The Asterisk Development Team has announced the release of Asterisk 13.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.2.0 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: 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 H??gh) * 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 Engstr??m) * 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, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0 Thank you for your continued support of Asterisk!
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