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!

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