[asterisk-dev] Asterisk 13.5.0-rc1 Now Available
Asterisk Development Team
asteriskteam at digium.com
Mon Jul 27 13:03:50 CDT 2015
The Asterisk Development Team has announced the first release candidate of
Asterisk 13.5.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.5.0-rc1 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 candidate:
Improvements made in this release:
-----------------------------------
* ASTERISK-25256 - [patch]Post AMI VarSet to empty string events
when Asterisk deletes a dialplan variable. (Reported by Richard
Mudgett)
* ASTERISK-25067 - Sorcery Caching: Implement a new caching module
(Reported by Matt Jordan)
* ASTERISK-25040 - pbx: Improve performance of reloads by making
hint destruction more performant (Reported by Matt Jordan)
* ASTERISK-25114 - res_pjsip: Add AMI events for chan_pjsip
contact lifecycle changes (Reported by George Joseph)
* ASTERISK-25072 - res_pjsip_outbound_registration: line
functionality. Additional check for using the request URI
(Reported by Dmitriy Serov)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25250 - chan_sip - Despite the channel being answered,
caller on a call established via Local channel continues to hear
ringback (Reported by Etienne Lessard)
* ASTERISK-25253 - confbridge volume options and other volume
controls such as func_volume don't work (Reported by Dmitriy
Serov)
* ASTERISK-25247 - choppy audio when spying on a g722 channel,
chan_sip or chan_pjsip (Reported by hristo)
* ASTERISK-24867 - Docs for 'e' option in ResetCDR say to use
CDR_PROP instead, CDR_PROP docs are unclear (Reported by Rusty
Newton)
* ASTERISK-24853 - Documentation claims chan_sip outbound
registrations support WS or WSS as valid transports (not true)
(Reported by PSDK)
* ASTERISK-25242 - PJSIP: No audio when Asterisk inside NAT and
endpoints outside NAT - implement functionality similar to
chan_sip 'rtpkeepalive'? (Reported by Mark Michelson)
* ASTERISK-25258 - chan_pjsip: Incorrect format switch on received
RTP packet (Reported by Joshua Colp)
* ASTERISK-25257 - [patch]channels/sig_pri.h -> sig_pri_span ->
force_restart_unavailable_chans in wrong scope (Reported by
Patric Marschall)
* ASTERISK-24934 - [patch]Asterisk manager output does not escape
control characters (Reported by warren smith)
* ASTERISK-25255 - Missing AMI VarSet events when setting to an
empty string. (Reported by Richard Mudgett)
* ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to an
empty string before Park. (Reported by Richard Mudgett)
* ASTERISK-25183 - PJSIP: Crash on NULL channel in
chan_pjsip_incoming_response despite previous checks for NULL
channel (Reported by Matt Jordan)
* ASTERISK-25201 - Crash in PJSIP distributor on already free'd
threadpool (Reported by Matt Jordan)
* ASTERISK-24782 - StasisEnd event not present for channel that
was swapped out for another after completing attended transfer
(Reported by John Bigelow)
* ASTERISK-25240 - bridge_native_rtp: Direct media wrongfully
started when completing attended transfer (Reported by Joshua
Colp)
* ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
(Reported by Rusty Newton)
* ASTERISK-22805 - res_rtp_asterisk: Crash when calling
BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
(Reported by Dmitry Burilov)
* ASTERISK-24550 - res_rtp_asterisk: Crash in
ast_rtp_on_ice_complete during DTLS handshake (Reported by
Osaulenko Alexander)
* ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
Badalian Vyacheslav)
* ASTERISK-24832 - [patch]DTLS-crashes within openssl (Reported
by Stefan Engström)
* ASTERISK-25127 - DTLS crashes following "Unable to cancel
schedule ID" in dtls_srtp_check_pending (Reported by Dade
Brandon)
* ASTERISK-25168 - Random Core Dumps on Asterisk 13.4 PJSIP, in
ast_channel_name at channel_internal_api.c (Reported by Carl
Fortin)
* ASTERISK-25115 - Crash related to func
sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c
(Reported by John Bigelow)
* ASTERISK-25226 - chan_sip: Channel leak in branch 13 on early
replaces call pickup (Reported by Walter Doekes)
* ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
(Reported by Walter Doekes)
* ASTERISK-25219 - [patch]Source and destination overlap in memcpy
in rtp_engine.c (Reported by Walter Doekes)
* ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
(Reported by Walter Doekes)
* ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
Bad file descriptor" (Reported by Barry Chern)
* ASTERISK-25165 - Testsuite - Sorcery memory cache leaks
(Reported by Corey Farrell)
* ASTERISK-25202 - Hints extension state broken between 13.3.2 and
13.4 (Reported by cervajs)
* ASTERISK-25196 - res_pjsip_nat: rewrite_contact should not be
applied to Contact header when Record-Route headers are present
(Reported by Mark Michelson)
* ASTERISK-24907 - res_pjsip_outbound_registration: crash during
unload if registration attempts are still occuring (Reported by
Kevin Harwell)
* ASTERISK-25204 - res_pjsip_refer: Duplicated Referred-By or
Replaces headers on outbound INVITEs. (Reported by Mark
Michelson)
* ASTERISK-25171 - Early completion of feature code attended
transfer results in intermittent one-way audio, "ghost ringing"
and robotic sound. (Reported by Rusty Newton)
* ASTERISK-25189 - AMI: Add Linkedid header to standard channel
snapshot information. (Reported by Richard Mudgett)
* ASTERISK-25172 - Crash in channels/sip/sip blind
transfer/caller_refer_only test in
ast_format_cap_append_from_cap during ast_request (Reported by
Matt Jordan)
* ASTERISK-25180 - res_pjsip_mwi: Unsolicited MWI requires reload
(Reported by Joshua Colp)
* ASTERISK-25182 - [patch] on CLI sip reload, new codecs get
appended only (Reported by Alexander Traud)
* ASTERISK-25163 - Deadlock in chan_sip between reload of sip peer
container and MWI Stasis callback (Reported by Dmitriy Serov)
* ASTERISK-25091 - Asterisk REST API - bridge.addChannel crash
asterisk when calling channel hangup while adding to bridge
(Reported by Ilya Trikoz)
* ASTERISK-24900 - Manager event ParkedCallSwap is not documented
(Reported by Rusty Newton)
* ASTERISK-25162 - func_pjsip_aor: Leak of contact in iterator
(Reported by Corey Farrell)
* ASTERISK-25158 - res_pjsip: Add option to use AAL2 packing when
negotiating g.726 (Reported by Kevin Harwell)
* ASTERISK-24344 - CDR_PROP(disable) disables CDR only for first
dialed party (Reported by Janusz Karolak)
* ASTERISK-24443 - CDR fields (dst, dcontext) empty in transfer
call started from Macro (Reported by Arveno Santoro)
* ASTERISK-25154 - [patch]fromtag may need to be updated after
successful call dialog match (Reported by Damian Ivereigh)
* ASTERISK-25156 - chan_pjsip’s CHAN_START cel event lacks the
correct context and exten (Reported by cloos)
* ASTERISK-25157 - bridging: Performing a blonde transfer does not
result in connected line updates (Reported by Joshua Colp)
* ASTERISK-25087 - Asterisk segfault when using Directory
application with alias option and specific mailbox configuration
(Reported by Chet Stevens)
* ASTERISK-24983 - IAX deadlock between hangup and scheduled
actions (ex. largrq) (Reported by Y Ateya)
* ASTERISK-25096 - [patch]Segfault when registering over
websockets with PJSIP (in ast_sockaddr_isnull at
/include/asterisk/netsock2.h) (Reported by Josh Kitchens)
* ASTERISK-24963 - ASAN: heap-use-after-free with PJSIP and WSS
(Reported by Badalian Vyacheslav)
* ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
but asterisk doesn't detect it. (Reported by ibercom)
* ASTERISK-25094 - PBX core: Investigate thread safety issues
(Reported by Corey Farrell)
* ASTERISK-25148 - res_pjsip NULL channel audit (Reported by Mark
Michelson)
* ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
| adpcm | ipc10} (Reported by Badalian Vyacheslav)
* 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-25131 - chan_pjsip: In-dialog authentication not
handled. (Reported by Richard Mudgett)
* ASTERISK-25100 - asterisk coredump if host has an IPv6 address
that end with ::80 (Reported by Mark Petersen)
* ASTERISK-25122 - Large SIP packet received via pjsip over
websocket crashes Asterisk (Reported by Ivan Poddubny)
* ASTERISK-25121 - Stasis: Fix unsafe use of stasis_unsubscribe in
modules. (Reported by Corey Farrell)
* ASTERISK-24988 - func_talkdetect: Test is bouncing sporadically
(Reported by Joshua Colp)
* ASTERISK-25105 - res_pjsip: Possible incompatibility between
qualify_timeout and pjproject-2.4 (Reported by George Joseph)
* ASTERISK-25117 - res_mwi_external_ami: Fix manager action
registrations. (Reported by Corey Farrell)
New Features made in this release:
-----------------------------------
* ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by
Joshua Colp)
* ASTERISK-25238 - ARI: Support push configuration (Reported by
Matt Jordan)
* ASTERISK-25173 - ARI: Add the ability to load/reload/unload an
Asterisk module (Reported by Matt Jordan)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.5.0-rc1
Thank you for your continued support of Asterisk!
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