
 * ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by
      Ross Beer)
 * ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B.
      Davis)
 * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
      against libsrtp-1.5.0 (Reported by Patrick Laimbock)
 * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar
      cr (Reported by Alexander Traud)
 * ASTERISK-26070 - ari/channels:  Creating a local channel without
      an originator adds all audio formats to it's capabilities
      (Reported by George Joseph)
 * ASTERISK-26078 - core: Memory leak in logging (Reported by
      Etienne Lessard)
 * ASTERISK-26065 - chan_pjsip: MWI NOTIFY contents not ordered
      properly (Reported by Ross Beer)
 * ASTERISK-26063 - ${PJSIP_HEADER(read,Call-ID)} does not work -
      documentation needs clarification for when read/write is
      possible (Reported by Private Name)
 * ASTERISK-25777 - data race in threadpool (Reported by Badalian
      Vyacheslav)
 * ASTERISK-26038 - 'make install' doesn't seem to install OS/X
      init files (Reported by Tzafrir Cohen)
 * ASTERISK-26029 - parking: ast_parking_park_call should return
      parking_space instead of parking_exten (Reported by Diederik de
      Groot)
 * ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
      LAST_INSERT_ID() always returns zero. (Reported by Edwin
      Vandamme)
 * ASTERISK-25941 - chan_pjsip: Crash on an immediate SIP final
      response (Reported by Javier Riveros )
 * ASTERISK-26014 - res_sorcery_astdb: Make tolerant of unknown
      fields (Reported by Joshua Colp)
 * ASTERISK-24986 - keepalive INFO packages ignored by asterisk
      (Reported by Ilya Trikoz)
 * ASTERISK-26034 - T.38 passthrough problem behind firewall due to
      early nosignal packet (Reported by George Joseph)
 * ASTERISK-26030 - call cut because of double Session-Expires
      header in re-invite after proxy authentication is required
      (Reported by George Joseph)
 * ASTERISK-25964 - Outbound registrations created via ARI/push
      configuration do not clean up outbound registrations currently
      in flight (Reported by Matt Jordan)
 * ASTERISK-26005 - res_pjsip: Multiple SIP messages are combined
      into 1 TCP packet (Reported by Ross Beer)
 * ASTERISK-25352 - res_hep_rtcp correlation_id is different then
      res_hep (Reported by Kevin Scott Adams)
 * ASTERISK-26008 - app_followme does not delete recorded name
      prompt (Reported by Tzafrir Cohen)
 * ASTERISK-26007 - res_pjsip: Endpoints deleting early after
      upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)
 * ASTERISK-25990 - PJSIP TLS registration should respect
      client_uri scheme when generating Contact URI (Reported by
      Sebastian Damm)
 * ASTERISK-25978 - res_pjsip_authenticator_digest: Should not use
      source port in nonce verification (Reported by Mark Michelson)
 * ASTERISK-25993 - pjproject: Allow bundling to not require
      everything it does (Reported by Joshua Colp)
 * ASTERISK-25956 - Compilation error in conditionally compiled
      code in config_options.c (Reported by Chris Trobridge)
 * ASTERISK-25998 - file: Crash when using nativeformats (Reported
      by Joshua Colp)
 * ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
      (Reported by Ross Beer)
 * ASTERISK-25968 - pjproject_bundled:  Configure and make need to
      be re-tested (Reported by George Joseph)
 * ASTERISK-24463 - Voicemail email address corrupt or not sent
      when message is in the process of being recorded during reload
      (Reported by John Campbell)
 * ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by
      Dmitriy Serov)
 * ASTERISK-25963 - func_odbc requires reconnect checks for stale
      connections (Reported by Ross Beer)
 * ASTERISK-25961 - tests/channels/SIP/sip_tls_call: Sporadic crash
      when running test (Reported by Joshua Colp)
 * ASTERISK-16115 - [patch] problem with ringinuse=no, queue
      members receive sometimes two calls (Reported by nik600)
 * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir
      only works if you manually add secret.conf yourself (Reported by
      Jonathan R. Rose)
 * ASTERISK-25950 - [patch]SIP channel does not send PeerStatus
      events for autocreated peers (Reported by Kirill Katsnelson)
 * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions
      are case sensitive to QueueName (Reported by Javier Acosta)

New Features made in this release:
-----------------------------------
 * ASTERISK-25904 - PJSIP: add contact.updated event (Reported by
      Alexei Gradinari)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.10.0

Thank you for your continued support of Asterisk!
2016-07-24 06:35:50 +00:00
jnemeth
a5ac47e94e Update to Asterisk 11.23.0: this is a bug fix release.
The Asterisk Development Team has announced the release of Asterisk 11.23.0.

The release of Asterisk 11.23.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to
      v21_details (Reported by Corey Farrell)
 * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a
      self-comparison (Reported by George Joseph)
 * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim
      generates a compile error (Reported by George Joseph)
 * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
      (Reported by Alexander Traud)
 * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in
      http.conf (Reported by Alexander Traud)
 * ASTERISK-26069 - Asterisk truncates To: header, dropping the
      closing '>' (Reported by Vasil Kolev)
 * ASTERISK-26097 - [patch] CLI: show maximum file descriptors
      (Reported by Alexander Traud)
 * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling
      against libsrtp-1.5.0 (Reported by Patrick Laimbock)
 * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar
      cr (Reported by Alexander Traud)
 * ASTERISK-26038 - 'make install' doesn't seem to install OS/X
      init files (Reported by Tzafrir Cohen)
 * ASTERISK-26034 - T.38 passthrough problem behind firewall due to
      early nosignal packet (Reported by George Joseph)
 * ASTERISK-26030 - call cut because of double Session-Expires
      header in re-invite after proxy authentication is required
      (Reported by George Joseph)
 * ASTERISK-26008 - app_followme does not delete recorded name
      prompt (Reported by Tzafrir Cohen)
 * ASTERISK-24463 - Voicemail email address corrupt or not sent
      when message is in the process of being recorded during reload
      (Reported by John Campbell)
 * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir
      only works if you manually add secret.conf yourself (Reported by
      Jonathan R. Rose)
 * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions
      are case sensitive to QueueName (Reported by Javier Acosta)
 * ASTERISK-16115 - [patch] problem with ringinuse=no, queue
      members receive sometimes two calls (Reported by nik600)
 * ASTERISK-25934 - chan_sip should not require sipregs or
      updateable sippeers table unless rt (Reported by Jaco Kroon)
 * ASTERISK-25888 - Frequent segfaults in function can_ring_entry()
      of app_queue.c (Reported by Sébastien Couture)
 * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
 * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE
      without adding them to the local hangupcauses via
      ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)
 * ASTERISK-25407 - Asterisk fails to log to multiple syslog
      destinations (Reported by Elazar Broad)
 * ASTERISK-25510 - [patch]Log to syslog failing (Reported by
      Michael Newton)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25444 - [patch]Music On Hold Warning misleading
      (Reported by Conrad de Wet)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0

Thank you for your continued support of Asterisk!
2016-07-23 08:27:44 +00:00
taca
fdfae7cea3 Update ruby-termios to 1.0.2.
pkgsrc change: Change HOEMDIR to https://github.com/arika/ruby-termios.

* Move extension files to ext/ directory.
* Several miscellaneous changes.
2016-07-17 14:25:00 +00:00
rillig
013eef793b Removed unused BUILDLINK_SETENV and made Makefile simpler. 2016-07-10 10:07:16 +00:00
rillig
6109c44c70 Fixed pkglint warnings. 2016-07-10 09:59:07 +00:00
wiz
ad0031c15e Remove python33: adapt all packages that refer to it. 2016-07-09 13:03:30 +00:00
wiz
2b0a009d0e Bump PKGREVISION for perl-5.24.0 for everything mentioning perl. 2016-07-09 06:37:46 +00:00
schnoebe
0d38a4e2fb Revert changing the default away from "inet6". 2016-06-20 15:25:39 +00:00
schnoebe
4107a5865d Bump PGKREVISION (forgot to do that when updating the PLIST.). 2016-06-16 16:05:51 +00:00
schnoebe
a0905a18be options.mk:
Don't default on inet6, since the inet6 code in conserver8 depends
    on some Linux-isms (ipv6 sockets can accept ipv4 packets.)

PLIST:
    add some example configurations that were missing.
2016-06-15 23:02:10 +00:00
jnemeth
062c98ae34 Upgrade to Asterisk 13.9.1: this is a bugfix release. Note that
since the package doesn't support PJSIP (yet), all reference to
PJSIP bugs are not applicable.

----- 13.9.1

The Asterisk Development Team has announced the release of Asterisk 13.9.1.

The release of Asterisk 13.9.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26007 - res_pjsip: Endpoints deleting early after
      upgrade from 13.8.2 to 13.9 (Reported by Greg Siemon)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.1

Thank you for your continued support of Asterisk!

----- 13.9.0

The Asterisk Development Team has announced the release of Asterisk 13.9.0.

The release of Asterisk 13.9.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25963 - func_odbc requires reconnect checks for stale
      connections (Reported by Ross Beer)
 * ASTERISK-25970 - Segfault in pjsip_url_compare (Reported by
      Dmitriy Serov)
 * ASTERISK-25938 - res_odbc: MySQL/MariaDB statement
      LAST_INSERT_ID() always returns zero. (Reported by Edwin
      Vandamme)
 * ASTERISK-25927 - Removed option "registertrying" is still
      documented in sip.conf.sample (Reported by Etienne Lessard)
 * ASTERISK-25947 - Protocol transfers to stasis applications are
      missing the StasisStart with the replace_channel object.
      (Reported by Richard Mudgett)
 * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
      fails to get app name (Reported by John Bigelow)
 * ASTERISK-24782 - StasisEnd event not present for channel that
      was swapped out for another after completing attended transfer
      (Reported by John Bigelow)
 * ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed
      ConnectedLine information (Reported by George Joseph)
 * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
      thread (Reported by Joshua Colp)
 * ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
      not raised (Reported by Joshua Colp)
 * ASTERISK-25934 - chan_sip should not require sipregs or
      updateable sippeers table unless rt (Reported by Jaco Kroon)
 * ASTERISK-25888 - Frequent segfaults in function can_ring_entry()
      of app_queue.c (Reported by Sébastien Couture)
 * ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets
      exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George
      Joseph)
 * ASTERISK-25707 - Long contact URIs or hostnames can crash
      pjproject/Asterisk under certain conditions (Reported by George
      Joseph)
 * ASTERISK-25123 - Bracketed IPv6 Contact header parameter
      unparsable with Asterisk/PJSIP (Reported by Anthony Messina)
 * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
 * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE
      without adding them to the local hangupcauses via
      ast_channel_hangupcause_hash_set (Reported by Jaco Kroon)
 * ASTERISK-25885 - res_pjsip: Race condition between adding
      contact and automatic expiration (Reported by Joshua Colp)
 * ASTERISK-25910 - pjproject:  Via headers are not parsed when
      "received" contains an IPv6 address (Reported by George Joseph)
 * ASTERISK-25890 - Asterisk 13.8.0 alembic database update fails
      (Reported by Harley Peters)
 * ASTERISK-25894 - [patch] webrtc video broken due to missing
      marker bits in RTP streams (Reported by Jacek Konieczny)
 * ASTERISK-25854 - No audio after HOLD/RESUME - incorrect
      a=recvonly in SDP from Asterisk (Reported by Robert McGilvray)
 * ASTERISK-25873 - res_pjsip: Bundled pjproject: compile error,
      cannot find -lasteriskpj (Reported by Hans van Eijsden)
 * ASTERISK-25882 - ARI: Crash can occur due to race condition when
      attempting to operate on a hung up channel (Part 2) (Reported by
      Richard Mudgett)
 * ASTERISK-25867 - [patch] Video delay on app_echo (Reported by
      Jacek Konieczny)
 * ASTERISK-24605 - res_parking option parkeddynamic does not work
      with the core Features 'parkcall' (DTMF initiated parking)
      (Reported by Philip Correia)
 * ASTERISK-25826 - PJSIP / Sorcery slow load from realtime
      (Reported by Ross Beer)
 * ASTERISK-24596 - Unclear how to use Park application with
      res_parking 'parkeddynamic' enabled. Documentation? (Reported by
      Philip Correia)
 * ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
      possible codecs configured for peer as opposed to intersection
      of configured codecs and offered codecs (Reported by Taylor
      Hawkes)
 * ASTERISK-25825 - Crashes during shutdown when running CLI
      commands (Reported by Mark Michelson)
 * ASTERISK-25407 - Asterisk fails to log to multiple syslog
      destinations (Reported by Elazar Broad)
 * ASTERISK-25510 - [patch]Log to syslog failing (Reported by
      Michael Newton)
 * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
      data corruption (Reported by Gianluca Merlo)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25865 - Message-Account Missing From PJSIP MWI
      (Reported by Ross Beer)
 * ASTERISK-25444 - [patch]Music On Hold Warning misleading
      (Reported by Conrad de Wet)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.9.0

Thank you for your continued support of Asterisk!
2016-06-09 04:41:48 +00:00
wiz
86a78fce2e Bump PKGREVISION for perl-5.24. 2016-06-08 19:22:13 +00:00
wiz
57199de455 Switch to MASTER_SITES_PYPI. 2016-06-08 17:43:20 +00:00
jperkin
36e6903fd8 Remove the stability entity, it has no meaning outside of an official context. 2016-06-08 10:16:50 +00:00
jperkin
31ffe7cbb6 Change the service_bundle name to "export" to reduce diffs between the
original manifest.xml file and the output from "svccfg export".
2016-06-08 09:46:01 +00:00
jnemeth
e7a8554dd7 Update to Asterisk 13.8.2: this is mainly a bug fix release. It
also contains fixes for AST-2016-004 and AST-2016-005.  However,
those issues only affected the pjsip module.  Since Asterisk in
pkgsrc doesn't (yet) use pjsip, it wasn't affected.

----- 13.8.2

The Asterisk Development Team has announced the release of Asterisk 13.8.2.

The release of Asterisk 13.8.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events
      not raised (Reported by Joshua Colp)
 * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP
      thread (Reported by Joshua Colp)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.2

Thank you for your continued support of Asterisk!

----- 13.8.0

The Asterisk Development Team has announced the release of Asterisk 13.8.0.

The release of Asterisk 13.8.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write
      contents to file (Reported by Ray Crumrine)
 * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
      Journo)
 * ASTERISK-25480 - [patch]Add field PauseReason on
      QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25849 - chan_pjsip: transfers with direct media
      sometimes drops audio (Reported by Kevin Harwell)
 * ASTERISK-25113 - install_prereq in Debian 8 without "standard
      system utilities" (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25814 - Segfault at f ip in res_pjsip_refer.so
      (Reported by Sergio Medina Toledo)
 * ASTERISK-25023 - Deadlock in chan_sip in
      update_provisional_keepalive (Reported by Arnd Schmitter)
 * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
      channel (Reported by Filip Frank)
 * ASTERISK-25829 - res_pjsip: PJSIP does not accept spaces when
      separating multiple AORs (Reported by Mateusz Kowalski)
 * ASTERISK-25771 - ARI:Crash - Attended transfers of channels into
      Stasis application. (Reported by Javier Riveros )
 * ASTERISK-25830 - Revision 2451d4e breaks NAT (Reported by Sean
      Bright)
 * ASTERISK-25582 - Testsuite: Reactor timeout error in
      tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
      Jordan)
 * ASTERISK-25811 - Unable to delete object from sorcery cache
      (Reported by Ross Beer)
 * ASTERISK-25800 - [patch] Calculate talktime when is first call
      answered (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25727 - RPM build requires OPTIONAL_API cflag due to
      PJSIP requirement (Reported by Gergely Dömsödi)
 * ASTERISK-25337 - Crash on PJSIP_HEADER Add P-Asserted-Identity
      when calling from Gosub (Reported by Jacques Peacock)
 * ASTERISK-25738 - res_pjsip_pubsub: Crash while executing
      OutboundSubscriptionDetail ami action (Reported by Kevin
      Harwell)
 * ASTERISK-25721 - [patch] res_phoneprov: memory leak and
      heap-use-after-free (Reported by Badalian Vyacheslav)
 * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
      returns garbage (Reported by Etienne Lessard)
 * ASTERISK-25751 - res_pjsip: Support
      pjsip_dlg_create_uas_and_inc_lock (Reported by Joshua Colp)
 * ASTERISK-25606 - Core dump when using transports in sorcery
      (Reported by Martin Moučka)
 * ASTERISK-20987 - non-admin users, who join muted conference are
      not being muted (Reported by hristo)
 * ASTERISK-25737 - res_pjsip_outbound_registration: line option
      not in Alembic (Reported by Joshua Colp)
 * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
      udptl_rx_packet cause ast_frdup crash (Reported by Walter
      Doekes)
 * ASTERISK-25742 - Secondary IFP Packets can result in accessing
      uninitialized pointers and a crash (Reported by Torrey Searle)
 * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
      Vulnerability - Investigate vulnerability of HTTP server
      (Reported by Alex A. Welzl)
 * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
      non-default timert1 (Reported by Alexander Traud)
 * ASTERISK-25702 - PjSip realtime DB and Cache Errors since
      upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
      Nic Colledge)
 * ASTERISK-25730 - build:  make uninstall after make distclean
      tries to remove root (Reported by George Joseph)
 * ASTERISK-25725 - core: Incorrect XML documentation may result in
      weird behavior (Reported by Joshua Colp)
 * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
      sip_sipredirect (Reported by Badalian Vyacheslav)
 * ASTERISK-25709 - ARI: Crash can occur due to race condition when
      attempting to operate on a hung up channel (Reported by Mark
      Michelson)
 * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
      by Badalian Vyacheslav)
 * ASTERISK-25685 - infrastructure: Run alembic in Jenkins build
      script (Reported by Joshua Colp)
 * ASTERISK-25712 - Second call to already-on-call phone and
      Asterisk sends "Ready" (Reported by Richard Mudgett)
 * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
      (Reported by Badalian Vyacheslav)
 * ASTERISK-25179 - CDR(billsec,f) and CDR(duration,f) report
      incorrect values (Reported by Gianluca Merlo)
 * ASTERISK-25611 - core: threadpool thread_timeout_thrash unit
      test sporadically failing (Reported by Joshua Colp)
 * ASTERISK-24097 - Documentation - CHANNEL function help text
      missing 'linkedid' argument (Reported by Steven T. Wheeler)
 * ASTERISK-25700 - main/config: Clean config maps on shutdown.
      (Reported by Corey Farrell)
 * ASTERISK-25696 - bridge_basic: don't cache xferfailsound during
      a transfer (Reported by Kevin Harwell)
 * ASTERISK-25697 - bridge_basic: don't play an attended transfer
      fail sound after target hangs up (Reported by Kevin Harwell)
 * ASTERISK-25683 - res_ari: Asterisk fails to start if compiled
      with MALLOC_DEBUG  (Reported by yaron nahum)
 * ASTERISK-25686 - PJSIP: qualify_timeout is a double, database
      schema is an integer (Reported by Marcelo Terres)
 * ASTERISK-25690 - Hanging up when executing connected line sub
      does not cause hangup (Reported by Joshua Colp)
 * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
      reload' cause a crash (Reported by Sean Bright)
 * ASTERISK-25632 - res_pjsip_sdp_rtp: RTP is sent from wrong IP
      address when multihomed (Reported by Olivier Krief)
 * ASTERISK-25637 - Multi homed server using wrong IP (Reported by
      Daniel Journo)
 * ASTERISK-25394 - pbx: Incorrect device and presence state when
      changing hint details (Reported by Joshua Colp)
 * ASTERISK-25640 - pbx: Deadlock on features reload and state
      change hint. (Reported by Krzysztof Trempala)
 * ASTERISK-25681 - devicestate: Engine thread is not shut down
      (Reported by Corey Farrell)
 * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
      shutdown (Reported by Corey Farrell)
 * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
      Corey Farrell)
 * ASTERISK-25675 - Endpoint not listed as Unreachable (Reported by
      Daniel Journo)
 * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
      by Corey Farrell)
 * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
      Farrell)
 * ASTERISK-25668 - res_pjsip: Deadlock in distributor (Reported by
      Mark Michelson)
 * ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
      (Reported by Corey Farrell)
 * ASTERISK-25647 - bug of cel_radius.c: wrong point of
      ADD_VENDOR_CODE (Reported by Aaron An)
 * ASTERISK-25317 - asterisk sends too many stun requests (Reported
      by Stefan Engström)
 * ASTERISK-25137 - endpoint stasis messages are delivered twice
      (Reported by Vitezslav Novy)
 * ASTERISK-25116 - res_pjsip:  Two PeerStatus AMI messages are
      sent for every status change (Reported by George Joseph)
 * ASTERISK-25641 - bridge: GOTO_ON_BLINDXFR doesn't work on
      transfer initiated channel (Reported by Dmitry Melekhov)
 * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
      Brandon)
 * ASTERISK-25442 - using realtime (mysql) queue members are never
      updated in wait_our_turn function (app_queue.c)  (Reported by
      Carlos Oliva)
 * ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
      caching (Reported by Joshua Colp)
 * ASTERISK-25601 - json: Audit reference usage and thread safety
      (Reported by Joshua Colp)
 * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
      sungtae kim)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25495 - [patch] Prevent old-update packages on
      repository Debian systems (Reported by Rodrigo Ramirez
      Norambuena)
 * ASTERISK-25846 - Gracefully deal with Absent Stasis Apps
      (Reported by Andrew Nagy)
 * ASTERISK-25791 - res_pjsip_caller_id: Lack of support for
      Anonymous <anonymous at anonymous.invalid> (Reported by Anthony
      Messina)
 * ASTERISK-24813 - asterisk.c: #if statement in listener()
      confuses code folding editors (Reported by Corey Farrell)
 * ASTERISK-25767 - [patch] Add check to configure for sanitizes
      (Reported by Badalian Vyacheslav)
 * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
      core set (Reported by Rusty Newton)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.8.0

Thank you for your continued support of Asterisk!
2016-05-06 07:41:06 +00:00
wiz
0976093b1e Fix braces in post-install rule.
Did this work?!
2016-05-05 06:23:57 +00:00
jnemeth
d5eb03db76 Update to Asterisk 11.22.0: this is mostly a bug fix release.
----- 11.22.0

The Asterisk Development Team has announced the release of Asterisk 11.22.0.

The release of Asterisk 11.22.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
      data corruption (Reported by Gianluca Merlo)
 * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
      channel (Reported by Filip Frank)
 * ASTERISK-25800 - [patch] Calculate talktime when is first call
      answered (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
      returns garbage (Reported by Etienne Lessard)
 * ASTERISK-20987 - non-admin users, who join muted conference are
      not being muted (Reported by hristo)
 * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
      Vulnerability - Investigate vulnerability of HTTP server
      (Reported by Alex A. Welzl)
 * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
      udptl_rx_packet cause ast_frdup crash (Reported by Walter
      Doekes)
 * ASTERISK-25742 - Secondary IFP Packets can result in accessing
      uninitialized pointers and a crash (Reported by Torrey Searle)
 * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
      non-default timert1 (Reported by Alexander Traud)
 * ASTERISK-25730 - build:  make uninstall after make distclean
      tries to remove root (Reported by George Joseph)
 * ASTERISK-25722 - ASAN & testsute: stack-buffer-overflow in
      sip_sipredirect (Reported by Badalian Vyacheslav)
 * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
      by Badalian Vyacheslav)
 * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
      (Reported by Badalian Vyacheslav)
 * ASTERISK-25701 - core: Endless loop in "core show
      taskprocessors" (Reported by ibercom)
 * ASTERISK-25700 - main/config: Clean config maps on shutdown.
      (Reported by Corey Farrell)
 * ASTERISK-25690 - Hanging up when executing connected line sub
      does not cause hangup (Reported by Joshua Colp)
 * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
      reload' cause a crash (Reported by Sean Bright)
 * ASTERISK-25394 - pbx: Incorrect device and presence state when
      changing hint details (Reported by Joshua Colp)
 * ASTERISK-25640 - pbx: Deadlock on features reload and state
      change hint. (Reported by Krzysztof Trempala)
 * ASTERISK-25681 - devicestate: Engine thread is not shut down
      (Reported by Corey Farrell)
 * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
      shutdown (Reported by Corey Farrell)
 * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
      Corey Farrell)
 * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
      by Corey Farrell)
 * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
      Farrell)
 * ASTERISK-25647 - bug of cel_radius.c: wrong point of
      ADD_VENDOR_CODE (Reported by Aaron An)
 * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
      Brandon)
 * ASTERISK-25442 - using realtime (mysql) queue members are never
      updated in wait_our_turn function (app_queue.c)  (Reported by
      Carlos Oliva)
 * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
      sungtae kim)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24813 - asterisk.c: #if statement in listener()
      confuses code folding editors (Reported by Corey Farrell)
 * ASTERISK-25767 - [patch] Add check to configure for sanitizes
      (Reported by Badalian Vyacheslav)
 * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
      core set (Reported by Rusty Newton)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0

Thank you for your continued support of Asterisk!

----- 11.21.2

The Asterisk Development Team has announced the release of Asterisk 11.21.2.

The release of Asterisk 11.21.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25770 - Check for OpenSSL defines before trying to use
      them. (Reported by Kevin Harwell)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2

Thank you for your continued support of Asterisk!
2016-05-05 02:00:33 +00:00
prlw1
104960e18b revbump for libsoup's ABI issue 2016-05-03 11:40:00 +00:00
dholland
19788bca04 PR 50556: comms/lirc doesn't build
Add patches to work around gcc inline mess.

Note: this package should be updated and the PR contains an update, but
I don't want to do that when I can't compile it (whereas simple mechanical
patches are much less likely to go astray...)
2016-05-02 02:32:14 +00:00
ryoon
ac20a93574 Recursive revbump from textproc/icu 57.1 2016-04-11 19:01:33 +00:00
dbj
dde82c09aa update PKG_{FAIL,SKIP}_REASON with += 2016-04-11 04:22:33 +00:00
ryoon
e37e5f4eb5 Drop setuid for bin/minicom, bump PKGREVISION
setuid bin/minicom makes zmodem support with comms/lrzsz unusable.
2016-03-22 23:24:10 +00:00
khorben
5e9c9bec7d Register missing dependency on security/openssl
This fixes the build with PKGSRC_MKPIE.
2016-03-22 11:29:11 +00:00
schnoebe
e9fe9116f6 Upstream: Update to 8.2.1
version 8.2.1 (Jun 2, 2015):
	- added TCP keepalives between client and server - TCP-based consoles
	  already had the code - this was mostly an oversight
	- patch for SEGV and task execution - patch by Artem Savkov
	  <asavkov@redhat.com>
	- expanded break sequences from [1-9] to [1-9a-z] - based on patch by
	  Artem Savkov <asavkov@redhat.com>

pkgsrc:
    options.mk:
	add support inet6
	The way the conserver is coded, you get inet6 or you get uds
    patches/patch-conserver_readcfg.c:
	new patch, fixing a setproctitle() bug with inet6.
	This needs to be sent upstream.
    Makefile:
	install sample configurations
	Change maintainer to pkgsrc-users@netbsd.org
	    old maintainer address bounced.
	drop ``--with-regex'' option, it's no longer supported.
2016-03-13 18:31:11 +00:00
jperkin
17661ff9a5 Bump PKGREVISION for security/openssl ABI bump. 2016-03-05 11:27:40 +00:00
jperkin
bafb0e6d43 Use OPSYSVARS. 2016-02-25 11:32:19 +00:00
khorben
9e4ab1ad8c Bump revision following fix for x11/deforaos-libdesktop 2016-02-20 02:28:14 +00:00
khorben
ce292f47fc Package DeforaOS Phone 0.5.0
This release brings:
- support the latest libSystem
- compatibility with Gtk+ 3
- improved hardware compatibility (GSM)
- improved handling of SMS and USSD messages
- new "console" plug-in
- improved "profiles" plug-in
- further improvements to the user interface
2016-02-20 02:10:07 +00:00
ryoon
28b3e65fd0 Add picocom 2016-02-14 07:55:55 +00:00
ryoon
9d9ad4c7cb Import picocom-2.1 as comms/picocom.
As its name suggests, picocom is a minimal dumb-terminal emulation
program. It is, in principle, very much like minicom, only it's
"pico" instead of "mini"!

It was designed to serve as a simple, manual, modem configuration,
testing, and debugging tool. It has also served (quite well) as a
low-tech serial communications program to allow access to all types
of devices that provide serial consoles. It could also prove useful
in many other similar tasks.
2016-02-14 07:54:51 +00:00
leot
d895e2b8df Update comms/py-gammu to 2.5.
Changes:
2.5
===
* Compatibility with Gammu >= 1.36.7

2.4
===
* Fixed possible crash when initializing SMSD with invalid parameters.
* Fixed crash on handling diverts on certain architectures.
2016-02-07 15:30:40 +00:00
leot
9ee7afbba3 Update comms/gammu to 1.37.0.
Changes:
20160203 - 1.37.0
[-] * Improved compatibility with ZTE MF190.
[-] * Improved compatibility with Huawei E1750.
[-] * Improved compatibility with Huawei E1752.
[-] * Increased detail of reported errors from SMSD.

20151208 - 1.36.8
[-] * Changed default value for ReceiveFrequency.
[-] * Fixed compatibility for PostgreSQL.
[-] * Fixed build failure with all disabled SMSD backends.
[-] * Documentation improvements.
[-] * Fixed mixing C++ code with SMSD.

20151129 - 1.36.7
[-] * Support devices which do not report full network status.
[-] * Disable Huawei unsolicited messages on startup.
[-] * Various improvements for Huawei modems.
[-] * Fixed compilation on Windows.
[-] * Fixed regression with Siemens AX75.
[-] * Improved decoding of USSD responses.
[-] * Properly decode emojis to console or files backend.
[+] * Added support for proxying the connection through arbitrary command.
[+] * SMSD now tracks retries count per message.

20151012 - 1.36.6
[-] * Fixed installation of bash-completion script.
[-] * Fixed timezone manipulation in SMSD.
[-] * Documentation improvements.
[-] * Fixed licensing of helper/win32-dirent.*.
[*] * Increased default speed for AT connection to 115200.
[*] * Improve AT module initialization.

20150826 - 1.36.5
[-] * Properly use timezones with SQLite in SMSD.
[-] * Improve support for Huawei E1752.
[-] * Fixed compilation on distros with old Glib.
2016-02-07 15:29:34 +00:00
jnemeth
1586aaebbd Update Asterisk to 13.7.2: this is mainly bug fixes with some minor
features and fixes for AST-2016-001, AST-2016-002, and AST-2016-003.
Also some pkglinting.

----- 13.7.2

The Asterisk Development Team has announced the release of Asterisk 13.7.2.

The release of Asterisk 13.7.2 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25702 - PjSip realtime DB and Cache Errors since
      upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2 (Reported by
      Nic Colledge)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.2

Thank you for your continued support of Asterisk!

----- 13.7.1

The Asterisk Development Team has announced security releases for Certified
Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.

The release of these versions resolves the following security vulnerabilities:

* AST-2016-001: BEAST vulnerability in HTTP server

  The Asterisk HTTP server currently has a default configuration which allows
  the BEAST vulnerability to be exploited if the TLS functionality is enabled.
  This can allow a man-in-the-middle attack to decrypt data passing through it.

* AST-2016-002: File descriptor exhaustion in chan_sip

  Setting the sip.conf timert1 value to a value higher than 1245 can cause an
  integer overflow and result in large retransmit timeout times. These large
  timeout values hold system file descriptors hostage and can cause the system
  to run out of file descriptors.

* AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.

  If no UDPTL packets are lost there is no problem. However, a lost packet
  causes Asterisk to use the available error correcting redundancy packets. If
  those redundancy packets have zero length then Asterisk uses an uninitialized
  buffer pointer and length value which can cause invalid memory accesses later
  when the packet is copied.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
 * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
 * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf

Thank you for your continued support of Asterisk!

----- 13.7.0

The Asterisk Development Team has announced the release of Asterisk 13.7.0.

The release of Asterisk 13.7.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-25419 - Dialplan Application for Integration of StatsD
      (Reported by Ashley Sanders)
 * ASTERISK-25549 - Confbridge: Add participant timeout option
      (Reported by Mark Michelson)
 * ASTERISK-24922 - ARI: Add the ability to intercept hold and
      raise an event (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25689 - pjsip show contacts not working in Asterisk
      13.7rc2 (Reported by Marcelo Terres)
 * ASTERISK-25640 - pbx: Deadlock on features reload and state
      change hint. (Reported by Krzysztof Trempala)
 * ASTERISK-25664 - ast_format_cap_append_by_type leaks a reference
      (Reported by Corey Farrell)
 * ASTERISK-25601 - json: Audit reference usage and thread safety
      (Reported by Joshua Colp)
 * ASTERISK-25625 - res_sorcery_memory_cache: Add full backend
      caching (Reported by Joshua Colp)
 * ASTERISK-25615 - res_pjsip: Setting transport async_operations >
      1 causes segfault on tls transports (Reported by George Joseph)
 * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
      thread of asterisk is not released (Reported by Hiroaki Komatsu)
 * ASTERISK-25619 - res_chan_stats not sending the correct
      information to StatsD (Reported by Tyler Cambron)
 * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
      Corey Farrell)
 * ASTERISK-25609 - [patch]Asterisk may crash when calling
      ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
 * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
      answer waiting time is more than ~7sec (Reported by Aleksei
      Kulakov)
 * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
      (Reported by Alexander Traud)
 * ASTERISK-25616 - Warning with a Codec Module which supports PLC
      with FEC (Reported by Alexander Traud)
 * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
      Dudás József)
 * ASTERISK-25608 - res_pjsip/contacts/statsd:  Lifecycle events
      aren't consistent (Reported by George Joseph)
 * ASTERISK-25584 - [patch] format-attribute module: VP8 missing
      (Reported by Alexander Traud)
 * ASTERISK-25583 - [patch] format-attribute module: RFC 7587 (Opus
      Codec) (Reported by Alexander Traud)
 * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
      that module installed (Reported by Ben Langfeld)
 * ASTERISK-25595 - Unescaped : in messge sent to statsd (Reported
      by Niklas Larsson)
 * ASTERISK-25476 - chan_sip loses registrations after a while
      (Reported by Michael Keuter)
 * ASTERISK-25598 - res_pjsip:  Contact status messages are
      printing a hash instead of the uri (Reported by George Joseph)
 * ASTERISK-25600 - bridging: Inconsistency in BRIDGEPEER (Reported
      by Jonathan Rose)
 * ASTERISK-25582 - Testsuite: Reactor timeout error in
      tests/fax/pjsip/directmedia_reinvite_t38 (Reported by Matt
      Jordan)
 * ASTERISK-25593 - fastagi: record file closed after sending
      result (Reported by Kevin Harwell)
 * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
      it's assumed to (Reported by Walter Doekes)
 * ASTERISK-25590 - CLI Usage info for 'pjsip send notify'
      references incorrect config (Reported by Corey Farrell)
 * ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
      (Reported by Corey Farrell)
 * ASTERISK-25575 - res_pjsip: Dynamic outbound registrations
      created via ARI are not loaded into memory on Asterisk
      start/restart (Reported by Matt Jordan)
 * ASTERISK-25545 - [patch] translation module gets cached not
      joint format (Reported by Alexander Traud)
 * ASTERISK-25573 - [patch] H.264 format attribute module: resets
      whole SDP (Reported by Alexander Traud)
 * ASTERISK-24958 - Forwarding loop detection inhibits certain
      desirable scenarios (Reported by Mark Michelson)
 * ASTERISK-25561 - app_queue.c line 6503 (try_calling): mutex
      'qe->chan' freed more times than we've locked! (Reported by Alec
      Davis)
 * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
      Joshua Colp)
 * ASTERISK-25160 - [patch] Opus Codec: SIP/SDP line fmtp missing
      when called internally (Reported by Alexander Traud)
 * ASTERISK-25535 - [patch] format creation on module load instead
      of cache (Reported by Alexander Traud)
 * ASTERISK-25449 - main/sched: Regression introduced by
      5c713fdf18f causes erroneous duplicate RTCP messages; other
      potential scheduling issues in chan_sip/chan_skinny (Reported by
      Matt Jordan)
 * ASTERISK-25546 - threadpool: Race condition between idle timeout
      and activation (Reported by Joshua Colp)
 * ASTERISK-25537 - [patch] format-attribute module: RFC or
      internal defaults? (Reported by Alexander Traud)
 * ASTERISK-25533 - [patch] buffer for ast_format_cap_get_names
      only 64 bytes (Reported by Alexander Traud)
 * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
      the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
      Doekes)
 * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
      Walter Doekes)
 * ASTERISK-24779 - Passthrough OPUS codec not working with
      chan_pjsip (Reported by PowerPBX)
 * ASTERISK-25522 - ARI: Crash when creating channel via ARI
      originate with requesting channel (Reported by Matt Jordan)
 * ASTERISK-25434 - Compiler flags not reported in 'core show
      settings' despite usage during compilation (Reported by Rusty
      Newton)
 * ASTERISK-24106 - WebSockets Automatically decides what driver it
      will use  (Reported by Andrew Nagy)
 * ASTERISK-25513 - Crash: malloc failed with high load of
      subscriptions. (Reported by John Bigelow)
 * ASTERISK-25505 - res_pjsip_pubsub: Crash on off-nominal when UAS
      dialog can't be created (Reported by Joshua Colp)
 * ASTERISK-24543 - Asterisk 13 responds to SIP Invite with all
      possible codecs configured for peer as opposed to intersection
      of configured codecs and offered codecs (Reported by Taylor
      Hawkes)
 * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
      bounds and missing paren issues (Reported by George Joseph)
 * ASTERISK-25485 - res_pjsip_outbound_registration: registration
      stops due to 400 response (Reported by Kevin Harwell)
 * ASTERISK-25486 - res_pjsip: Fix deadlock when validating URIs
      (Reported by Joshua Colp)
 * ASTERISK-7803 - [patch] Update the maximum packetization values
      in frame.c (Reported by dea)
 * ASTERISK-25484 - [patch] autoframing=yes has no effect (Reported
      by Alexander Traud)
 * ASTERISK-25461 - Nested dialplan #includes don't work as
      expected. (Reported by Richard Mudgett)
 * ASTERISK-25455 - Deadlock of PJSIP realtime over
      res_config_pgsql  (Reported by mdu113)
 * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
      (Reported by Olle Johansson)
 * ASTERISK-25435 - Asterisk periodically hangs. UDP Recv-Q greatly
      exceeds zero. (Reported by Dmitriy Serov)
 * ASTERISK-25451 - Broken video - erased rtp marker bit (Reported
      by Stefan Engström)
 * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
      exist in AstDB (Reported by Andrew Nagy)
 * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
      parsing (Reported by ffs)
 * ASTERISK-25404 - segfault/crash in chan_pjsip_hangup ... at
      chan_pjsip.c (Reported by Chet Stevens)
 * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
      (Reported by Bojan Nemčić)
 * ASTERISK-25441 - Deadlock in res_sorcery_memory_cache. (Reported
      by Richard Mudgett)
 * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
      ICE is not enabled (Reported by Joshua Colp)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25618 - res_pjsip:  Check for readability of TLS files
      at startup (Reported by George Joseph)
 * ASTERISK-25572 - Endpoints: Add StatsD stats for Asterisk
      endpoints (Reported by Matt Jordan)
 * ASTERISK-25571 - PJSIP: Add StatsD stats for some common PJSIP
      objects (Reported by Matt Jordan)
 * ASTERISK-25518 - taskprocessor: Add high water mark (Reported by
      Jonathan Rose)
 * ASTERISK-25477 - pjsip show "command" like [criteria] (Reported
      by Bryant Zimmerman)
 * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
      configure (Reported by Badalian Vyacheslav)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.7.0

Thank you for your continued support of Asterisk!
2016-02-07 09:13:34 +00:00
jnemeth
97457ca749 Update to Asterisk 11.21.1: this is mainly a bug patch update plus
fixes for AST-2016-001, AST-2016-002, and AST-2016-003.  Also some
pkglinting.

----- 11.21.1

The Asterisk Development Team has announced security releases for Certified
Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.

The release of these versions resolves the following security vulnerabilities:

* AST-2016-001: BEAST vulnerability in HTTP server

  The Asterisk HTTP server currently has a default configuration which allows
  the BEAST vulnerability to be exploited if the TLS functionality is enabled.
  This can allow a man-in-the-middle attack to decrypt data passing through it.

* AST-2016-002: File descriptor exhaustion in chan_sip

  Setting the sip.conf timert1 value to a value higher than 1245 can cause an
  integer overflow and result in large retransmit timeout times. These large
  timeout values hold system file descriptors hostage and can cause the system
  to run out of file descriptors.

* AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.

  If no UDPTL packets are lost there is no problem. However, a lost packet
  causes Asterisk to use the available error correcting redundancy packets. If
  those redundancy packets have zero length then Asterisk uses an uninitialized
  buffer pointer and length value which can cause invalid memory accesses later
  when the packet is copied.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
 * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
 * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf

Thank you for your continued support of Asterisk!

----- 11.21.0

The Asterisk Development Team has announced the release of Asterisk 11.21.0.

The release of Asterisk 11.21.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25640 - pbx: Deadlock on features reload and state
      change hint. (Reported by Krzysztof Trempala)
 * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
      thread of asterisk is not released (Reported by Hiroaki Komatsu)
 * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
      Corey Farrell)
 * ASTERISK-25609 - [patch]Asterisk may crash when calling
      ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
 * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
      answer waiting time is more than ~7sec (Reported by Aleksei
      Kulakov)
 * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
      (Reported by Alexander Traud)
 * ASTERISK-25616 - Warning with a Codec Module which supports PLC
      with FEC (Reported by Alexander Traud)
 * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
      Dudás József)
 * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
      that module installed (Reported by Ben Langfeld)
 * ASTERISK-25476 - chan_sip loses registrations after a while
      (Reported by Michael Keuter)
 * ASTERISK-25593 - fastagi: record file closed after sending
      result (Reported by Kevin Harwell)
 * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
      it's assumed to (Reported by Walter Doekes)
 * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
      Joshua Colp)
 * ASTERISK-25449 - main/sched: Regression introduced by
      5c713fdf18f causes erroneous duplicate RTCP messages; other
      potential scheduling issues in chan_sip/chan_skinny (Reported by
      Matt Jordan)
 * ASTERISK-25537 - [patch] format-attribute module: RFC or
      internal defaults? (Reported by Alexander Traud)
 * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
      the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
      Doekes)
 * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
      Walter Doekes)
 * ASTERISK-25434 - Compiler flags not reported in 'core show
      settings' despite usage during compilation (Reported by Rusty
      Newton)
 * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
      bounds and missing paren issues (Reported by George Joseph)
 * ASTERISK-7803 - [patch] Update the maximum packetization values
      in frame.c (Reported by dea)
 * ASTERISK-25461 - Nested dialplan #includes don't work as
      expected. (Reported by Richard Mudgett)
 * ASTERISK-25455 - Deadlock of PJSIP realtime over
      res_config_pgsql  (Reported by mdu113)
 * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
      (Reported by Olle Johansson)
 * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
      exist in AstDB (Reported by Andrew Nagy)
 * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
      parsing (Reported by ffs)
 * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
      (Reported by Bojan Nemčić)
 * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
      ICE is not enabled (Reported by Joshua Colp)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
      configure (Reported by Badalian Vyacheslav)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0

Thank you for your continued support of Asterisk!
2016-02-07 08:18:43 +00:00
richard
9a24cc8ca1 Recent versions of Illumos implement flock() so add an additional guard
for SOLARIS.
2016-02-01 17:10:38 +00:00
dholland
a3d1589e21 Whitespace. 2015-12-29 04:54:34 +00:00
dholland
f13d6671da Fix missing/broken rcsids. 2015-12-29 04:04:26 +00:00
dholland
a333fd0c41 Modernize rc scripts, for PR 18681. Add hfaxd.sh, faxq.sh; remove
hylafax.sh.
2015-12-29 00:05:54 +00:00
wiedi
5c3729d188 Link network libs on SunOS 2015-12-12 20:51:13 +00:00
jnemeth
01535f173d add and enable asterisk13 2015-12-05 23:43:54 +00:00
jnemeth
8417dd3e35 Initial import of Asterisk 13. It has been tested to compile
and run, but not a lot of functional testing.  This does not have
the new PJSIP, which will be coming in a followup commit.  This
also does not have the patches for compiling with Clang.  For
upgrading instructions, please see:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13

-----

The Asterisk Development Team is pleased to announce the release
of Asterisk 13.0.0.

Asterisk 13 is the next major release series of Asterisk. It is a
Long Term Support (LTS) release, similar to Asterisk 11. For more
information about support time lines for Asterisk releases, see
the Asterisk versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 13, please
see the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13

A short list of new features includes:

* Asterisk security events are now provided via AMI, allowing end
  users to monitor their Asterisk system in real time for security
  related issues.

* Both AMI and ARI now allow external systems to control the state
  of a mailbox.  Using AMI actions or ARI resources, external
  systems can programmatically trigger Message Waiting Indicators
  (MWI) on subscribed phones. This is of particular use to those
  who want to build their own VoiceMail application using ARI.

* ARI now supports the reception/transmission of out of call text
  messages using any supported channel driver/protocol stack through
  ARI. Users receive out of call text messages as JSON events over
  the ARI websocket connection, and can send out of call text
  messages using HTTP requests.

* The PJSIP stack now supports RFC 4662 Resource Lists, allowing
  Asterisk to act as a Resource List Server. This includes defining
  lists of presence state, mailbox state, or lists of presence
  state/mailbox state; managing subscriptions to lists; and batched
  delivery of NOTIFY requests to subscribers.

* The PJSIP stack can now be used as a means of distributing device
  state or mailbox state via PUBLISH requests to other Asterisk
  instances.  This is analogous to Asterisk's clustering support
  using XMPP or Corosync; unlike existing clustering mechanisms,
  using the PJSIP stack to perform the distribution of state does
  not rely on another daemon or server to perform the work.

And much more!

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation

A full list of all new features can also be found in the CHANGES file:

http://svnview.digium.com/svn/asterisk/branches/13/CHANGES

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.1.0.

The release of Asterisk 13.1.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-24554 - AMI/ARI: Generate events on connected line
      changes (Reported by Matt Jordan)

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

Improvements made in this release:
-----------------------------------
 * ASTERISK-24279 - Documentation: Clarify the behaviour of the CDR
      property 'unanswered' (Reported by Matt Jordan)
 * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
      column in the cel_odbc module (Reported by Etienne Lessard)
 * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
      recordings (Reported by Ben Smithurst)
 * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
      lookups (Reported by Birger "WIMPy" Harzenetter)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.1.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.2.0.

The release of Asterisk 13.2.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

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

Improvements made in this release:
-----------------------------------
 * ASTERISK-24552 - ARI: Allow associating a channel as an
      initiator of an Origination for record keeping purposes
      (Reported by Matt Jordan)
 * ASTERISK-24553 - ARI/AMI: Include language in standard channel
      snapshot output (Reported by Matt Jordan)
 * ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by
      Matt Jordan)
 * ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for
      connection-oriented transports. (Reported by Matt Jordan)
 * ASTERISK-24412 - [patch]Incomplete channel originate/continue
      handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
      Israel))
 * ASTERISK-24678 - [PATCH] Added atxfer* settings to
      features.conf.sample (Reported by Niklas Larsson)
 * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
      by cloos)
 * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by
      Dan Jenkins)
 * ASTERISK-24316 - For httpd server, need option to define server
      name for security purposes (Reported by Andrew Nagy)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.2.1.

The release of Asterisk 13.2.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

* --- pjsip: resolve compatibility problem with ast_sip_session
  (Closes issue ASTERISK-24941. Reported by Matt Jordan)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.1

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.3.0.

The release of Asterisk 13.3.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a
      channel (Reported by Matt Jordan)
 * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
      (Reported by Dwayne Hubbard)

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

Improvements made in this release:
-----------------------------------
 * ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes
      (Reported by Ben Merrills)
 * ASTERISK-24811 - asterisk-publication sorcery object does not
      use realtime (Reported by Matt Hoskins)
 * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
      Couldn't find mailbox %s in context (Reported by Graham Barnett)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.3.1.

The release of Asterisk 13.3.1 resolves an issue reported by the
community and would have not been possible without your participation.
Thank you!

The following is the issue resolved in this release:

* --- pjsip: resolve compatibility problem with ast_sip_seesion
  (Closes issue ASTERISK-24941. Reported by Matt Jordan)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.1

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.4.0.

The release of Asterisk 13.4.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-24922 - ARI: Add the ability to intercept hold and
      raise an event (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25112 - Logger: Configuration settings are not reset to
      default during reload. (Reported by Corey Farrell)
 * ASTERISK-24944 - main/audiohook.c change prevents G722 call
      recording (Reported by Ronald Raikes)
 * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
      or more digits (Reported by Makoto Dei)
 * ASTERISK-25086 - [patch]PJSIP crashes if endpoint missing in
      Dial() (Reported by snuffy)
 * ASTERISK-25089 - res_pjsip_config_wizard: Variable specified in
      templates aren't being processed correctly (Reported by George
      Joseph)
 * ASTERISK-25090 - CLI core show channel truncates cdr variables
      (Reported by snuffy)
 * ASTERISK-25085 - [patch]Potential crash after unload of
      func_periodic_hook or test_message (Reported by Corey Farrell)
 * ASTERISK-25083 - Message.c: Message channel becomes saturated
      with frames leading to spammy log messages (Reported by Jonathan
      Rose)
 * ASTERISK-25082 - Asterisk deletes message after doing a playback
      of an INBOX message using ast_vm_play when the Old folder is
      full for that mailbox. (Reported by Jonathan Rose)
 * ASTERISK-25041 - [patch]Broken column type checking in
      res_config_mysql addon (Reported by Alexandre Fournier)
 * ASTERISK-21893 - Segfault after call hangup, in
      ast_channel_hangupcause_set, at channel_internal_api.c (Reported
      by Alexandr Gordeev)
 * ASTERISK-25074 - Regression: Recent clang-related change broke
      cross compiling of Asterisk (Reported by Sebastian Kemper)
 * ASTERISK-25042 - asterisk.conf options override command-line
      options. (Reported by Corey Farrell)
 * ASTERISK-24442 - Outgoing call files don't work properly when
      set in the future (Reported by tootai)
 * ASTERISK-25057 - res_pjsip_pubsub: Crash in send_notify due to
      invalid root pointer in sub_tree (Reported by Matt Jordan)
 * ASTERISK-24938 - ARI Snoop Channel results in excessive
      escalating CPU usage (Reported by George Ladoff)
 * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
      ignore ISDN RESTART requests. (Reported by Richard Mudgett)
 * ASTERISK-25003 - Asterisk crashes on attended transfer (using
      feature) (Reported by Artem Volodin)
 * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
      contain waiting time (Reported by Etienne Lessard)
 * ASTERISK-25027 - Build System: Many ARI modules are missing
      dependencies. (Reported by Corey Farrell)
 * ASTERISK-25061 - pbx_config: Register manager actions with
      module version of macro. (Reported by Corey Farrell)
 * ASTERISK-25025 - Periodic crashes (in
      ast_channel_snapshot_create at stasis_channels.c) with Certified
      Asterisk 13. (Reported by Chet Stevens)
 * ASTERISK-25053 - Unit test category /main/presence missing
      trailing slash. (Reported by Corey Farrell)
 * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
      not respected, failover between DSNs doesn't work (Reported by
      JoshE)
 * ASTERISK-25054 - Formats interface's cannot be unregistered,
      needs to hold modules until shutdown. (Reported by Corey
      Farrell)
 * ASTERISK-24896 - [patch] Using force black background leads to
      colours not being reset (Reported by dant)
 * ASTERISK-25033 - Asterisk 13 (branch head) won't compile without
      PJSip (Reported by Peter Whisker)
 * ASTERISK-25028 - Build System: Unneeded defines in
      asterisk/buildopts.h (Reported by Corey Farrell)
 * ASTERISK-25048 - Astobj2: Initialization order wrong when both
      refdebug and AO2_DEBUG are both enabled. (Reported by Corey
      Farrell)
 * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
      cause code 44 after some time. (Reported by Denis Alberto
      Martinez)
 * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25037 - res_pjsip_outbound_registration: Potential
      crash in off-nominal failure case when sending message (Reported
      by Joshua Colp)
 * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
      (Reported by Steve Davies)
 * ASTERISK-22790 - check_modem_rate() may return incorrect rate
      for V.27 (Reported by not here)
 * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
      to minrate=2400, then res_fax refuse to load (Reported by David
      Brillert)
 * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
      which is disallowed in res_fax's check_modem_rate (Reported by
      Matt Jordan)
 * ASTERISK-24996 - chan_pjsip: Creating Channel Causes Asterisk to
      Crash When Duplicate AOR Sections Exist in pjsip.conf (Reported
      by Ashley Sanders)
 * ASTERISK-25020 - Mismatched response to outgoing REGISTER
      request (Reported by Mark Michelson)
 * ASTERISK-25018 - pjsip show endpoints crashes asterisk when
      qualified aors present (Reported by Ivan Poddubny)
 * ASTERISK-24749 - ConfBridge: Wrong language on playing
      conf-hasjoin and conf-hasleft when played to bridge (Reported by
      Philippe Bolduc)
 * ASTERISK-24845 - pjsip send notify not working with Cisco phone
      (Reported by Carl Fortin)
 * ASTERISK-25004 - Crash in authenticated reinvite after
      originated T.38 FAX (Reported by Mark Michelson)
 * ASTERISK-24999 - PJSIP crashes with malformed contact line
      (Reported by snuffy)
 * ASTERISK-24998 - res_corosync:  res_corosync tries to load even
      if res_corosync.conf is missing (Reported by George Joseph)
 * ASTERISK-24997 - Astobj2: Some callers of __adjust_lock do not
      pre-check the object (Reported by Corey Farrell)
 * ASTERISK-24982 - res_pjsip_mwi: Unsolicited MWI NOTIFY only sent
      on mailbox changes (Reported by Joshua Colp)
 * ASTERISK-24991 - Check for ao2_alloc failure in
      __ast_channel_internal_alloc (Reported by Corey Farrell)
 * ASTERISK-24895 - After hangup on the side of the ISDN network no
      HangupRequest event comes for the dahdi channel. (Reported by
      Andrew Zherdin)
 * ASTERISK-24977 - Contacts that don't use qualify are being
      marked as unavailable (Reported by George Joseph)
 * ASTERISK-24774 - Segfault in ast_context_destroy with
      extensions.ael and extensions.conf (Reported by Corey Farrell)
 * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when
      channels have multiple native formats (Reported by Matt Jordan)
 * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
      to Fail (Reported by Ashley Sanders)
 * ASTERISK-24958 - Forwarding loop detection inhibits certain
      desirable scenarios (Reported by Mark Michelson)
 * ASTERISK-24863 - res_pjsip: No endpoint events raised via AMI
      when contacts cannot be reached/qualified (Reported by Dmitriy
      Serov)
 * ASTERISK-24869 - Asterisk segfaults on DAHDI attended transfer
      due to application (appl) being NULL on unbridged channel
      (Reported by viniciusfontes)
 * ASTERISK-24970 - Crash in res_pjsip_pubsub handling of failed
      notify (Reported by Scott Griepentrog)
 * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-24954 - Git migration: Asterisk version numbers are
      incompatible with the Test Suite (Reported by Matt Jordan)
 * ASTERISK-17608 - func_aes.so cannot be loaded if res_crypto /
      openssl not compiled (Reported by Warren Selby)
 * ASTERISK-24928 - [patch]t38_udptl_maxdatagram in pjsip.conf not
      honored (Reported by Juergen Spies)
 * ASTERISK-24835 - Early Media Not working with Chan SIP and
      Asterisk 13 (Reported by Andrew Nagy)
 * ASTERISK-21777 - Asterisk tries to transcode video instead of
      audio (Reported by Nick Ruggles)
 * ASTERISK-24380 - core: Native formats are set to h264 with
      certain audio/video codec configuration, resulting in path
      translation WARNINGs (Reported by Matt Jordan)
 * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
      into account (Reported by Frederic Van Espen)
 * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
      short (Reported by Y Ateya)
 * ASTERISK-24935 - res_pjsip_phoneprov_provider: Fix leaked
      OBJ_MULTIPLE iterator. (Reported by Corey Farrell)
 * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
      (Reported by Vadim)
 * ASTERISK-24933 - T38 fails negotiation (Reported by Jonathan
      Rose)
 * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
      byte prefix bug (Reported by Matt Jordan)
 * ASTERISK-21211 - chan_iax2 - unprotected access of
      iaxs[peer->callno] potentially results in segfault (Reported by
      Jaco Kroon)
 * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
      (Reported by Christoph Timm)
 * ASTERISK-24782 - StasisEnd event not present for channel that
      was swapped out for another after completing attended transfer
      (Reported by John Bigelow)
 * ASTERISK-24910 - "timer=no" and "timer=required" settings in
      pjsip.conf fail (Reported by Ray Crumrine)
 * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
      (Reported by Jeffrey C. Ollie)
 * ASTERISK-24914 - Division by zero in file.c when playback of
      voicemail with video as h264 (Reported by Marcello Ceschia)
 * ASTERISK-24899 - Parking fall-through behavior different in 13
      (Reported by Malcolm Davenport)
 * ASTERISK-24937 - [patch]res_pjsip_messaging: Messages may be
      sent out of order (Reported by Mark Michelson)
 * ASTERISK-24920 - Asterisk handles duplicate SIP requests as if
      they were each a new request (Reported by Mark Michelson)
 * ASTERISK-24857 - [patch] "timing test", pjsip incoming/outgoing
      calls, voicemail prompts and recordings all fail when using the
      kqueue timer source on FreeBSD 10.x (Reported by Justin T.
      Gibbs)
 * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
      detection in ast_malloc (Reported by Timo Teräs)
 * ASTERISK-24142 - CCSS: crash during shutdown due to device
      lookup in destroyed container (Reported by David Brillert)
 * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
      core restart now (Reported by Peter Katzmann)
 * ASTERISK-24805 - [patch] - ASAN: Race condition
      (heap-use-after-free) on asterisk closing (Reported by Badalian
      Vyacheslav)
 * ASTERISK-24881 - ast_register_atexit should only be used when
      absolutely needed (Reported by Corey Farrell)
 * ASTERISK-24731 - res_pjsip_session cannot be unloaded (Reported
      by Corey Farrell)
 * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
      (Reported by Kevin Harwell)
 * ASTERISK-14233 - [patch] Buddies are always auto-registered when
      processing the roster (Reported by Simon Arlott)
 * ASTERISK-24780 - [patch] - Buddies are always auto-registered
      when processing the roster (Reported by Simon Arlott)
 * ASTERISK-24781 - PJSIP: Unnecessary 180 Ringing messages sent
      with undesireabe consequences. (Reported by Richard Mudgett)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25044 - sorcery:  Add ability to insert a new wizard
      into an object type's list (Reported by George Joseph)
 * ASTERISK-24892 - Super Awesome Company sound prompts (Reported
      by Rusty Newton)
 * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
      Hjelm)
 * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
      (Reported by Alexander Traud)
 * ASTERISK-25045 - vector:  Add new capabilities and unit tests
      (Reported by George Joseph)
 * ASTERISK-24706 - [patch]add auto-dtmf mode for pjsip (Reported
      by yaron nahum)
 * ASTERISK-25051 - Remove unneeded uses of optional_api providers.
      (Reported by Corey Farrell)
 * ASTERISK-25040 - pbx: Improve performance of reloads by making
      hint destruction more performant (Reported by Matt Jordan)
 * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
      Diederik de Groot)
 * ASTERISK-24949 - res_pjsip_outbound_registration: Backport line
      functionality (Reported by Joshua Colp)
 * ASTERISK-24965 - cel_pgsql - log_error string references CDR
      instead of CEL (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-24918 - pjsip: add CLI options to display global and
      system configuration (Reported by Scott Griepentrog)
 * ASTERISK-24862 - [patch] Support in-dialog OPTIONS (Reported by
      yaron nahum)
 * ASTERISK-24802 - stasis: set a channel variable on websocket
      disconnect error (Reported by Kevin Harwell)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.4.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.5.0.

The release of Asterisk 13.5.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Improvements made in this release:
-----------------------------------
 * ASTERISK-25256 - [patch]Post AMI VarSet to empty string events
      when Asterisk deletes a dialplan variable. (Reported by Richard
      Mudgett)
 * ASTERISK-25067 - Sorcery Caching: Implement a new caching module
      (Reported by Matt Jordan)
 * ASTERISK-25040 - pbx: Improve performance of reloads by making
      hint destruction more performant (Reported by Matt Jordan)
 * ASTERISK-25114 - res_pjsip:  Add AMI etents for chan_pjsip
      contact lifecycle changes (Reported by George Joseph)
 * ASTERISK-25072 - res_pjsip_outbound_registration: line
      functionality. Additional check for using the request URI
      (Reported by Dmitriy Serov)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25250 - chan_sip - Despite the channel being answered,
      caller on a call established via Local channel continues to hear
      ringback (Reported by Etienne Lessard)
 * ASTERISK-25253 - confbridge volume options and other volume
      controls such as func_volume don't work (Reported by Dmitriy
      Serov)
 * ASTERISK-25247 - choppy audio when spying on a g722 channel,
      chan_sip or chan_pjsip (Reported by hristo)
 * ASTERISK-24867 - Docs for 'e' option in ResetCDR say to use
      CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty
      Newton)
 * ASTERISK-24853 - Documentation claims chan_sip outbound
      registrations support WS or WSS as valid transports (not true)
      (Reported by PSDK)
 * ASTERISK-25242 - PJSIP: No audio when Asterisk inside NAT and
      endpoints outside NAT - implement functionality similar to
      chan_sip 'rtpkeepalive'? (Reported by Mark Michelson)
 * ASTERISK-25258 - chan_pjsip: Incorrect format switch on received
      RTP packet (Reported by Joshua Colp)
 * ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
      force_restart_unavailable_chans in wrong scope (Reported by
      Patric Marschall)
 * ASTERISK-24934 - [patch]Asterisk manager output does not escape
      control characters (Reported by warren smith)
 * ASTERISK-25255 - Missing AMI VarSet events when setting to an
      empty string. (Reported by Richard Mudgett)
 * ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to an
      empty string before Park. (Reported by Richard Mudgett)
 * ASTERISK-25183 - PJSIP: Crash on NULL channel in
      chan_pjsip_incoming_response despite previous checks for NULL
      channel (Reported by Matt Jordan)
 * ASTERISK-25201 - Crash in PJSIP distributor on already free'd
      threadpool (Reported by Matt Jordan)
 * ASTERISK-24782 - StasisEnd event not present for channel that
      was swapped out for another after completing attended transfer
      (Reported by John Bigelow)
 * ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully
      started when completing attended transfer (Reported by Joshua
      Colp)
 * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
      (Reported by Rusty Newton)
 * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
      BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
      (Reported by Dmitry Burilov)
 * ASTERISK-24550 - res_rtp_asterisk: Crash in
      ast_rtp_on_ice_complete during DTLS handshake (Reported by
      Osaulenko Alexander)
 * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
      Badalian Vyacheslav)
 * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
      by Stefan Engström)
 * ASTERISK-25127 - DTLS crashes following "Unable to cancel
      schedule ID" in dtls_srtp_check_pending (Reported by Dade
      Brandon)
 * ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in
      ast_channel_name at channel_internal_api.c (Reported by Carl
      Fortin)
 * ASTERISK-25115 - Crash related to func
      sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c
      (Reported by John Bigelow)
 * ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early
      replaces call pickup (Reported by Walter Doekes)
 * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
      (Reported by Walter Doekes)
 * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
      in rtp_engine.c (Reported by Walter Doekes)
 * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
      (Reported by Walter Doekes)
 * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
      Bad file descriptor" (Reported by Barry Chern)
 * ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
      (Reported by Corey Farrell)
 * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
      13.4 (Reported by cervajs)
 * ASTERISK-25196 - res_pjsip_nat: rewrite_contact should not be
      applied to Contact header when Record-Route headers are present
      (Reported by Mark Michelson)
 * ASTERISK-24907 - res_pjsip_outbound_registration: crash during
      unload if registration attempts are still occuring (Reported by
      Kevin Harwell)
 * ASTERISK-25204 - res_pjsip_refer: Duplicated Referred-By or
      Replaces headers on outbound INVITEs. (Reported by Mark
      Michelson)
 * ASTERISK-25171 - Early completion of feature code attended
      transfer results in intermittent one-way audio, "ghost ringing"
      and robotic sound. (Reported by Rusty Newton)
 * ASTERISK-25189 - AMI: Add Linkedid header to standard channel
      snapshot information. (Reported by Richard Mudgett)
 * ASTERISK-25172 - Crash in channels/sip/sip blind
      transfer/caller_refer_only test in
      ast_format_cap_append_from_cap during ast_request (Reported by
      Matt Jordan)
 * ASTERISK-25180 - res_pjsip_mwi: Unsolicited MWI requires reload
      (Reported by Joshua Colp)
 * ASTERISK-25182 - [patch] on CLI sip reload, new codecs get
      appended only (Reported by Alexander Traud)
 * ASTERISK-25163 - Deadlock in chan_sip between reload of sip peer
      container and MWI Stasis callback (Reported by Dmitriy Serov)
 * ASTERISK-25091 - Asterisk REST API - bridge.addChannel crash
      asterisk when calling channel hangup while adding to bridge
      (Reported by Ilya Trikoz)
 * ASTERISK-24900 - Manager event ParkedCallSwap is not documented
      (Reported by Rusty Newton)
 * ASTERISK-25162 - func_pjsip_aor: Leak of contact in iterator
      (Reported by Corey Farrell)
 * ASTERISK-25158 - res_pjsip: Add option to use AAL2 packing when
      negotiating g.726 (Reported by Kevin Harwell)
 * ASTERISK-24344 - CDR_PROP(disable) disables CDR only for first
      dialed party (Reported by Janusz Karolak)
 * ASTERISK-24443 - CDR fields (dst, dcontext) empty in transfer
      call started from Macro (Reported by Arveno Santoro)
 * ASTERISK-25154 - [patch]fromtag may need to be updatep after
      successful call dialog match (Reported by Damian Ivereigh)
 * ASTERISK-25156 - chan_pjsip’s CHAN_START cel event lacks the
      correct context and exten (Reported by cloos)
 * ASTERISK-25157 - bridging: Performing a blonde transfer does not
      result in connected line updates (Reported by Joshua Colp)
 * ASTERISK-25087 - Asterisk segfault when using Directory
      application with alias option and specific mailbox configuration
      (Reported by Chet Stevens)
 * ASTERISK-24983 - IAX deadlock between hangup and scheduled
      actions (ex. largrq) (Reported by Y Ateya)
 * ASTERISK-25096 - [patch]Segfault when registering over
      websockets with PJSIP (in ast_sockaddr_isnull at
      /include/asterisk/netsock2.h) (Reported by Josh Kitchens)
 * ASTERISK-24963 - ASAN: heap-use-after-free with PJSIP and WSS
      (Reported by Badalian Vyacheslav)
 * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
      but asterisk doesn't detect it. (Reported by ibercom)
 * ASTERISK-25094 - PBX core: Investigate thread safety issues
      (Reported by Corey Farrell)
 * ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark
      Michelson)
 * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
      | adpcm | ipc10} (Reported by Badalian Vyacheslav)
 * ASTERISK-25137 - endpoint stasis messages are delivered twice
      (Reported by Vitezslav Novy)
 * ASTERISK-25116 - res_pjsip:  Two PeerStatus AMI messages are
      sent for every status change (Reported by George Joseph)
 * ASTERISK-25131 - chan_pjsip: In-dialog authentication not
      handled. (Reported by Richard Mudgett)
 * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
      that end with ::80 (Reported by Mark Petersen)
 * ASTERISK-25122 - Large SIP packet received via pjsip over
      websocket crashes Asterisk  (Reported by Ivan Poddubny)
 * ASTERISK-25121 - Stasis: Fix unsafe use of stasis_unsubscribe in
      modules. (Reported by Corey Farrell)
 * ASTERISK-24988 - func_talkdetect: Test is bouncing sporadically
      (Reported by Joshua Colp)
 * ASTERISK-25105 - res_pjsip:  Possible incompatibility between
      qualify_timeout and pjproject-2.4 (Reported by George Joseph)
 * ASTERISK-25117 - res_mwi_external_ami: Fix manager action
      registrations. (Reported by Corey Farrell)

New Features made in this release:
-----------------------------------
 * ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by
      Joshua Colp)
 * ASTERISK-25238 - ARI: Support push configuration (Reported by
      Matt Jordan)
 * ASTERISK-25173 - ARI: Add the ability to load/reload/unload an
      Asterisk module (Reported by Matt Jordan)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.5.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 13.6.0.

The release of Asterisk 13.6.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-25377 - res_pjsip: Change default "From user" from UUID
      to something more palatable (Reported by Mark Michelson)
 * ASTERISK-25252 - ARI: Add the ability to manipulate log channels
      (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25449 - main/sched: Regression introduced by
      5c713fdf18f causes erroneous duplicate RTCP messages; other
      potential scheduling issues in chan_sip/chan_skinny (Reported by
      Matt Jordan)
 * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
      ICE is not enabled (Reported by Joshua Colp)
 * ASTERISK-25383 - Core dumps on startup and shutdown with
      MALLOC_DEBUG enabled (Reported by yaron nahum)
 * ASTERISK-25423 - Caller gets no Connected line update during
      call pickup. (Reported by Richard Mudgett)
 * ASTERISK-25305 - Dynamic logger channels can be added multiple
      times (Reported by Mark Michelson)
 * ASTERISK-25418 - On-hold channels redirected out of a bridge
      appear to still be on hold (Reported by Mark Michelson)
 * ASTERISK-25384 - Regular Asterisk crashes when using Page
      application. "user_data is NULL" (Reported by Chet Stevens)
 * ASTERISK-25407 - Asterisk fails to log to multiple syslog
      destinations (Reported by Elazar Broad)
 * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
      populated (Reported by Kevin Harwell)
 * ASTERISK-25394 - pbx: Incorrect device and presence state when
      changing hint details (Reported by Joshua Colp)
 * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
      invalid SIP (Reported by Walter Doekes)
 * ASTERISK-25399 - app_queue: AgentComplete event has wrong reason
      (Reported by Kevin Harwell)
 * ASTERISK-25185 - Segfault in app_queue on transfer scenarios
      (Reported by Etienne Lessard)
 * ASTERISK-25353 - [patch] Transcoding while different in Frame
      size = Frames lost (Reported by Alexander Traud)
 * ASTERISK-25325 - ARI PUT reload chan_sip HTTP response 404
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25390 - default_from_user can crash with certain
      configuration backends (Reported by Mark Michelson)
 * ASTERISK-25387 - res_pjsip_nat: Malformed REGISTER request
      causes NAT'd Contact header to not be rewritten (Reported by
      Matt Jordan)
 * ASTERISK-25227 - No audio at in-band announcements in ooh323
      channel (Reported by Alexandr Dranchuk)
 * ASTERISK-25369 - res_parking: ParkAndAnnounce - Inheritable
      variables aren't applied to the announcer channel (Reported by
      Jonathan Rose)
 * ASTERISK-25295 - res_pjsip crash - pjsip_uri_get_uri at
      /usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov)
 * ASTERISK-25381 - res_pjsip: AoRs deleted via ARI (or other
      mechanism) do not destroy their related contacts (Reported by
      Matt Jordan)
 * ASTERISK-25367 - pbx: Long pattern match hints may cause "core
      show hints" to crash (Reported by Joshua Colp)
 * ASTERISK-25365 - Persistent subscriptions have extra
      Content-Length/corrupted messages (Reported by Mark Michelson)
 * ASTERISK-25362 - Deadlock due to presence state callback
      (Reported by Mark Michelson)
 * ASTERISK-25356 - res_pjsip_sdp_rtp: Multiple keepalive scheduled
      items may exist (Reported by Joshua Colp)
 * ASTERISK-25355 - sched: ast_sched_del may return prematurely due
      to spurious wakeup (Reported by Joshua Colp)
 * ASTERISK-25318 -
      tests/rest_api/applications/subscribe-endpoint/nominal/resource:
      Sporadically failing (Reported by Joshua Colp)
 * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
      cause on call pickup (Reported by Joshua Colp)
 * ASTERISK-25342 - res_pjsip: Repeated usage of pj_gethostip may
      block (Reported by Joshua Colp)
 * ASTERISK-25341 - bridge: Hangups may get lost when executing
      actions (Reported by Joshua Colp)
 * ASTERISK-25339 - res_pjsip: Empty "auth" sections from
      non-config backgrounds are interpreted as valid (Reported by
      Matt Jordan)
 * ASTERISK-25215 - Differences in queue.log between Set
      QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
      Gaetz)
 * ASTERISK-25322 - Crash occurs when using MixMonitor with t() or
      r() options. (Reported by Richard Mudgett)
 * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
      for wrong or non existent peer on invite (Reported by Kevin
      Harwell)
 * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
      tones. (Reported by Richard Mudgett)
 * ASTERISK-25312 - res_http_websocket: Terminate connection on
      fatal cases (Reported by Joshua Colp)
 * ASTERISK-25306 - Persistent subscriptions can save multiple SIP
      messages at once, leading to potential crashes. (Reported by
      Mark Michelson)
 * ASTERISK-25309 - [patch] iLBC 20 advertised (Reported by
      Alexander Traud)
 * ASTERISK-25304 - res_pjsip: XML sanitization may write past
      buffer (Reported by Joshua Colp)
 * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
      Firefox 39 - add ECDH support and fallback to prime256v1
      (Reported by Stefan Engström)
 * ASTERISK-25296 - RTP performance issue with several channel
      drivers. (Reported by Richard Mudgett)
 * ASTERISK-25297 - Crashes running
      channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
      (Reported by Richard Mudgett)
 * ASTERISK-25292 - Testuite:
      tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
      (Reported by Kevin Harwell)
 * ASTERISK-25271 - Parking & blind transfer: Transferer channel
      not hung up if no MOH (Reported by Kevin Harwell)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24870 - ARI: Subscriptions to bridges generally not
      super useful (Reported by Matt Jordan)
 * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
      defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.6.0

Thank you for your continued support of Asterisk!
2015-12-05 23:42:44 +00:00
jnemeth
e2320c9334 cat ../comment 2015-12-05 23:29:05 +00:00
adam
7f3b4730ad Extend PYTHON_VERSIONS_INCOMPATIBLE to 35 2015-12-05 21:25:27 +00:00
jnemeth
64380a54c9 add information about the version, requested by gdt@ 2015-11-25 13:15:40 +00:00
jperkin
7480bda70b Remove mk/find-prefix.mk usage from the comms category.
The find-prefix infrastructure was required in a pkgviews world where
packages installed from pkgsrc could have different installation
prefixes, and this was a way for a dependency prefix to be determined.

Now that pkgviews has been removed there is no longer any need for the
overhead of this infrastructure.  Instead we use BUILDLINK_PREFIX.pkg
for dependencies pulled in via buildlink, or LOCALBASE/PREFIX where the
dependency is coming from pkgsrc.

Provides a reasonable performance win due to the reduction of `pkg_info
-qp` calls, some of which were redundant anyway as they were duplicating
the same information provided by BUILDLINK_PREFIX.pkg.
2015-11-25 12:48:55 +00:00
dholland
60efa5863f Fix openbsd build failure. 2015-11-07 23:47:52 +00:00
dholland
dc0bc29af9 fix openbsd/bitrig build 2015-11-07 23:43:56 +00:00
dholland
27987c9b9c Take out upstream's --traditional-cpp for MacOS as it breaks the build,
even on PPC (old) MacOS.
2015-11-07 23:39:15 +00:00
dholland
8bb68a98e6 add configurations for openbsd, bitrig, macos 2015-11-07 23:26:34 +00:00
dholland
fc401a32fd Use termios, not sgtty.h. Always. 2015-11-07 23:20:59 +00:00
dholland
a17f222a03 Prevent cmake from finding glib. 2015-11-07 23:16:18 +00:00
agc
ad1e2a0a92 Add SHA512 digests for distfiles for comms category
Existing SHA1 digests verified, all found to be the same on the
machine holding the existing distfiles (morden).  Existing SHA1
digests retained for now as an audit trail.
2015-11-03 01:34:52 +00:00
tnn
adfd856e04 extraneous parenthesis crept in in Darwin conditional 2015-11-02 12:02:23 +00:00
tnn
4fc361a10a appease pkglint 2015-11-02 00:34:04 +00:00
tnn
2967e94da1 Use ${COMPILER_INCLUDE_DIRS} instead of hardcoded /usr/include 2015-11-02 00:03:59 +00:00
jnemeth
0d179dc23c Update Asterisk to 11.20.0: this is mainly a bug fix release.
pkgsrc changes:
- from joerg@
  - srtp support
  - new asterisk-config option to control installing of sample config files
  - manifest.xml for Solaris' SMF
  - various bugfixes, some reworked by myself
- backport kqueue timer update from Asterisk 13

-----

The Asterisk Development Team has announced the release of Asterisk 11.20.0.

The release of Asterisk 11.20.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25449 - main/sched: Regression introduced by
      5c713fdf18f causes erroneous duplicate RTCP messages; other
      potential scheduling issues in chan_sip/chan_skinny (Reported by
      Matt Jordan)
 * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
      ICE is not enabled (Reported by Joshua Colp)
 * ASTERISK-25427 - Callerid change does not always emit
      NewCallerid AMI event (Reported by Ivan Poddubny)
 * ASTERISK-25407 - Asterisk fails to log to multiple syslog
      destinations (Reported by Elazar Broad)
 * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
      populated (Reported by Kevin Harwell)
 * ASTERISK-25394 - pbx: Incorrect device and presence state when
      changing hint details (Reported by Joshua Colp)
 * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
      invalid SIP (Reported by Walter Doekes)
 * ASTERISK-25353 - [patch] Transcoding while different in Frame
      size = Frames lost (Reported by Alexander Traud)
 * ASTERISK-25227 - No audio at in-band announcements in ooh323
      channel (Reported by Alexandr Dranchuk)
 * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
      cause on call pickup (Reported by Joshua Colp)
 * ASTERISK-25215 - Differences in queue.log between Set
      QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
      Gaetz)
 * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
      for wrong or non existent peer on invite (Reported by Kevin
      Harwell)
 * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
      tones. (Reported by Richard Mudgett)
 * ASTERISK-25312 - res_http_websocket: Terminate connection on
      fatal cases (Reported by Joshua Colp)
 * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
      Firefox 39 - add ECDH support and fallback to prime256v1
      (Reported by Stefan Engström)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
      defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0

Thank you for your continued support of Asterisk!
2015-10-27 08:49:01 +00:00
ryoon
b141232e29 Recursive revbump from textproc/icu 2015-10-10 01:57:50 +00:00
tnn
3e1c48dfa7 replace optional socks5 dependencies with net/dante 2015-09-30 08:25:37 +00:00
joerg
fbcea479cd Fix inline use. 2015-09-14 13:30:03 +00:00
jnemeth
07ec558cad asterisk10 is history 2015-09-14 09:06:39 +00:00
jnemeth
f923197bdb Remove Asterisk 10.x package as it is two years past EOL as per pkgsrc-users. 2015-09-14 09:05:57 +00:00
joerg
a61984c205 Add srtp. 2015-09-06 14:02:29 +00:00
joerg
fa21eb32cd Add srtp-1.4.4, an implementation of Secure RTP. 2015-09-06 14:02:08 +00:00
wiz
8afd3739fd Mark as not-for-python-3.x.
UnicodeDecodeError: 'ascii' codec can't decode byte 0xc4 in position 1648: ordinal not in range(128)
2015-08-28 10:24:23 +00:00
khorben
1ed777fd31 Package DeforaOS Phone 0.4.3
This release brings:
- fewer dependencies (both "purple" and "sofia-sip" modem backends are now
  maintained externally, likewise for the "locker" plug-in)
- easier integration of third-party extensions (with pkg-config)
- improvements to the user interface
- spanish translation
- minor bugfixes
2015-08-24 23:46:03 +00:00
leot
39eb673cc5 Use egg.mk. Bump PKGREVISION. 2015-08-23 20:20:03 +00:00
is
371bb45b14 There were a few places where time_t was passed to printf-like functions,
but the format string specifies %d.
As all of them are time differences, and a fax transmission shouldn't
need more than 2^31 (normally not even 2^15) seconds, cast to (int),
like already in a few other places.
Needed because sizeof(time_t) > sizeof(int) in NetBSD-6 and later.
2015-08-21 11:08:36 +00:00
is
fc4344267b Workaround for NetBSD-6, but problem not understood: sendfax would
overflow the modem with data when FLOW_HARD (FLOWHARD|FLOW_SOFT) was
used.
2015-08-21 11:03:43 +00:00
wiz
c7383780db Bump all packages that depend on curses.bui* or terminfo.bui* since they
might incur ncurses dependencies on some platforms, and ncurses just bumped
its shlib.
Some packages were bumped twice now, sorry for that.
2015-08-18 07:31:00 +00:00
leot
2c009310ef Update comms/py-gammu to py-gammu-2.3.
ok wiz@

pkgsrc changes:
* No longer use Makefile.common now that py-gammu is released as a separate
  package by upstream too.

Changes:
2.3
===
* License changed tp GPL version 2 or later.
* Documentation improvements.

2.2
===
* Documentation improvements.
* Code cleanups.

2.1
===
* Include data required for tests in tarball.
* Include NEWS.rst in tarball.
* Fixed possible crash when changing debug file.
* Fixed various errors found by coverity.

2.0
===
* Separate Python module.
* Compiles using distutils.
* Support Python 3.
2015-08-17 16:46:11 +00:00
leot
36d1dbad54 Update comms/gammu to gammu-1.36.4.
ok wiz@.

pkgsrc changes:
 * Now comms/gammu depends on devel/libusb1 (instead of devel/libusb)
 * Get rid of Makefile.common: it is no more needed now that comms/py-gammu is
   distribuited also upstream as a separate package.

Changes:
20150814 - 1.36.4

[-] * Use advisory locking to prevent two Gammu instances share one device.
[!] * Include child process stdout and stderr in SMSD logs to ease debugging.
[-] * Fix string quoting with ODBC driver.
[+] * Added RunOnSent option to SMSD.
[+] * Store message reference in outbox in files SMSD.
[-] * Improved C API documentation in manual.

20150707 - 1.36.3

[-] * Updated list of GSM country codes and networks.
[-] * Fixed bash completition install path (Ville Skyttä).
[-] * Better logging of delivery report failures in SMSD.
[-] * Improved support for Huawei E3372.

20150615 - 1.36.2

[-] * Fixed compilation using MSVC.
[-] * Fix siemenssatnetmon (Daniel Glöckner).
[-] * Documentation improvements.
[-] * Fixed smsd startup with non existing folders.
[-] * Fixed possible stack overflows on Windows.

20150520 - 1.36.1

[-] * Compatibility with libdbi from git.
[-] * Fix siemenssatnetmon (Daniel Glöckner).
[-] * Fixed reconnecting to SQL server.
[+] * Don't split a surrogate pair between message segments (David Brown).

20150413 - 1.36.0

[!] * The python-gammu module is now shipped separately.
[!] * Removed usage of __TIME__ and __DATE__ macros in codebase.
[-] * Fixed encoding of special chars to iCalendar format.
[-] * Fixed decoding of priority from vTODO.
[-] * Avoid infinite loops with ignored messages.
[-] * Improved stability of checking phone SMS memory.
[-] * Fixed parsing of some backup files.

20150302 - 1.35.0

[-] * Fixed encoding of UTF-8 for higher code points.
[-] * Improved provided udev rules.
[-] * Fixed possible lock while getting network status in SMSD.
[-] * Various localization updates.

20141230 - 1.34.0

[+] * Add phone power ON/OFF function.
[!] * Removed deprecated Python modules gammu.Data and gammu.Worker.
[+] * Store network name and code in SMSD tables.
[-] * Fixed build with recent clang compiler.
[-] * Fixed several possible issues found by Coverity scan.
[-] * Fixed possible crash on SMSD startup.
[-] * Fixed decoding unicode SMS messages.
[-] * Added identification for several Nokia phones.
[-] * Fixed compilation issues on various platforms.
[-] * SMSD now honors loglevel for all logging targets.
[+] * SMSD can automatically hangup incoming calls.
[-] * Correctly detect Network errors.
2015-08-17 16:42:53 +00:00
joerg
53183d54f0 Don't use variable strings as format strings. Don't link with -lc_r on
the BSDs, use -lpthread. Accept openjdk8.
2015-08-13 20:16:22 +00:00
dholland
c88ecc7f1e Fix broken build, caused by wrapper reordering of .a files vs. -l options.
Symptom: HYLAFAX_VERSION_STRING not found while linking.
2015-08-10 05:03:36 +00:00
khorben
b20037003f Add support for CFLAGS
No functional change intended.
2015-08-09 14:55:58 +00:00
jnemeth
3e366fc9d3 quickly eliminate PKGREVISION on update 2015-08-09 04:10:01 +00:00
jnemeth
ca5fb69583 Update to Asterisk 11.19.0: this is mainly a bug fix release with
minor features

pkgsrc changes:
- new version of core sounds
- add options for SNMP and PostgreSQL from Mike Bowie in PR/49661
  and by popular demand
- add back support for menuselect personalization as that's how I was
  doing menuselect non-interactively
  - XXX need to look at a better way of doing this
- disable PJSIP for now as it doesn't work well on NetBSD from Mike Bowie

Since I added an option for PostgreSQL I also looked at adding an
option for directly using MySQL.  Turns out that all the MySQL
modules are in the addons directory and are marked as being
deprecated.  So I didn't bother.  While investigating this, I also
noted that all the pgsql modules are marked as "extended" support.
This basically means that it is supported by the community, but
there is no one person listed as being responsible who would take
the lead for maintaining them.  This basically means that they are
unsupported / low priority.  See
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States .
Also with the pgsql modules, there is no way to do a database query
from the dialplan.  Thus it is recommended to use the unixodbc
option as the modules are supported and offer the most functionality.

-----

The Asterisk Development Team has announced the release of Asterisk 11.19.0.

The release of Asterisk 11.19.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25250 - chan_sip - Despite the channel being answered,
      caller on a call established via Local channel continues to hear
      ringback (Reported by Etienne Lessard)
 * ASTERISK-25247 - choppy audio when spying on a g722 channel,
      chan_sip or chan_pjsip (Reported by hristo)
 * ASTERISK-24853 - Documentation claims chan_sip outbound
      registrations support WS or WSS as valid transports (not true)
      (Reported by PSDK)
 * ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
      force_restart_unavailable_chans in wrong scope (Reported by
      Patric Marschall)
 * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
      (Reported by Rusty Newton)
 * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
      BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
      (Reported by Dmitry Burilov)
 * ASTERISK-24550 - res_rtp_asterisk: Crash in
      ast_rtp_on_ice_complete during DTLS handshake (Reported by
      Osaulenko Alexander)
 * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
      Badalian Vyacheslav)
 * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
      by Stefan Engström)
 * ASTERISK-25127 - DTLS crashes following "Unable to cancel
      schedule ID" in dtls_srtp_check_pending (Reported by Dade
      Brandon)
 * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
      INVITE early Replace code (Reported by Walter Doekes)
 * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
      (Reported by Walter Doekes)
 * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
      in rtp_engine.c (Reported by Walter Doekes)
 * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
      (Reported by Walter Doekes)
 * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
      Bad file descriptor" (Reported by Barry Chern)
 * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
      13.4 (Reported by cervajs)
 * ASTERISK-25154 - [patch]fromtag may need to be updated after
      successful call dialog match (Reported by Damian Ivereigh)
 * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
      (Reported by Gregory Massel)
 * ASTERISK-25094 - PBX core: Investigate thread safety issues
      (Reported by Corey Farrell)
 * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
      but asterisk doesn't detect it. (Reported by ibercom)
 * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
      | adpcm | ipc10} (Reported by Badalian Vyacheslav)
 * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
      that end with ::80 (Reported by Mark Petersen)

Improvements made in this release:
-----------------------------------
 * ASTERISK-25040 - pbx: Improve performance of reloads by making
      hint destruction more performant (Reported by Matt Jordan)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 11.18.0.

The release of Asterisk 11.18.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25112 - Logger: Configuration settings are not reset to
      default during reload. (Reported by Corey Farrell)
 * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
      or more digits (Reported by Makoto Dei)
 * ASTERISK-24944 - main/audiohook.c change prevents G722 call
      recording (Reported by Ronald Raikes)
 * ASTERISK-25083 - Message.c: Message channel becomes saturated
      with frames leading to spammy log messages (Reported by Jonathan
      Rose)
 * ASTERISK-25041 - [patch]Broken column type checking in
      res_config_mysql addon (Reported by Alexandre Fournier)
 * ASTERISK-21893 - Segfault after call hangup, in
      ast_channel_hangupcause_set, at channel_internal_api.c (Reported
      by Alexandr Gordeev)
 * ASTERISK-25074 - Regression: Recent clang-related change broke
      cross compiling of Asterisk (Reported by Sebastian Kemper)
 * ASTERISK-25042 - asterisk.conf options override command-line
      options. (Reported by Corey Farrell)
 * ASTERISK-24442 - Outgoing call files don't work properly when
      set in the future (Reported by tootai)
 * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
      ignore ISDN RESTART requests. (Reported by Richard Mudgett)
 * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
      contain waiting time (Reported by Etienne Lessard)
 * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
      not respected, failover between DSNs doesn't work (Reported by
      JoshE)
 * ASTERISK-25028 - Build System: Unneeded defines in
      asterisk/buildopts.h (Reported by Corey Farrell)
 * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
      cause code 44 after some time. (Reported by Denis Alberto
      Martinez)
 * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
      (Reported by Steve Davies)
 * ASTERISK-22790 - check_modem_rate() may return incorrect rate
      for V.27 (Reported by not here)
 * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
      to minrate=2400, then res_fax refuse to load (Reported by David
      Brillert)
 * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
      which is disallowed in res_fax's check_modem_rate (Reported by
      Matt Jordan)
 * ASTERISK-24916 - Increasing memory usage when multiple reinvite
      during call (Reported by Christophe Osuna)
 * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
      redundancy (Reported by Alex)
 * ASTERISK-24749 - ConfBridge: Wrong language on playing
      conf-hasjoin and conf-hasleft when played to bridge (Reported by
      Philippe Bolduc)
 * ASTERISK-24991 - Check for ao2_alloc failure in
      __ast_channel_internal_alloc (Reported by Corey Farrell)
 * ASTERISK-24895 - After hangup on the side of the ISDN network no
      HangupRequest event comes for the dahdi channel. (Reported by
      Andrew Zherdin)
 * ASTERISK-24774 - Segfault in ast_context_destroy with
      extensions.ael and extensions.conf (Reported by Corey Farrell)
 * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
      to Fail (Reported by Ashley Sanders)
 * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-24954 - Git migration: Asterisk version numbers are
      incompatible with the Test Suite (Reported by Matt Jordan)
 * ASTERISK-21777 - Asterisk tries to transcode video instead of
      audio (Reported by Nick Ruggles)
 * ASTERISK-24380 - core: Native formats are set to h264 with
      certain audio/video codec configuration, resulting in path
      translation WARNINGs (Reported by Matt Jordan)
 * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
      into account (Reported by Frederic Van Espen)
 * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
      short (Reported by Y Ateya)
 * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
      (Reported by Vadim)
 * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
      byte prefix bug (Reported by Matt Jordan)
 * ASTERISK-21211 - chan_iax2 - unprotected access of
      iaxs[peer->callno] potentially results in segfault (Reported by
      Jaco Kroon)
 * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
      (Reported by Christoph Timm)
 * ASTERISK-24942 - Voicemail API: message is deleted when
      destination mailbox is at maxmsg (Reported by Scott Griepentrog)
 * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
      (Reported by Jeffrey C. Ollie)
 * ASTERISK-21854 - Long Asterisk-version strings display
      improperly in the 'Connected to ...' line upon remote console
      connection (Reported by klaus3000)
 * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
      detection in ast_malloc (Reported by Timo Teräs)
 * ASTERISK-24142 - CCSS: crash during shutdown due to device
      lookup in destroyed container (Reported by David Brillert)
 * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
      core restart now (Reported by Peter Katzmann)
 * ASTERISK-24805 - [patch] - ASAN: Race condition
      (heap-use-after-free) on asterisk closing (Reported by Badalian
      Vyacheslav)
 * ASTERISK-24881 - ast_register_atexit should only be used when
      absolutely needed (Reported by Corey Farrell)
 * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
      (Reported by Kevin Harwell)
 * ASTERISK-14233 - [patch] Buddies are always auto-registered when
      processing the roster (Reported by Simon Arlott)
 * ASTERISK-24780 - [patch] - Buddies are always auto-registered
      when processing the roster (Reported by Simon Arlott)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
      Hjelm)
 * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
      (Reported by Alexander Traud)
 * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
      Diederik de Groot)
 * ASTERISK-25040 - pbx: Improve performance of reloads by making
      hint destruction more performant (Reported by Matt Jordan)
 * ASTERISK-24965 - cel_pgsql - log_error string references CDR
      instead of CEL (Reported by Rodrigo Ramirez Norambuena)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0

Thank you for your continued support of Asterisk!
2015-08-09 04:07:13 +00:00
bsiegert
5dbaf10309 Add unixodbc dependency. Otherwise, the build picks up unixodbc if
installed and fails to find sql.h. Fixes PR pkg/49893. Bump revision.
2015-08-01 15:40:25 +00:00
adam
1e4ee78122 1.22.2
---
- Fix buffering for funny sample formats (namely, 24 bit), that do not
  fit nicely into 32768 bytes. Effect was a nasty endless loop where
  mpg123 needs to be externally killed.

1.22.1
---
- Fix mpg123-id3dump when writing images with funny (manipulated) MIME type.
  Stupid mistake in length computation of the fallback file extension caused
  junk from memory being appended to the filename if the pointer size
  is less than 64 bit. For 64 bit pointers (or longer) it was correct by
  accident.
- Fix pedantic build by cleaning up out123 source, also now really showing
  the encoding list in --longhelp instead of possibly, again, writing junk
  from memory in there.
- Not linking libmpg123 against libltdl anymore (bug 215).
- Update MSVC++ ports a bit to make them work again.
2015-07-24 10:33:27 +00:00
youri
80faca6f0a + xfce4-modemlights-plugin 2015-07-13 19:54:59 +00:00
youri
5714b55afb Import xfce4-modemlights-plugin-0.1.3.99 as comms/xfce4-modemlights-plugin.
Xfce 4 Modem Lights panel plugin is intended to simplify establishing a ppp
connection via a modem. It is primarily designed to work with the debian ppp
package and the pon/poff scripts provided by that package, but should be usable
with any scripts that create a lock file during dialing and retain it through
the connection.
2015-07-13 19:54:26 +00:00
wiz
40bbad7ac6 Comment out dependencies of the style
{perl>=5.16.6,p5-ExtUtils-ParseXS>=3.15}:../../devel/p5-ExtUtils-ParseXS
since pkgsrc enforces the newest perl version anyway, so they
should always pick perl, but sometimes (pkg_add) don't due to the
design of the {,} syntax.

No effective change for the above reason.

Ok joerg
2015-07-12 18:56:06 +00:00
joerg
11d2712a27 Remove USE_X11BASE and X11PREFIX. 2015-07-04 16:18:28 +00:00
dsainty
33a2352f72 Add oracle-jdk8 to the accepted list.
NB: I'm not game enough to do it in the freeze, but it looks like the
JVM version detection patching could be removed - it appears no longer
necessary now that Pkgsrc passes in the correct RXTX_PATH and JHOME_PATH
itself.  At any rate, adding version 8 is not required for the oracle-jdk8
build to complete smoothly.
2015-06-27 06:30:03 +00:00
khorben
8a9169ccbc Disable the "modems" test for now
This should fix the build for the moment.
2015-06-17 03:02:26 +00:00
wiz
0982effce2 Recursive PKGREVISION bump for all packages mentioning 'perl',
having a PKGNAME of p5-*, or depending such a package,
for perl-5.22.0.
2015-06-12 10:48:20 +00:00
bad
34af42955d Update jpilot to 1.8.2.
While here restore old behaviour of not alphabetically sorting memos by default.

Changes since 1.8.1:
1.8.2 - 05/18/14
 Many bug fixes
 Fixed VCard output
 Added export for B-Folders
 Added export for KeePassX
 Changed the "enye" letter in Manana an "n", got tired of it causing problems
  (Ma\303\261ana to Manana)
 Made lots of stupid code changes to make the compiler warnings go away
2015-05-19 10:58:46 +00:00
jnemeth
b710376266 Update to Asterisk 11.17.1: this contains a security fix, plus various bugs.
pkgsrc changes:
- adapt to upstream support for clang
- more comprehensive sweep for 64-bit time_t related stuff
- XXX pjsip has its own time related stuff that is 32-bit only

-----

The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.

The release of these versions resolves the following security vulnerability:

* AST-2015-003: TLS Certificate Common name NULL byte exploit

  When Asterisk registers to a SIP TLS device and and verifies the server,
  Asterisk will accept signed certificates that match a common name other than
  the one Asterisk is expecting if the signed certificate has a common name
  containing a null byte after the portion of the common name that Asterisk
  expected. This potentially allows for a man in the middle attack.

For more information about the details of this vulnerability, please read
security advisory AST-2015-003, which was released at the same time as this
announcement.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1

The security advisory is available at:

* http://downloads.asterisk.org/pub/security/AST-2015-003.pdf

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 11.17.0.

The release of Asterisk 11.17.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
      (Reported by Dwayne Hubbard)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
      res_odbc (Reported by ibercom)
 * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
      with replaces (Reported by Eelco Brolman)
 * ASTERISK-24479 - Enable REF_DEBUG for module references
      (Reported by Corey Farrell)
 * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
      fully disconnect underlying socket, leading to events being
      dropped with no additional information (Reported by Matt Jordan)
 * ASTERISK-24772 - ODBC error in realtime sippeers when device
      unregisters under MariaDB (Reported by Richard Miller)
 * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
      (Reported by Corey Farrell)
 * ASTERISK-24799 - [patch] make fails with undefined reference to
      SSLv3_client_method (Reported by Alexander Traud)
 * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
      for playing back messages stored in IMAP - play_message: No
      origtime (Reported by Graham Barnett)
 * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
      OSX with 64 bit integers (Reported by Corey Farrell)
 * ASTERISK-24796 - Codecs and bucket schema's prevent module
      unload (Reported by Corey Farrell)
 * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
      (Reported by Ashley Sanders)
 * ASTERISK-24797 - bridge_softmix: G.729 codec license held
      (Reported by Kevin Harwell)
 * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
      thread ID being passed to pthread_kill (Reported by JoshE)
 * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
      fail (Reported by Terry Wilson)
 * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
      SRTP for audio, but they responded without it' is ambiguous and
      wrong in some cases (Reported by Rusty Newton)
 * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
      error response and BYE are sent to the caller (Reported by
      Makoto Dei)
 * ASTERISK-18105 - most of asterisk modules are unbuildable in
      cygwin environment (Reported by feyfre)
 * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
 * ASTERISK-24838 - chan_sip: Locking inversion occurs when
      building a peer causes a peer poke during request handling
      (Reported by Richard Mudgett)
 * ASTERISK-24825 - Caller ID not recognized using
      Centrex/Distinctive dialing (Reported by Richard Mudgett)
 * ASTERISK-24739 - [patch] - Out of files -- call fails --
      numerous files with inodes from under /usr/share/zoneinfo,
      mostly posixrules (Reported by Ed Hynan)
 * ASTERISK-23390 - NewExten Event with application AGI shows up
      before and after AGI runs (Reported by Benjamin Keith Ford)
 * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
      voicemail stored in LDAP (Reported by Graham Barnett)
 * ASTERISK-24808 - res_config_odbc: Improper escaping of
      backslashes occurs with MySQL (Reported by Javier Acosta)
 * ASTERISK-20850 - [patch]Nested functions aren't portable.
      Adapting RAII_VAR to use clang/llvm blocks to get the
      same/similar functionality. (Reported by Diederik de Groot)
 * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
      by Frank DiGennaro)
 * ASTERISK-21038 - Bad command completion of "core set debug
      channel" (Reported by Richard Kenner)
 * ASTERISK-18708 - func_curl hangs channel under load (Reported by
      Dave Cabot)
 * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
      Atis Lezdins)
 * ASTERISK-24876 - Investigate reference leaks from
      tests/channels/local/local_optimize_away (Reported by Corey
      Farrell)
 * ASTERISK-24817 - init_logger_chain: unreachable code block
      (Reported by Corey Farrell)
 * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
      snuffy)
 * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
      under OpenBSD (Reported by snuffy)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
      Couldn't find mailbox %s in context (Reported by Graham Barnett)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0

Thank you for your continued support of Asterisk!

-----

The Asterisk Development Team has announced the release of Asterisk 11.16.0.

The release of Asterisk 11.16.0 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
      from JSSIP (Reported by Badalian Vyacheslav)
 * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
      enabled (Reported by Richard Mudgett)
 * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
      enabled (Reported by Andreas Steinmetz)
 * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
      casts char to unsigned int (Reported by Walter Doekes)
 * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
      level - 'Remote address is null, most likely RTP has been
      stopped' (Reported by Rusty Newton)
 * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
      on startup (Reported by Richard Kenner)
 * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
      destination when 'sendrpid=yes' (in proxy environment) (Reported
      by Karsten Wemheuer)
 * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
      (Reported by Kristian Høgh)
 * ASTERISK-20744 - [patch] Security event logging does not work
      over syslog (Reported by Michael Keuter)
 * ASTERISK-23850 - Park Application does not respect Return
      Context Priority (Reported by Andrew Nagy)
 * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
      in the CFlags returned (Reported by Diederik de Groot)
 * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
      voicemail is not deleted after review, hangup (Reported by LEI
      FU)
 * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
      32-bit packages on 64-bit hosts (Reported by Ben Klang)
 * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
      m() option does not queue an MWI event (Reported by Gareth
      Palmer)
 * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
      column comparison for 'defaultuser' (Reported by
      HZMI8gkCvPpom0tM)
 * ASTERISK-24719 - ConfBridge recording channels get stuck when
      recording started/stopped more than once (Reported by Richard
      Mudgett)
 * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
      by Kevin Harwell)
 * ASTERISK-24728 - tcptls: Bad file descriptor error when
      reloading chan_sip (Reported by Kevin Harwell)
 * ASTERISK-24676 - Security Vulnerability: URL request injection
      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
 * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
      versions (Reported by Jared Biel)
 * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
      Stephan Eisvogel)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0

Thank you for your continued support of Asterisk!
2015-05-19 07:52:14 +00:00
dsainty
c9e3bbc6f7 If the JVM is built-in, at least install under PREFIX still, rather than
attempting to install into the built-in JAVA_HOME.

Use 'arm' as JAVA_MACHINE_ARCH generally (correct for at least Linux, as
well as NetBSD).
2015-05-14 18:12:11 +00:00
manu
c23c5703db Fix crash in asterisk18 startup
The added patch fixes startup crash and was submitted upstream.
While there also remove the ban on i386, as it was tested to run fine.
2015-04-28 08:48:11 +00:00
ryoon
22bee7a3c3 Recursive revbump from databases/unixodbc. 2015-04-26 11:52:18 +00:00
tnn
255d0cb0b8 Recursive revbump following MesaLib update, categories a through f. 2015-04-25 14:20:17 +00:00
joerg
2307751955 Uses libtool, so replace with our version. Use the correct machine
directory on NetBSD/ARM.
2015-04-24 20:16:54 +00:00
jnemeth
25b33df245 Update to Asterisk 1.8.32.3: this is a security fix update.
The Asterisk Development Team has announced security releases for
Certified Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11,
12, and 13. The available security releases are released as versions
1.8.28.cert-5, 1.8.32.3, 11.6-cert11, 11.17.1, 12.8.2, 13.1-cert2,
and 13.3.2.

The release of these versions resolves the following security vulnerability:

* AST-2015-003: TLS Certificate Common name NULL byte exploit

  When Asterisk registers to a SIP TLS device and verifies the
  server, Asterisk will accept signed certificates that match a
  common name other than the one Asterisk is expecting if the signed
  certificate has a common name containing a null byte after the
  portion of the common name that Asterisk expected. This potentially
  allows for a man in the middle attack.

For more information about the details of this vulnerability, please read
security advisory AST-2015-003, which was released at the same time as this
announcement.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3

The security advisory is available at:

* http://downloads.asterisk.org/pub/security/AST-2015-003.pdf

Thank you for your continued support of Asterisk!
2015-04-12 03:35:38 +00:00
adam
9071d6b787 Revbump after updating textproc/icu 2015-04-06 08:17:13 +00:00
rodent
47d353dbf7 Include devel/libusb1/buildlink3.mk here. 2015-04-04 13:18:54 +00:00
rodent
49d6c01f8a Be lazy and use GITHUB_* variables in WRKSRC and INSTALLATION_DIRS. 2015-04-03 22:36:02 +00:00
rodent
609fd328c1 +libhidapi - packaged originally in pkgsrc-wip by:
nathanialsloss@yahoo.com.au
2015-04-03 22:31:32 +00:00
rodent
b55d91cf95 Import libhidapi-0.7.0 as comms/libhidapi.
HIDAPI is a multi-platform library which allows an application to interface
with USB and Bluetooth HID-Class devices on Windows, Linux, and Mac OS X.
On Windows, a DLL is built.  On other platforms (and optionally on Windows),
the single source file can simply be dropped into a target application.

HIDAPI has four back-ends:
	* Windows (using hid.dll)
	* Linux/hidraw (using the Kernel's hidraw driver)
	* Linux/libusb (using libusb-1.0)
	* Mac (using IOHidManager)

This package includes only the libusb backend.
2015-04-03 22:29:12 +00:00
khorben
7e8319bbee Update DeforaOS Phone to version 0.4.2
This version is essentially a bugfix release, with:
- minor improvements to the user interface;
- possibility to build outside of the source tree;
- dropped dependency on DeforaOS Panel;
- all tests should pass.

Hopefully will fix the issue encountered in the latest bulk build report.
2015-03-25 01:17:48 +00:00
jnemeth
21e904be90 NOT_FOR_PLATFORM->BROKEN_ON_PLATFORM as requested by dholland@ 2015-03-15 22:26:26 +00:00
tnn
4656e9d765 mark as NOT_FOR_UNPRIVILEGED 2015-03-15 15:08:30 +00:00
tnn
e71c4df136 honour PKGMANDIR 2015-03-12 15:23:39 +00:00
taca
17a1a339eb Add ${GEM_EXTSDIR}/gem.build_complete for new rubygems and updated ruby. 2015-03-08 15:17:17 +00:00
mef
3df8e3785f Update PLIST for doxygen 1.8.9.1, PKGREVISION++ 2015-02-02 02:45:09 +00:00
jnemeth
9545043a0d Update to Asterisk 11.15.1: this is a security fix.
pkgsrc change: adapt to splitting up of speex

The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
11.15.1, 12.8.1, and 13.1.1.

The release of these versions resolves the following security vulnerabilities:

* AST-2015-001: File descriptor leak when incompatible codecs are offered

                Asterisk may be configured to only allow specific audio or
                video codecs to be used when communicating with a
                particular endpoint. When an endpoint sends an SDP offer
                that only lists codecs not allowed by Asterisk, the offer
                is rejected. However, in this case, RTP ports that are
                allocated in the process are not reclaimed.

                This issue only affects the PJSIP channel driver in
                Asterisk. Users of the chan_sip channel driver are not
                affected.

* AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability

                CVE-2014-8150 reported an HTTP request injection
                vulnerability in libcURL. Asterisk uses libcURL in its
                func_curl.so module (the CURL() dialplan function), as well
                as its res_config_curl.so (cURL realtime backend) modules.

                Since Asterisk may be configured to allow for user-supplied
                URLs to be passed to libcURL, it is possible that an
                attacker could use Asterisk as an attack vector to inject
                unauthorized HTTP requests if the version of libcURL
                installed on the Asterisk server is affected by
                CVE-2014-8150.

For more information about the details of these vulnerabilities, please read
security advisory AST-2015-001 and AST-2015-002, which were released at the same
time as this announcement.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1

The security advisories are available at:

* http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2015-002.pdf

Thank you for your continued support of Asterisk!
2015-01-29 21:54:33 +00:00
jnemeth
8f3e6d6a96 Update to Asterisk 10.12.4rc6.
This update is just to accomodate the speex splitup.

Note that Asterisk 10.x is dead upstream and should not be used
anymore.  This package will be removed at some point.
2015-01-29 21:52:28 +00:00
jnemeth
5fb63ec5f0 Update to asterisk 1.8.32.2: this is a security fix.
pkgsrc change: adapt to splitting up of speex

The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
11.15.1, 12.8.1, and 13.1.1.

The release of these versions resolves the following security vulnerabilities:

* AST-2015-001: File descriptor leak when incompatible codecs are offered

                Asterisk may be configured to only allow specific audio or
                video codecs to be used when communicating with a
                particular endpoint. When an endpoint sends an SDP offer
                that only lists codecs not allowed by Asterisk, the offer
                is rejected. However, in this case, RTP ports that are
                allocated in the process are not reclaimed.

                This issue only affects the PJSIP channel driver in
                Asterisk. Users of the chan_sip channel driver are not
                affected.

* AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability

                CVE-2014-8150 reported an HTTP request injection
                vulnerability in libcURL. Asterisk uses libcURL in its
                func_curl.so module (the CURL() dialplan function), as well
                as its res_config_curl.so (cURL realtime backend) modules.

                Since Asterisk may be configured to allow for user-supplied
                URLs to be passed to libcURL, it is possible that an
                attacker could use Asterisk as an attack vector to inject
                unauthorized HTTP requests if the version of libcURL
                installed on the Asterisk server is affected by
                CVE-2014-8150.

For more information about the details of these vulnerabilities, please read
security advisory AST-2015-001 and AST-2015-002, which were released at the same
time as this announcement.

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

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1

The security advisories are available at:

* http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
* http://downloads.asterisk.org/pub/security/AST-2015-002.pdf

Thank you for your continued support of Asterisk!
2015-01-29 21:48:07 +00:00
bsiegert
20a5c0d763 Switch license to modified-bsd. Move socks4 option over to use dante.
Patch provided by Kirk Russell in PR pkg/49546.
2015-01-17 15:30:03 +00:00
taca
830c875fdf Fix typo, s/GEM_CLEANBUOLD_EXTENSIONS/GEM_CLEANBUILD_EXTENSIONS/. 2015-01-16 09:18:47 +00:00
joerg
e8faa47526 Don't depend on parser skeleton to include stdlib.h. 2015-01-09 14:28:42 +00:00