[asterisk-dev] Asterisk 13.2.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Fri Feb 6 15:53:45 CST 2015

The Asterisk Development Team has announced the release of Asterisk 13.2.0.
This release is available for immediate download at

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
 * 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
 * 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
 * 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
 * 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
 * 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
 * ASTERISK-24719 - ConfBridge recording channels get stuck when
      recording started/stopped more than once (Reported by Richard
 * 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 -
 * 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:


Thank you for your continued support of Asterisk!

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