Asterisk 13.2.0-rc1 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.2.0-rc1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

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:

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-rc1

Thank you for your continued support of Asterisk!

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