[asterisk-users] Asterisk 13.15.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Thu Mar 23 17:34:08 CDT 2017

The Asterisk Development Team has announced the release of Asterisk 13.15.0-rc1.
This release is available for immediate download at

The release of Asterisk 13.15.0-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:

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-26850 - res_hep_pjsip: Asterisk insert wrong protocol
      name in "Protocol ID" field in HEP packets (Reported by Max
 * ASTERISK-26484 - res_pjsip_messaging: Crash when using invalid
      URI in MessageSend 'from' argument. (Reported by Vinod
 * 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
 * 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-26851 - res_pjsip_sdp_rtp: RTP instance does not use
      same IP as explicit transport (Reported by Richard Begg)
 * 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
 * ASTERISK-26717 - Document the fact that Asterisk HEP support
      only works with the PJSIP channel driver (Reported by Olivier
 * 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-26705 - libasteriskssl.so not found when asterisk is
      installed for the 1st time (Reported by George Joseph)
 * ASTERISK-26781 - bridge: Passing the 'p' (play tone) flag to
      Bridge() application results in garbled audio (Reported by Sean
 * 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
 * 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
 * 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
 * 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
 * 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
 * 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:


Thank you for your continued support of Asterisk!

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