Asterisk 13.6.0 Now Available

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

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

The release of Asterisk 13.6.0 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-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.
  • [ASTERISK-25438] – res_rtp_asterisk: ICE role message even when ICE is not enabled
  • [ASTERISK-25449] – main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny

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, please see the ChangeLog:

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

Thank you for your continued support of Asterisk!

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