[asterisk-dev] Asterisk 14.0.0 Now Available!

The Asterisk Development Team is pleased to announce the release of
Asterisk 14.0.0. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

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!

[asterisk-dev] Asterisk 13.8.0 Now Available

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

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

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 – 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 – 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 – 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 – res_phoneprov: memory leak and

     heap-use-after-free (Reported by Badalian Vyacheslav)

* ASTERISK-25272 – 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 – 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 – 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 – 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@anonymous.invalid> (Reported by Anthony

     Messina)

* ASTERISK-24813 – asterisk.c: #if statement in listener()

     confuses code folding editors (Reported by Corey Farrell)

* ASTERISK-25767 – 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!

[asterisk-dev] Asterisk 11.22.0 Now Available

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

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

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 – DeadLock ChanSpy with call over Local

     channel (Reported by Filip Frank)

* ASTERISK-25800 – Calculate talktime when is first call

     answered (Reported by Rodrigo Ramirez Norambuena)

* ASTERISK-25272 – 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 – 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 – 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 – 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!

[asterisk-dev] libpri 1.5.0 Now Available

The Asterisk Development Team has announced the release of libpri 1.5.0.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri

The release of libpri 1.5.0 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:

Bugs fixed in this release:

———————————–

* PRI-180 – Incorrect handling of DISCONNECT with Progress

     Indicator #8 (Reported by Alexandr Dranchuk)

* PRI-173 – libpri does not handle keypad facility IE in overlap

     mode (Reported by Gerald Schnabel)

* PRI-182 – Tighten mandatory ie checks and other misc items

     (Reported by Kevin Harwell)

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

http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.5.0

Thank you for your continued support of Asterisk!

[asterisk-dev] DAHDI-Linux and DAHDI-Tools 2.11.1 Now Available

The Asterisk Development Team has announced the releases of:

DAHDI-Linux-v2.11.1

DAHDI-Tools-v2.11.1

dahdi-linux-complete-2.11.1+2.11.1

This release is available for immediate download at:

http://downloads.asterisk.org/pub/telephony/dahdi-linux

http://downloads.asterisk.org/pub/telephony/dahdi-tools

http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

Notable changes:

Raised E1 pulse level for WCTE23x and WCTE43x.

Fixed dahdi-tools compilation errors on newer Linux systems.

For a full list of changes in these releases, please see the shortlog at:

http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.11.1

http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.11.1

Issues found in this release can be reported in the DAHDI-Linux [1] and

DAHDI-Tools [2] projects at https://issues.asterisk.org/jira

[1] https://issues.asterisk.org/jira/browse/DAHLIN

[2] https://issues.asterisk.org/jira/browse/DAHTOOL

Thank you for your continued support of Asterisk!

[asterisk-dev] Asterisk 11.21.2 Now Available

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

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

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!

[asterisk-dev] Asterisk 13.7.2 Now Available

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

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

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!

Asterisk 13.7.0 Now Available

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

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

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 – 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 – Asterisk may crash when calling

     ast_channel_get_t38_state(c) (Reported by Filip Jenicek)

* ASTERISK-24146 – No audio on WebRtc caller side when

     answer waiting time is more than ~7sec (Reported by Aleksei

     Kulakov)

* ASTERISK-25599 – 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 – format-attribute module: VP8 missing

     (Reported by Alexander Traud)

* ASTERISK-25583 – 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 – 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 – translation module gets cached not

     joint format (Reported by Alexander Traud)

* ASTERISK-25573 – 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 – Opus Codec: SIP/SDP line fmtp missing

     when called internally (Reported by Alexander Traud)

* ASTERISK-25535 – 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 – format-attribute module: RFC or

     internal defaults? (Reported by Alexander Traud)

* ASTERISK-25533 – 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 – Update the maximum packetization values

     in frame.c (Reported by dea)

* ASTERISK-25484 – 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 – 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 – 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 – 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!

Asterisk 11.21.0 Now Available

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

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

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 – 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 – Asterisk may crash when calling

     ast_channel_get_t38_state(c) (Reported by Filip Jenicek)

* ASTERISK-24146 – No audio on WebRtc caller side when

     answer waiting time is more than ~7sec (Reported by Aleksei

     Kulakov)

* ASTERISK-25599 – 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 – 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 – 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 – 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 – 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 – 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 – 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!

[asterisk-dev] DAHDI-Linux and DAHDI-Tools 2.11.0 Now Available

The Asterisk Development Team has announced the releases of:

DAHDI-Linux-v2.11.0

DAHDI-Tools-v2.11.0

dahdi-linux-complete-2.11.0+2.11.0

This release is available for immediate download at:

http://downloads.asterisk.org/pub/telephony/dahdi-linux

http://downloads.asterisk.org/pub/telephony/dahdi-tools

http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

* This release adds automake support to dahdi-tools. This should be a transparent change to users of this tarball, but changes in our release tools may

potentially cause issues. Please test this RC fully to ensure it remains compatible with your integration cycle.

* Adds and improves support for Digium’s b433, b434, b233, and b234

* Fixes compile issues with 4.0+ kernels

Shortlog of dahdi-linux changes since v2.10.1:

John Sloan (1):

     wcxb: Fix “I/O error reported by firmware” followed by underruns

Oron Peled (1):

     xpp: improve handling of USB sluggishness

Russ Meyerriecks (9):

     wcb4xxp: Add support for zarlink echocan

     wctc4xxp: Fix continuous “errored receive packets” with 2+ transcoders

     readme: Updated supported products section

     octasic: Added slab.h include to fix ARM compile error

     wcb4xxp: Remove “card sync source” logic

     wcb4xxp: minor: Squelch initializing message on shutdown

     wcb4xxp: Protect indirect register writes with sequence lock

     wcb4xxp: Add support for wcb23x series

     wcb4xxp: Print serial number of gen 2 cards.

Shaun Ruffell (6):

     dahdi: Fix “void value not ignored…” error when compiling against kernel 4.0.

     dahdi: strnicmp() -> strncasecmp()

     build_tools/make_version: Remove support for subversion working copies.

     Remove DAHDI_IRQF_[SHARED|DISABLED] flags.

     dahdi: Remove IRQF_DISABLED.

     wctc4xxp: Clear packet error count when reloading firmware.

Tzafrir Cohen (4):

     xpp: module_refcount is back to int on 3.19

     README: use file time for reproducable build

     xpp: USB_FW*: fix incorrect chan num with 2FXS6FXO

     dahdi-modules: load and unloads the modules

Shortlog of dahdi-tools changes since v2.10.1:

Oron Peled (38):

     build: remove autoconf generated files

     build: remove unused build_tools/menuselect-deps.in

     build: remove unused xpp/oct612x/Makefile

     build: fix ppp/Makefile CFLAGS

     build: xpp — remove legacy usb-hotplug

     build: generate version.c during configure

     build: remove unused “update” target from Makefile

     autoconf: rename “dahdi” to “dahdi-tools”

     automake: add basic framwork

     automake: add basic libtool support

     automake: full ppp/ support

     automake: handle xpp/ compilation via Makefile.am:

     automake: xpp: man-pages and perl-scripts

     automake: xpp: handle udev rules

     automake: xpp: handle /usr/share/dahdi

     automake: xpp: remove xpp/Makefile.legacy

     automake: handle doc/ man-pages

     automake: migrate tools from Makefile.legacy

     automake: migrated “–enable-dev-mode”

     automake: migrate dahdi_pcap from Makefile.legacy

     automake: handle “make dist”

     automake: bugfix: fix installation paths

     automake: remove unused stuff from Makefile.legacy

     autotools: now “make distcheck” also works.

     xpp: bugfix: waitfor_xpds twinstar, auto_assign_spans

     xpp: automake: cleanup $man_MANS handling

     configure.ac: add libusb/libusbx support (for xpp)

     xpp: add all base libxtalk files

     xpp: migrate everything to libxtalk

     xpp: xtalk — no private status range

     xpp: don’t use USB “clear-halt” by default

     xpp: strict compilation flags in oct612x/

     configure.ac: xpp: now libusbx is the default

     xpp: allow XTALK_OPTIONS from a file

     xpp: Move astribank_license to libastribank

     xpp: refactor manual pages out of conditionals

     configure.ac: remove unused PKG_CONFIG_LIBUSB

     xpp: move tools man-pages into PBX_USB conditional

Russ Meyerriecks (2):

     xpp: Fix a logical not being applied to the wrong operand

     wcb4xxp: Adds support for b43x/b23x products

Tzafrir Cohen (10):

     xpp_fxloader: handle empty span-type.conf

     Update README

     A placeholder for the m4 directory

     Force-link libtonezone.so.2.0 on make install

     README: document the need for autoreconf

     Makefile: build asciidoc with TZ=UTC

     make_version: cut off slashes in git

     ignore: dahdi_pcap and .version

     xpp.rules: increase xpp_fxloader timeout to 180s

     dahdi_cfg: -S has assumtions on system.conf order

For a full list of changes in these releases, please see the shortlog at:

http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.11.0

http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.11.0

Issues found in this release can be reported in the DAHDI-Linux [1] and DAHDI-Tools [2] projects at https://issues.asterisk.org/jira

[1] https://issues.asterisk.org/jira/browse/DAHLIN

[2] https://issues.asterisk.org/jira/browse/DAHTOOL