Asterisk 13.15.0-rc1 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.15.0-rc1.

This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

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:

Bug

  • [ASTERISK-15858] – Fix query with double backslash in string literals and stop log warnings
  • [ASTERISK-17067] – Long lines in call files cause spurious syntax error
  • [ASTERISK-18271] – Pattern matching with res_config_mysql extensions does not behave as expected
  • [ASTERISK-18286] – ‘Silence’ is truncated in Record()
  • [ASTERISK-18731] – DUNDi weight parameter not processed correctly
  • [ASTERISK-23457] – SQlite3: Realtime queue loading fails after PRAGMA query result
  • [ASTERISK-24287] – Race conditons and other problems in res_config_pgsql
  • [ASTERISK-24562] – app_voicemail: Cannot set fromstring on a per-mailbox basis
  • [ASTERISK-25237] – stasis_cache.c:845 caching_topic_exec: – misleading ERROR message
  • [ASTERISK-25628] – res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
  • [ASTERISK-25893] – Function vmauthenticate accesses uninitialized memory
  • [ASTERISK-26057] – res_config_sqlite3 uses incorrect query – unnecessary escape
  • [ASTERISK-26109] – Asterisk fails building with OpenSSL 1.1.0
  • [ASTERISK-26115] – pbx: AMI Originate ignore “failed” extension on call failure
  • [ASTERISK-26248] – chan_pjsip: Error when calling PJSIP client with domain specified
  • [ASTERISK-26313] – chan_sip : Asterisk restart seems to be required for changing encryption option
  • [ASTERISK-26340] – PJSIP 2.5.5 DNS error with IPv4
  • [ASTERISK-26484] – res_pjsip_messaging: Crash when using invalid URI in MessageSend ‘from’ argument.
  • [ASTERISK-26580] – Error during LDAP modify action when user unregisters
  • [ASTERISK-26598] – Saynumber is trying to get “and” from “digits/” subfolder
  • [ASTERISK-26623] – res_pjsip: Crash when calling PJSIPShowEndpoint
  • [ASTERISK-26643] – Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
  • [ASTERISK-26668] – core: Malformed pattern matching extension (various factors) results in crash
  • [ASTERISK-26669] – PJSIP Segfault 13.13.1 (Bundled PJSIP)
  • [ASTERISK-26685] – res_pjsip: Crash when using IPv6 and Transport ws,wss
  • [ASTERISK-26696] – pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
  • [ASTERISK-26705] – libasteriskssl.so not found when asterisk is installed for the 1st time
  • [ASTERISK-26714] – Phone default have not ringing on ARM
  • [ASTERISK-26717] – Document the fact that Asterisk HEP support only works with the PJSIP channel driver
  • [ASTERISK-26723] – VoiceMailPlayMsg not playing messages via realtime
  • [ASTERISK-26732] – res_rtp_asterisk: Implement RTCP Multiplexing – breaking WebRTC in Chrome
  • [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
  • [ASTERISK-26756] – res_pjsip_mwi: Asterisk does not terminate MWI subscription
  • [ASTERISK-26770] – res_stasis_device_state: Duplicate subscriptions when multiple received at same time
  • [ASTERISK-26772] – Crash in srv.c on startup with pjsip
  • [ASTERISK-26776] – res_pjsip_pubsub: Crash when generating xpidf content
  • [ASTERISK-26781] – bridge: Passing the ‘p’ (play tone) flag to Bridge() application results in garbled audio
  • [ASTERISK-26782] – res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
  • [ASTERISK-26785] – configs/samples: The ‘identify’ entry is in the wrong section in sorcery.conf.sample
  • [ASTERISK-26788] – core: Protect flags during ast_waitfor
  • [ASTERISK-26794] – http: Crash on Reload Only in ast_tcptls_server_start
  • [ASTERISK-26796] – res_pjsip_transport_websocket: Via header is ‘WS’ when it should be ‘WSS’
  • [ASTERISK-26799] – res_pjsip: Using an auth object for inbound and outbound authentication fails.
  • [ASTERISK-26802] – Integrity Check Of PJSIP Download Fails
  • [ASTERISK-26808] – res_pjsip_outbound_registration doesn’t know about network change events
  • [ASTERISK-26812] – Fix download_externals To Allow The Use Of curl Or wget
  • [ASTERISK-26822] – pjsip/cli_commands: pjsip show channelstats shows wrong codec
  • [ASTERISK-26823] – PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
  • [ASTERISK-26825] – pjsip.conf.sample: user_agent: still refers to branch 12
  • [ASTERISK-26841] – chan_sip: Call not cancelled after receiving a 422 response
  • [ASTERISK-26850] – res_hep_pjsip: Asterisk insert wrong protocol name in “Protocol ID” field in HEP packets
  • [ASTERISK-26851] – res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
  • [ASTERISK-26857] – chan_pjsip: Dialplan function race condition
  • [ASTERISK-26862] – app_queue: Queue stops calling members with local interface after forwarding in previous call
  • [ASTERISK-26865] – chan_iax2: Reload of iax peer results in loss of host address/port
  • [ASTERISK-26867] – autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
  • [ASTERISK-26869] – res_pjsip_refer: blind call transfer w/o a user name doesn’t go to the s extension
  • [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)
  • [ASTERISK-26879] – PJSIP external_media_address ignored if no local_net options are provided
  • [ASTERISK-26880] – Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled

Improvement

New Feature

  • [ASTERISK-17428] – Allow “Comedian Mail” branding to be removed
  • [ASTERISK-26863] – res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
  • [ASTERISK-26878] – func_channel: Add ability to get the callid so dialplan has access to it.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.15.0-rc1

Thank you for your continued support of Asterisk!

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