Asterisk 13.15.0 Now Available


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The Asterisk Development Team would like to announce the release of Asterisk 13.15.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.15.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

New Features made in this release:
-----------------------------------
 * ASTERISK-26878 - func_channel: Add ability to get the callid
      so dialplan has access to it.
      (Reported by Richard
      Mudgett)
 * ASTERISK-26863 - res_pjsip: Add endpoint identification
      scheme based on a configured SIP header/value
      (Reported by
      Matt Jordan)
 * ASTERISK-17428 - [patch] Allow "Comedian Mail" branding to be
      removed
      (Reported by John Covert)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-26851 - res_pjsip_sdp_rtp: RTP instance does not use
      same IP as explicit transport
      (Reported by Richard Begg)
 * ASTERISK-26897 - chan_sip: Security vulnerability with client
      code header
      (Reported by Alex Villacís Lasso)
 * ASTERISK-26916 - res_pjsip: Excessive refcount reached on
      transport ao2 object
      (Reported by Ross Beer)
 * ASTERISK-26705 - libasteriskssl.so not found when asterisk is
      installed for the 1st time
      (Reported by George Joseph)
 * ASTERISK-26850 - res_hep_pjsip: Asterisk insert wrong
      protocol name in "Protocol ID" field in HEP packets
     
      (Reported by Max Norba)
 * ASTERISK-26484 - res_pjsip_messaging: Crash when using
      invalid URI in MessageSend 'from' argument.
      (Reported by
      Vinod Dharashive)
 * ASTERISK-26776 - res_pjsip_pubsub: Crash when generating
      xpidf content
      (Reported by Andrew Green)
 * ASTERISK-26880 - Asterisk crashes when multiple speex users
      join confbridge with pp_vad and dtx enabled
      (Reported by
      Kirsty Tyerman)
 * ASTERISK-26862 - app_queue: Queue stops calling members with
      local interface after forwarding in previous call
     
      (Reported by Robert Mordec)
 * ASTERISK-26732 - res_rtp_asterisk: Implement RTCP
      Multiplexing - breaking WebRTC in Chrome
      (Reported by Dan
      Jenkins)
 * ASTERISK-26879 - PJSIP external_media_address ignored if no
      local_net options are provided
      (Reported by Matt Jordan)
 * ASTERISK-26867 - autochan: Locking in a function
      ast_autochan_destroy() on destroyed channel (after masquerade).

      (Reported by Krzysztof Trempala)
 * ASTERISK-26869 - res_pjsip_refer: blind call transfer w/o a
      user name doesn't go to the s extension
      (Reported by
      Torrey Searle)
 * ASTERISK-26668 - core: Malformed pattern matching extension
      (various factors) results in crash
      (Reported by xrobau)
 * ASTERISK-26865 - chan_iax2: Reload of iax peer results in
      loss of host address/port
      (Reported by Richard Begg)
 * ASTERISK-26872 - Bundled pjproject fails to build when
      tarball downloaded with curl due to md5 verification failure in
      Docker containers (or when there is no terminal)
      (Reported
      by Matt Jordan)
 * ASTERISK-26717 - Document the fact that Asterisk HEP support
      only works with the PJSIP channel driver
      (Reported by
      Olivier Krief)
 * ASTERISK-26643 - Extra new line in Device field of
      DeviceStateChange AMI Event after restart of Asterisk
     
      (Reported by Roman Bedros)
 * ASTERISK-25237 -  stasis_cache.c:845 caching_topic_exec: -
      misleading ERROR message
      (Reported by Smirnov Aleksey)
 * ASTERISK-26857 - chan_pjsip: Dialplan function race
      condition
      (Reported by Joshua Colp)
 * ASTERISK-26841 - chan_sip: Call not cancelled after receiving
      a 422 response
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-26822 - pjsip/cli_commands: pjsip show channelstats
      shows wrong codec
      (Reported by Kevin Harwell)
 * ASTERISK-26685 - res_pjsip: Crash when using IPv6 and
      Transport ws,wss
      (Reported by Michael Balen)
 * ASTERISK-24562 - app_voicemail: Cannot set fromstring on a
      per-mailbox basis
      (Reported by Mark Scholten)
 * ASTERISK-26598 - Saynumber is trying to get "and" from
      "digits/" subfolder
      (Reported by Jonathan Harris)
 * ASTERISK-17067 - Long lines in call files cause spurious
      syntax error
      (Reported by Dave Olszewski)
 * ASTERISK-26796 - res_pjsip_transport_websocket: Via header is
      'WS' when it should be 'WSS'
      (Reported by Jørgen H)
 * ASTERISK-25628 - res_config_pgsql: should match the behavior
      of other drivers so that queue_log can disable adaptive logging

      (Reported by Dmitry Wagin)
 * ASTERISK-26825 - pjsip.conf.sample: user_agent: still refers
      to branch 12
      (Reported by Tzafrir Cohen)
 * ASTERISK-26823 - PJSIP: Persistent subscriptions can cause
      FRACKs if endpoint does not exist
      (Reported by Mark
      Michelson)
 * ASTERISK-26623 - res_pjsip: Crash when calling
      PJSIPShowEndpoint
      (Reported by Jørgen H)
 * ASTERISK-26808 - res_pjsip_outbound_registration doesn't know
      about network change events
      (Reported by George Joseph)
 * ASTERISK-26313 - chan_sip : Asterisk restart seems to be
      required for changing encryption option
      (Reported by
      benasse)
 * ASTERISK-26781 - bridge: Passing the 'p' (play tone) flag to
      Bridge() application results in garbled audio
      (Reported by
      Sean Bright)
 * ASTERISK-26782 - res_pjsip: URI requirement for fields is not
      consistently documented and error does not provide indication
  
      (Reported by Peter Sokolov)
 * ASTERISK-26812 - [patch] Fix download_externals To Allow The
      Use Of curl Or wget
      (Reported by Michael L. Young)
 * ASTERISK-18271 - Pattern matching with res_config_mysql
      extensions does not behave as expected
      (Reported by
      Charlie Smurthwaite)
 * ASTERISK-26669 - PJSIP Segfault 13.13.1 (Bundled PJSIP)
     
      (Reported by Nic Colledge)
 * ASTERISK-18731 - [patch] DUNDi weight parameter not processed
      correctly
      (Reported by Peter Racz)
 * ASTERISK-26580 - [patch] Error during LDAP modify action when
      user unregisters
      (Reported by Nicholas John Koch)
 * ASTERISK-26799 - res_pjsip: Using an auth object for inbound
      and outbound authentication fails.
      (Reported by Richard
      Mudgett)
 * ASTERISK-26738 - Frequent segfaults since activation of DNS
      SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c,
      and pj_atomic_inc_and_get at pj/os_core_unix.c
      (Reported
      by Michael Maier)
 * ASTERISK-25893 - Function vmauthenticate accesses
      uninitialized memory
      (Reported by Filip Jenicek)
 * ASTERISK-26802 - [patch] Integrity Check Of PJSIP Download
      Fails
      (Reported by Michael L. Young)
 * ASTERISK-15858 - [patch] Fix query with double backslash in
      string literals and stop log warnings
      (Reported by
      Humberto Figuera)
 * ASTERISK-26057 - res_config_sqlite3 uses incorrect query -
      unnecessary escape
      (Reported by Stepan)
 * ASTERISK-23457 - SQlite3: Realtime queue loading fails after
      PRAGMA query result
      (Reported by Scott Griepentrog)
 * ASTERISK-26794 - http: Crash on Reload Only in
      ast_tcptls_server_start
      (Reported by Joshua Elson)
 * ASTERISK-26714 - Phone default have not ringing on ARM
     
      (Reported by Igor Goncharovsky)
 * ASTERISK-26696 - pjsip_pubsub: PJSIP Subscription Persistence
      in AstDB Does not update on subscription refresh
      (Reported
      by Zach R)
 * ASTERISK-26756 - res_pjsip_mwi: Asterisk does not terminate
      MWI subscription
      (Reported by Carl Fortin)
 * ASTERISK-26109 - Asterisk fails building with OpenSSL 1.1.0
 
      (Reported by Tzafrir Cohen)
 * ASTERISK-26723 - VoiceMailPlayMsg not playing messages via
      realtime
      (Reported by Ryan Rittgarn)
 * ASTERISK-18286 - [patch] 'Silence' is truncated in Record()
 
      (Reported by var)
 * ASTERISK-26248 - chan_pjsip: Error when calling PJSIP client
      with domain specified
      (Reported by Norbert Varga)
 * ASTERISK-26788 - core: Protect flags during ast_waitfor
     
      (Reported by Joshua Colp)
 * ASTERISK-26115 - pbx: AMI Originate ignore "failed" extension
      on call failure
      (Reported by Nasir Iqbal)
 * ASTERISK-26785 - configs/samples:  The 'identify' entry is in
      the wrong section in sorcery.conf.sample
      (Reported by
      Torrey Searle)
 * ASTERISK-26772 - Crash in srv.c on startup with pjsip
     
      (Reported by nappsoft)
 * ASTERISK-26770 - res_stasis_device_state: Duplicate
      subscriptions when multiple received at same time
     
      (Reported by Joshua Colp)

Improvements made in this release:
-----------------------------------
 * ASTERISK-26864 - res_pjsip_session: Add support for overlap
      dialling
      (Reported by Richard Begg)
 * ASTERISK-26846 - chan_sip: Add rtcp-mux support
     
      (Reported by Sean Bright)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.15.0

Thank you for your continued support of Asterisk!

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