The Asterisk Development Team would like to announce the first
release candidate of Asterisk 13.26.0.
This release candidate is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.26.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release candidate:
New Features made in this release:
-----------------------------------
* ASTERISK-28267 - res_stasis: Add ability to switch
applications
(Reported by Benjamin Keith Ford)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate
(Reported by Olivier Krief)
* ASTERISK-28350 - manager: Stasis backed up due to locking
(Reported by Joshua C. Colp)
* ASTERISK-25792 - chan_sip: qualifygap bounds checking
(Reported by Paul Sandys)
* ASTERISK-28341 - res_config_odbc eliminates empty custom (â@â
prefix) variables
(Reported by Alexei Gradinari)
* ASTERISK-28333 - StasisEnd event makes wrong timestamp value
(Reported by sungtae kim)
* ASTERISK-28306 - res_pjsip_mwi: MWI NOTIFY occasionally takes
minutes to be sent
(Reported by Jared Hull)
* ASTERISK-27964 - app_queue: ring_entry accesses nativeformats
without channel lock or reference
(Reported by Francisco
Seratti)
* ASTERISK-28314 - ARI: API changed but "apiVersion" in
rest-api\resources.json did not
(Reported by Stefan Repke)
* ASTERISK-28335 - stasis: Make topic and maybe subscription
names unique and more useful
(Reported by Joshua C. Colp)
* ASTERISK-28321 - res_rtp_asterisk: Fixing possible divide by
zero for rtcp stat calculation
(Reported by sungtae kim)
* ASTERISK-28332 - Variable ALTCONF ignored when service is
used in Debian
(Reported by Cirillo Ferreira)
* ASTERISK-28322 - chan_pjsip: Add option to allow ignoring of
183 without SDP
(Reported by Torrey Searle)
* ASTERISK-28328 - MeetMe global non-admin mute is muting
admins that subsequently join
(Reported by Philip Mott)
* ASTERISK-28168 - app_queue: Adding a blank entry into sql
queue_members crashes asterisk.
(Reported by Michael)
* ASTERISK-28323 - pjsip: sip.conf to pjsip.conf conversion
script fails
(Reported by Guido Weckwerth)
* ASTERISK-28272 - The basic-pbx config samples don't produce a
running asterisk
(Reported by George Joseph)
* ASTERISK-28312 - res_pjsip_diversion: Corrupted SIP Diversion
field after handling a 302 redirect
(Reported by Alex
Odrov)
* ASTERISK-24173 - File menuselect/menuselect_gtk.c has no
license header
(Reported by Jeremy Lainé)
* ASTERISK-28166 - app_voicemail: Asterisk unresponsive after
changing voicemail password with ODBC
(Reported by
Michael)
* ASTERISK-28309 - res_pjsip: Wrong Contact and Via fields with
multiple UDP interfaces
(Reported by Nikolay shakin)
* ASTERISK-27992 - PJSIP: Adding `sends_registrations = yes` to
pjsip_wizard.conf causes crash
(Reported by Jonathan
Harris)
* ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
AOR is blocked
(Reported by Ross Beer)
* ASTERISK-28301 - Allow voicemail boxes to be subscribed to
with a presence event package
(Reported by George Joseph)
* ASTERISK-28303 - res_rtp_asterisk: Interaction between
smoother and DTMF can cause out of order timestamps
(Reported by Torrey Searle)
* ASTERISK-28302 - ARI: "Error destroying mutex" when listing
all ARI applications
(Reported by Stefan Repke)
* ASTERISK-28300 - AST_PBX_MAX_STACK is too low for some
applications
(Reported by George Joseph)
* ASTERISK-28106 - Astricon Feedback: Unable to filter ARI
events when GETting causes overload of events
(Reported by
George Joseph)
* ASTERISK-28284 - switching between native_bridge and
simple_bridge can cause one way audio
(Reported by Torrey
Searle)
* ASTERISK-28288 - Resources (udptl fd) leaking for T.38 calls
(Reported by Paulo Vicentini)
* ASTERISK-28251 - CI: Fix CI so it reverifies commit message
changes
(Reported by George Joseph)
* ASTERISK-28277 - database: Add some basic logging
(Reported by Joshua C. Colp)
* ASTERISK-28181 - ari: Originating overwrites channel start
time
(Reported by sungtae kim)
Improvements made in this release:
-----------------------------------
* ASTERISK-28326 - ari: Added timestamp for some ari events.
(Reported by sungtae kim)
* ASTERISK-28317 - Add logical group at DAHDIChannel event and
create "dahdi_group" at CHANNEL function
(Reported by
Cirillo Ferreira)
* ASTERISK-28279 - Added creation timestamp for bridge
(Reported by sungtae kim)
* ASTERISK-28292 - Changed to show all channel stats including
wrong media
(Reported by sungtae kim)
For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.26.0-rc1
Thank you for your continued support of Asterisk!
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