The Asterisk Development Team would like to announce the first
release candidate of Asterisk 13.24.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.24.0-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:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-28013 - res_http_websocket: Crash when reading HTTP
Upgrade requests
(Reported by Sean Bright)
New Features made in this release:
-----------------------------------
* ASTERISK-28087 - add flag to allow CALLERID(num) to be placed
in Contact header in chan_pjsip
(Reported by Torrey
Searle)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-28125 - app_queue: Revert broken queue channel
reference patch
(Reported by lvl)
* ASTERISK-28151 - app_voicemail: MWI fails with
mailboxes=##@device instead of mailboxes=##@default
(Reported by Ronald Raikes)
* ASTERISK-28157 - Asterisk crashes when the res_pjsip_*
modules unload
(Reported by sungtae kim)
* ASTERISK-28159 - SIGABRT caused by stack corruption in
hashkeys_read when no matching keys present
(Reported by
Michael Walton)
* ASTERISK-28140 - repeated segmentation faults
(Reported by Eyal Hasson)
* ASTERISK-28103 - stasis: Filter messages at publishing to
reduce work done
(Reported by Joshua C. Colp)
* ASTERISK-28129 - Incorrect Behavior for rewrite_contact when
Re-Invite omits routset
(Reported by Torrey Searle)
* ASTERISK-28158 - Some conditions prevent running of el_end,
break the terminal.
(Reported by Corey Farrell)
* ASTERISK-28162 - [patch] need to reset DTMF last sequence
number and timestamp on voice packet with marker bit
(Reported by Alexei Gradinari)
* ASTERISK-28110 - rtp: Incorrect Packetization
(Reported
by Robert Cripps)
* ASTERISK-28146 - pbx_config: Only the first [globals] section
is processed.
(Reported by Corey Farrell)
* ASTERISK-28150 - Formatting error in documentation
(Reported by Scott Griepentrog)
* ASTERISK-28081 - chan_sip: Asterisk 12+ chan_sip doesn't
report AST_CEL_PICKUP in handle_invite_replaces
(Reported
by Luit van Drongelen)
* ASTERISK-28137 - res_pjsip_notify: improve realtime
performance on CLI completion on the endpoint
(Reported by
Alexei Gradinari)
* ASTERISK-27980 - Caller ID cannot be changed on Attended
Transfer before dialing out
(Reported by Alexei Gradinari)
* ASTERISK-28089 - function ast_sendtext() create RTP realtime
packets with a trailing null byte in the payload
(Reported
by Emmanuel BUU)
* ASTERISK-28076 - bridging: Asterisk crashes when receiving an
empty realtime text frame
(Reported by Emmanuel BUU)
* ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding
AMI
(Reported by Andrej)
* ASTERISK-28077 - res_pjsip: improve realtime performance on
CLI 'pjsip show contacts'
(Reported by Alexei Gradinari)
* ASTERISK-26094 - stasis: Playing MOH to bridge with ARI does
not work
(Reported by Cameron)
* ASTERISK-27920 - app_queue: Queue member considered inuse
after immediately hanging up during dialing.
(Reported by
Cao Minh Hiep)
* ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use
ports below 10000
(Reported by Joshua C. Colp)
* ASTERISK-28065 - res_odbc: missing SQL error diagnostic
(Reported by Alexei Gradinari)
* ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload
(Reported by Sergej Kasumovic)
* ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version
2.8
(Reported by Joshua C. Colp)
* ASTERISK-28057 - chan_sip: SipNotify via AMI behaves
differently to CLI
(Reported by Peter Katzmann)
* ASTERISK-28049 - res_pjproject build failure
(Reported
by Jaco Kroon)
* ASTERISK-28029 - [patch] res_musiconhold : music on hold will
not start if previous hold just reached end of file
(Reported by Frederic LE FOLL)
* ASTERISK-28032 - Realtime queuemembers are not updated during
retry phase
(Reported by lvl)
* ASTERISK-27988 - alembic: PJSIP
"mwi_subscribe_replaces_unsolicited" field is integer not
boolean
(Reported by Joshua C. Colp)
* ASTERISK-28020 - res_pjsip_transport_websocket: Properly set
'received' for IPv6
(Reported by Sean Bright)
Improvements made in this release:
-----------------------------------
* ASTERISK-28144 - [patch] New function PJSIP_PARSE_URI to
parse an URI and return a specified part of the URI
(Reported by Alexei Gradinari)
* ASTERISK-28136 - Allow the sip_to_pjsip script to be used in
a pipe
(Reported by Pascal Cadotte Michaud)
* ASTERISK-28046 - Remove stale nonoptreq references
(Reported by Walter Doekes)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.24.0-rc1
Thank you for your continued support of Asterisk!
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