Il giorno 09 ottobre 2015, il Team di Sviluppo di Asterisk ha annunciato il rilascio di Asterisk 13.6.0.
Dal post originale:
[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] - [patch]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] - [patch] 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] - [patch] 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
[ASTERISK-24870] - ARI: Subscriptions to bridges generally not super useful
[ASTERISK-25310] - [patch]on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED
[ASTERISK-25252] - ARI: Add the ability to manipulate log channels
[ASTERISK-25377] - res_pjsip: Change default "From user" from UUID to something more palatable
Per la lista completa, questo il link al ChangeLog: