./comms/asterisk13, The Asterisk Software PBX

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Branch: CURRENT, Version: 13.9.1, Package name: asterisk-13.9.1, 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]


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

Master sites: (Expand)


Version history: (Expand)


CVS history: (Expand)


   2016-06-09 06:41:49 by John Nemeth | Files touched by this commit (4) | Package updated
Log message:
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/telep … Log-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/telep … Log-13.9.0

Thank you for your continued support of Asterisk!
   2016-06-08 12:16:57 by Jonathan Perkin | Files touched by this commit (89)
Log message:
Remove the stability entity, it has no meaning outside of an official context.
   2016-06-08 11:46:05 by Jonathan Perkin | Files touched by this commit (47)
Log message:
Change the service_bundle name to "export" to reduce diffs between the
original manifest.xml file and the output from "svccfg export".
   2016-05-06 09:41:06 by John Nemeth | Files touched by this commit (7) | Package updated
Log message:
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/telep … Log-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/telep … Log-13.8.0

Thank you for your continued support of Asterisk!
   2016-04-11 21:02:08 by Ryo ONODERA | Files touched by this commit (527)
Log message:
Recursive revbump from textproc/icu 57.1
   2016-03-05 12:29:49 by Jonathan Perkin | Files touched by this commit (1813) | Package updated
Log message:
Bump PKGREVISION for security/openssl ABI bump.
   2016-02-25 12:32:19 by Jonathan Perkin | Files touched by this commit (4)
Log message:
Use OPSYSVARS.
   2016-02-07 10:13:34 by John Nemeth | Files touched by this commit (5) | Package updated
Log message:
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/telep … Log-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/telep … Log-13.7.1

The security advisories are available at:

 * http://downloads.asterisk.org/pub/secur … 16-001.pdf
 * http://downloads.asterisk.org/pub/secur … 16-002.pdf
 * http://downloads.asterisk.org/pub/secur … 16-003.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/telep … Log-13.7.0

Thank you for your continued support of Asterisk!