Asterisk 13.6.0-rc1 Now Available

The Asterisk Development Team has announced the release of Asterisk 13.6.0-rc1.

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:

Bug

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

Improvement

  • [ASTERISK-24870] – ARI: Subscriptions to bridges generally not super useful
  • [ASTERISK-25310] – on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED

New Feature

  • [ASTERISK-25252] – ARI: Add the ability to manipulate log channels
  • [ASTERISK-25377] – res_pjsip: Change default “From user” from UUID to something more palatable

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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