asterisk-lupaEl equipo de desarrolladores de Asterisk acaba de publicar la nueva versión Asterisk13.2.0-rc1 

Puedes ver la lista de cambios de esta versión en el siguiente enlace:
Descargar Changelog

También puedes esta versión en el siguiente enlace:
Descargar Asterisk

El anuncio oficial:

The release of Asterisk 13.2.0-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:

Bugs fixed in this release:
———————————–
* ASTERISK-24342 – PJSIP: Qualifying endpoints attempts to do them
all at the same time. (Reported by Richard Mudgett)
* ASTERISK-24514 – res_pjsip_outbound_registration: stack overflow
when using non-default sorcery wizard (Reported by Kevin
Harwell)
* ASTERISK-24472 – Asterisk Crash in OpenSSL when calling over WSS
from JSSIP (Reported by Badalian Vyacheslav)
* ASTERISK-24607 – res_pjsip_session: re-INVITE with declined
media streams results in 488 (Reported by Matt Jordan)
* ASTERISK-24563 – Direct Media calls within private network
sometimes get one way audio (Reported by Kevin Harwell)
* ASTERISK-24604 – res_rtp_asterisk: Crash during restart due to
race condition in accessing codec in stored ast_frame and codec
core (Reported by Matt Jordan)
* ASTERISK-24614 – Deadlock when DEBUG_THREADS compiler flag
enabled (Reported by Richard Mudgett)
* ASTERISK-24449 – Reinvite for T.38 UDPTL fails if SRTP is
enabled (Reported by Andreas Steinmetz)
* ASTERISK-24619 – [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
casts char to unsigned int (Reported by Walter Doekes)
* ASTERISK-24536 – AMI redirect with PJSIP fails to move extra
channel (Reported by Niklas Larsson)
* ASTERISK-24459 – bridge_native_rtp: Native RTP bridging is
chosen for RTP compatible channels when the DTMF mode is not
compatible (Reported by Yaniv Simhi)
* ASTERISK-24337 – Spammy DEBUG message needs to be at a higher
level – ‘Remote address is null, most likely RTP has been
stopped’ (Reported by Rusty Newton)
* ASTERISK-24513 – Local channel apparently leaked in off-nominal
DTMF attended transfer (Reported by Mark Michelson)
* ASTERISK-23733 – ‘reload acl’ fails if acl.conf is not present
on startup (Reported by Richard Kenner)
* ASTERISK-24628 – [patch] chan_sip – CANCEL is sent to wrong
destination when ‘sendrpid=yes’ (in proxy environment) (Reported
by Karsten Wemheuer)
* ASTERISK-23841 – DTMF atxfer doesn’t set CallerID for the recall
calls to the transferrer. (Reported by Richard Mudgett)
* ASTERISK-24376 – res_pjsip_refer: REFER request for remote
session attempts to direct channel to external_replaces
extension instead of context, without providing for the
Referred-To SIP URI (Reported by Matt Jordan)
* ASTERISK-24591 – Stasis() side of an ARI originated channel
cannot be Redirected (Reported by Kinsey Moore)
* ASTERISK-24049 – Asterisk Manager Interface: A number of list
type responses aren’t using astman_send_listack (Reported by
Jonathan Rose)
* ASTERISK-24637 – Channel re-enters Stasis() when it should not
(Reported by John Bigelow)
* ASTERISK-24474 – sip_to_pjsip.py lacks documentation and does
not function (Reported by John Kiniston)
* ASTERISK-24672 – [PATCH] Memory leak in func_curl CURLOPT
(Reported by Kristian Høgh)
* ASTERISK-20744 – [patch] Security event logging does not work
over syslog (Reported by Michael Keuter)
* ASTERISK-24665 – Configure check required for
pjsip_get_dest_info() (Reported by Mark Michelson)
* ASTERISK-23850 – Park Application does not respect Return
Context Priority (Reported by Andrew Nagy)
* ASTERISK-23991 – [patch]asterisk.pc file contains a small error
in the CFlags returned (Reported by Diederik de Groot)
* ASTERISK-24655 – res_pjsip_outbound_publish: Hang on shutdown
while attempting to publish (Reported by Kevin Harwell)
* ASTERISK-24485 – res_pjsip cannot be unloaded or shutdown
(Reported by Corey Farrell)
* ASTERISK-24663 – [patch] Unnamed semaphore autoconf check fails
on cross compilation (Reported by abelbeck)
* ASTERISK-24624 – Transfer to invalid extension results in hung
channel. (Reported by Zane Conkle)
* ASTERISK-24615 – When Multiple Transports Exist in pjsip.conf,
Incorrect External Addresses is Used in SIP Packets When
Responding to INVITE (Reported by David Justl)
* ASTERISK-24288 – [patch] – ODBC usage with app_voicemail –
voicemail is not deleted after review, hangup (Reported by LEI
FU)
* ASTERISK-24048 – [patch] contrib/scripts/install_prereq selects
32-bit packages on 64-bit hosts (Reported by Ben Klang)
* ASTERISK-24600 – Stuck IAX channels, Asterisk stops responding
to most traffic, potential deadlock (Reported by Jeff Collell)
* ASTERISK-24560 – Creating a named ARI bridge twice causes a
crash (Reported by Kinsey Moore)
* ASTERISK-24682 – app_dial: Multiple DialEnd events emitted when
MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported
by Matt Jordan)
* ASTERISK-24640 – Registration pending stays forever after sip
reload (Reported by Max Man)
* ASTERISK-24673 – outgoing sip registers cannot be removed or
modified without doing restart (or doing module unload
chan_sip.so) (Reported by Stefan Engström)
* ASTERISK-24709 – [patch] msg_create_from_file used by MixMonitor
m() option does not queue an MWI event (Reported by Gareth
Palmer)
* ASTERISK-24649 – Pushing of channel into bridge fails; Stasis
fails to get app name (Reported by John Bigelow)
* ASTERISK-24355 – [patch] chan_sip realtime uses case sensitive
column comparison for ‘defaultuser’ (Reported by
HZMI8gkCvPpom0tM)
* ASTERISK-24693 – Investigate and fix memory leaks in Asterisk
(Reported by Kevin Harwell)
* ASTERISK-24626 – Voicemail passwords not being stored in ARA
(Reported by Paddy Grice)
* ASTERISK-24539 – Compile fails on OSX because of sem_timedwait
in bridge_channel.c (Reported by George Joseph)
* ASTERISK-24544 – Compile fails on OSX Yosemite because of
incorrect detection of htonll and ntohll (Reported by George
Joseph)
* ASTERISK-24723 – confbridge: CLI command ‘confbridge list XXXX’
no longer displays user menus (Reported by Matt Jordan)
* ASTERISK-24721 – manager: ModuleLoad action incorrectly reports
‘module not found’ during a Reload operation (Reported by Matt
Jordan)
* ASTERISK-24719 – ConfBridge recording channels get stuck when
recording started/stopped more than once (Reported by Richard
Mudgett)
* ASTERISK-24715 – chan_sip: stale nonce causes failure (Reported
by Kevin Harwell)
* ASTERISK-24728 – tcptls: Bad file descriptor error when
reloading chan_sip (Reported by Kevin Harwell)
* ASTERISK-24729 – Outbound registration not occuring on new
registrations after reload. (Reported by Richard Mudgett)
* ASTERISK-24676 – Security Vulnerability: URL request injection
in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
* ASTERISK-24666 – Security Vulnerability: RTP not closed after
sip call using unsupported codec (Reported by Y Ateya)
* ASTERISK-24711 – DTLS handshake broken with latest OpenSSL
versions (Reported by Jared Biel)
* ASTERISK-24646 – PJSIP changeset 4899 breaks TLS (Reported by
Stephan Eisvogel)
* ASTERISK-24736 – Memory Leak Fixes (Reported by Mark Michelson)
* ASTERISK-24635 – PJSIP outbound PUBLISH crashes when no response
is ever received (Reported by Marco Paland)
* ASTERISK-24737 – When agent not logged in, agent status shows
unavailable, queue status shows agent invalid (Reported by
Richard Mudgett)

Improvements made in this release:
———————————–
* ASTERISK-24552 – ARI: Allow associating a channel as an
initiator of an Origination for record keeping purposes
(Reported by Matt Jordan)
* ASTERISK-24553 – ARI/AMI: Include language in standard channel
snapshot output (Reported by Matt Jordan)
* ASTERISK-24643 – res_pjsip: Add user=phone option (Reported by
Matt Jordan)
* ASTERISK-24644 – res_pjsip_keepalive: Add keepalive module for
connection-oriented transports. (Reported by Matt Jordan)
* ASTERISK-24412 – [patch]Incomplete channel originate/continue
handling with ARI (Reported by Nir Simionovich (GreenfieldTech –
Israel))
* ASTERISK-24678 – [PATCH] Added atxfer* settings to
features.conf.sample (Reported by Niklas Larsson)
* ASTERISK-24575 – [patch]Make capath work for res_pjsip (Reported
by cloos)
* ASTERISK-24671 – Missing docs for the CDR AMI Event (Reported by
Dan Jenkins)
* ASTERISK-24316 – For httpd server, need option to define server
name for security purposes (Reported by Andrew Nagy)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0-rc1

Thank you for your continued support of Asterisk!

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