Asterisk 13.2.0 Now Available

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:

Improvements

  • [ASTERISK-24316] – For httpd server, need option to define server name for security purposes
  • [ASTERISK-24412] – Incomplete channel originate/continue handling with ARI
  • [ASTERISK-24552] – ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes
  • [ASTERISK-24553] – ARI/AMI: Include language in standard channel snapshot output
  • [ASTERISK-24575] – Make capath work for res_pjsip
  • [ASTERISK-24643] – res_pjsip: Add user=phone option
  • [ASTERISK-24644] – res_pjsip_keepalive: Add keepalive module for connection-oriented transports.
  • [ASTERISK-24671] – Missing docs for the CDR AMI Event
  • [ASTERISK-24678] – [PATCH] Added atxfer* settings to features.conf.sample

Bugs

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

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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