[asterisk-dev] Asterisk 13.6.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Wed Sep 30 11:33:34 CDT 2015


The Asterisk Development Team has announced the first release candidate of
Asterisk 13.6.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.6.0-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 candidate:

New Features made in this release:
-----------------------------------
 * ASTERISK-25377 - res_pjsip: Change default "From user" from UUID
      to something more palatable (Reported by Mark Michelson)
 * ASTERISK-25252 - ARI: Add the ability to manipulate log channels
      (Reported by Matt Jordan)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-25383 - Core dumps on startup and shutdown with
      MALLOC_DEBUG enabled (Reported by yaron nahum)
 * ASTERISK-25423 - Caller gets no Connected line update during
      call pickup. (Reported by Richard Mudgett)
 * ASTERISK-25305 - Dynamic logger channels can be added multiple
      times (Reported by Mark Michelson)
 * ASTERISK-25418 - On-hold channels redirected out of a bridge
      appear to still be on hold (Reported by Mark Michelson)
 * ASTERISK-25384 - Regular Asterisk crashes when using Page
      application. "user_data is NULL" (Reported by Chet Stevens)
 * ASTERISK-25407 - Asterisk fails to log to multiple syslog
      destinations (Reported by Elazar Broad)
 * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
      populated (Reported by Kevin Harwell)
 * ASTERISK-25394 - pbx: Incorrect device and presence state when
      changing hint details (Reported by Joshua Colp)
 * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
      invalid SIP (Reported by Walter Doekes)
 * ASTERISK-25399 - app_queue: AgentComplete event has wrong reason
      (Reported by Kevin Harwell)
 * ASTERISK-25185 - Segfault in app_queue on transfer scenarios
      (Reported by Etienne Lessard)
 * ASTERISK-25353 - [patch] Transcoding while different in Frame
      size = Frames lost (Reported by Alexander Traud)
 * ASTERISK-25325 - ARI PUT reload chan_sip HTTP response 404
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-25390 - default_from_user can crash with certain
      configuration backends (Reported by Mark Michelson)
 * ASTERISK-25387 - res_pjsip_nat: Malformed REGISTER request
      causes NAT'd Contact header to not be rewritten (Reported by
      Matt Jordan)
 * ASTERISK-25227 - No audio at in-band announcements in ooh323
      channel (Reported by Alexandr Dranchuk)
 * ASTERISK-25369 - res_parking: ParkAndAnnounce - Inheritable
      variables aren't applied to the announcer channel (Reported by
      Jonathan Rose)
 * ASTERISK-25295 - res_pjsip crash - pjsip_uri_get_uri at
      /usr/include/pjsip/sip_uri.h (Reported by Dmitriy Serov)
 * ASTERISK-25299 - RTP port leaks with incoming OOH323 calls
      (Reported by Alexandr Dranchuk)
 * ASTERISK-25381 - res_pjsip: AoRs deleted via ARI (or other
      mechanism) do not destroy their related contacts (Reported by
      Matt Jordan)
 * ASTERISK-25367 - pbx: Long pattern match hints may cause "core
      show hints" to crash (Reported by Joshua Colp)
 * ASTERISK-25365 - Persistent subscriptions have extra
      Content-Length/corrupted messages (Reported by Mark Michelson)
 * ASTERISK-25362 - Deadlock due to presence state callback
      (Reported by Mark Michelson)
 * ASTERISK-25356 - res_pjsip_sdp_rtp: Multiple keepalive scheduled
      items may exist (Reported by Joshua Colp)
 * ASTERISK-25355 - sched: ast_sched_del may return prematurely due
      to spurious wakeup (Reported by Joshua Colp)
 * ASTERISK-25318 -
      tests/rest_api/applications/subscribe-endpoint/nominal/resource:
      Sporadically failing (Reported by Joshua Colp)
 * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
      cause on call pickup (Reported by Joshua Colp)
 * ASTERISK-25342 - res_pjsip: Repeated usage of pj_gethostip may
      block (Reported by Joshua Colp)
 * ASTERISK-25341 - bridge: Hangups may get lost when executing
      actions (Reported by Joshua Colp)
 * ASTERISK-25339 - res_pjsip: Empty "auth" sections from
      non-config backgrounds are interpreted as valid (Reported by
      Matt Jordan)
 * ASTERISK-25215 - Differences in queue.log between Set
      QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
      Gaetz)
 * ASTERISK-25322 - Crash occurs when using MixMonitor with t() or
      r() options. (Reported by Richard Mudgett)
 * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
      for wrong or non existent peer on invite (Reported by Kevin
      Harwell)
 * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
      tones. (Reported by Richard Mudgett)
 * ASTERISK-25312 - res_http_websocket: Terminate connection on
      fatal cases (Reported by Joshua Colp)
 * ASTERISK-25306 - Persistent subscriptions can save multiple SIP
      messages at once, leading to potential crashes. (Reported by
      Mark Michelson)
 * ASTERISK-25309 - [patch] iLBC 20 advertised (Reported by
      Alexander Traud)
 * ASTERISK-25304 - res_pjsip: XML sanitization may write past
      buffer (Reported by Joshua Colp)
 * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
      Firefox 39 - add ECDH support and fallback to prime256v1
      (Reported by Stefan Engström)
 * ASTERISK-25296 - RTP performance issue with several channel
      drivers. (Reported by Richard Mudgett)
 * ASTERISK-25297 - Crashes running
      channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
      (Reported by Richard Mudgett)
 * ASTERISK-25292 - Testuite:
      tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
      (Reported by Kevin Harwell)
 * ASTERISK-25271 - Parking & blind transfer: Transferer channel
      not hung up if no MOH (Reported by Kevin Harwell)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24870 - ARI: Subscriptions to bridges generally not
      super useful (Reported by Matt Jordan)
 * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
      defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.6.0-rc1

Thank you for your continued support of Asterisk!



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