Asterisk 14.4.0 Now Available

The Asterisk Development Team would like to announce the release of Asterisk 14.4.0.
This release is available for immediate download at

The release of Asterisk 14.4.0 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
 * 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
  (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 - 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
 * 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)
  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
  (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-26353 - res_musiconhold: musiconhold seems to think
  that the general section is a class and issues warning
  (Reported by Jonathan Harris)
 * 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-26774 - core: Playback URL fails after some time
  (Reported by Igor Gamayunov)
 * 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
 * ASTERISK-26623 - res_pjsip: Crash when calling
  (Reported by Jørgen H)
 * ASTERISK-26808 - res_pjsip_outbound_registration doesn't know
  about network change events
  (Reported by George Joseph)
 * 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
  (Reported by Peter Racz)
 * ASTERISK-26799 - res_pjsip: Using an auth object for inbound
  and outbound authentication fails.
  (Reported by Richard
 * 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
  by Michael Maier)
 * ASTERISK-25893 - Function vmauthenticate accesses
  uninitialized memory
  (Reported by Filip Jenicek)
 * ASTERISK-26580 - [patch] Error during LDAP modify action when
  user unregisters
  (Reported by Nicholas John Koch)
 * ASTERISK-26802 - [patch] Integrity Check Of PJSIP Download
  (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
  (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
  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
  (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
  (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:

Thank you for your continued support of Asterisk!