./comms/asterisk13, The Asterisk Software PBX

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Branch: CURRENT, Version: 13.13.0nb1, Package name: asterisk-13.13.0nb1, Maintainer: jnemeth

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

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

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

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


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

Required to build:
[pkgtools/cwrappers]

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

Master sites: (Expand)


Version history: (Expand)


CVS history: (Expand)


   2016-12-04 06:17:46 by Ryo ONODERA | Files touched by this commit (667)
Log message:
Recursive revbump from textproc/icu 58.1
   2016-11-27 09:48:19 by John Nemeth | Files touched by this commit (3) | Package updated
Log message:
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/telep … og-13.13.0

Thank you for your continued support of Asterisk!
   2016-11-11 16:44:16 by John Nemeth | Files touched by this commit (2) | Package updated
Log message:
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/telep … og-13.12.2

Thank you for your continued support of Asterisk!
   2016-10-29 04:10:06 by John Nemeth | Files touched by this commit (2) | Package updated
Log message:
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/telep … og-13.12.1

Thank you for your continued support of Asterisk!
   2016-10-27 03:08:17 by John Nemeth | Files touched by this commit (4) | Package updated
Log message:
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/telep … og-13.12.0

Thank you for your continued support of Asterisk!
   2016-10-09 23:42:04 by Thomas Klausner | Files touched by this commit (110)
Log message:
Recursive bump for all users of pgsql now that the default is 95.
   2016-10-07 20:26:14 by Adam Ciarcinski | Files touched by this commit (611) | Package updated
Log message:
Revbump post boost update
   2016-09-23 19:50:19 by John Nemeth | Files touched by this commit (6) | Package updated
Log message:
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/telep … og-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/telep … og-13.11.0

Thank you for your continued support of Asterisk!