Nova versão do Asterisk 13.6.0 - Muitos bugs foram corrigidos.
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: 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-25449 - main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny (Reported by Matt Jordan) * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when ICE is not enabled (Reported by Joshua Colp) * 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-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, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.6.0 Thank you for your continued support of Asterisk! -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- Sylvio Jollenbeck skype: sylvio.jollenbeck www.hosannatecnologia.com.br
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